Merge remote-tracking branch 'asoc/for-5.9' into asoc-next

Documentation/devicetree/bindings/sound/adi,adau1977.txt

 Analog Devices ADAU1977/ADAU1978/ADAU1979
 
 Datasheets:
-http://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1977.pdf
-http://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1978.pdf
-http://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1979.pdf
+https://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1977.pdf
+https://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1978.pdf
+https://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1979.pdf
 
 This driver supports both the I2C and SPI bus.
 

Documentation/devicetree/bindings/sound/ak4613.txt

-AK4613 I2C transmitter
-
-This device supports I2C mode only.
-
-Required properties:
-
-- compatible : "asahi-kasei,ak4613"
-- reg : The chip select number on the I2C bus
-
-Optional properties:
-- asahi-kasei,in1-single-end	: Boolean. Indicate input / output pins are single-ended.
-- asahi-kasei,in2-single-end	  rather than differential.
-- asahi-kasei,out1-single-end
-- asahi-kasei,out2-single-end
-- asahi-kasei,out3-single-end
-- asahi-kasei,out4-single-end
-- asahi-kasei,out5-single-end
-- asahi-kasei,out6-single-end
-
-Example:
-
-&i2c {
-	ak4613: ak4613@10 {
-		compatible = "asahi-kasei,ak4613";
-		reg = <0x10>;
-	};
-};

Documentation/devicetree/bindings/sound/ak4613.yaml

+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/ak4613.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: AK4613 I2C transmitter Device Tree Bindings
+
+maintainers:
+  - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+
+properties:
+  compatible:
+    const: asahi-kasei,ak4613
+
+  reg:
+    maxItems: 1
+
+  clocks:
+    maxItems: 1
+
+  "#sound-dai-cells":
+    const: 0
+
+patternProperties:
+  "^asahi-kasei,in[1-2]-single-end$":
+    description: Input Pin 1 - 2.
+    $ref: /schemas/types.yaml#/definitions/flag
+
+  "^asahi-kasei,out[1-6]-single-end$":
+    description: Output Pin 1 - 6.
+    $ref: /schemas/types.yaml#/definitions/flag
+
+required:
+  - compatible
+  - reg
+
+additionalProperties: false
+
+examples:
+  - |
+    i2c {
+        #address-cells = <1>;
+        #size-cells = <0>;
+        ak4613: codec@10 {
+            compatible = "asahi-kasei,ak4613";
+            reg = <0x10>;
+        };
+    };

Documentation/devicetree/bindings/sound/ak4642.txt

-AK4642 I2C transmitter
-
-This device supports I2C mode only.
-
-Required properties:
-
-  - compatible : "asahi-kasei,ak4642" or "asahi-kasei,ak4643" or "asahi-kasei,ak4648"
-  - reg : The chip select number on the I2C bus
-
-Optional properties:
-
-  - #clock-cells :		common clock binding; shall be set to 0
-  - clocks :			common clock binding; MCKI clock
-  - clock-frequency :		common clock binding; frequency of MCKO
-  - clock-output-names :	common clock binding; MCKO clock name
-
-Example 1:
-
-&i2c {
-	ak4648: ak4648@12 {
-		compatible = "asahi-kasei,ak4642";
-		reg = <0x12>;
-	};
-};
-
-Example 2:
-
-&i2c {
-	ak4643: codec@12 {
-		compatible = "asahi-kasei,ak4643";
-		reg = <0x12>;
-		#clock-cells = <0>;
-		clocks = <&audio_clock>;
-		clock-frequency = <12288000>;
-		clock-output-names = "ak4643_mcko";
-	};
-};

Documentation/devicetree/bindings/sound/ak4642.yaml

+# SPDX-License-Identifier: GPL-2.0
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/ak4642.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: AK4642 I2C transmitter Device Tree Bindings
+
+maintainers:
+  - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+
+properties:
+  compatible:
+    enum:
+      - asahi-kasei,ak4642
+      - asahi-kasei,ak4643
+      - asahi-kasei,ak4648
+
+  reg:
+    maxItems: 1
+
+  "#clock-cells":
+    const: 0
+  "#sound-dai-cells":
+    const: 0
+
+  clocks:
+    maxItems: 1
+
+  clock-frequency:
+    description: common clock binding; frequency of MCKO
+    $ref: /schemas/types.yaml#/definitions/uint32
+
+  clock-output-names:
+    description: common clock name
+    $ref: /schemas/types.yaml#/definitions/string
+
+required:
+  - compatible
+  - reg
+
+additionalProperties: false
+
+examples:
+  - |
+    i2c {
+        #address-cells = <1>;
+        #size-cells = <0>;
+        ak4643: codec@12 {
+            compatible = "asahi-kasei,ak4643";
+            #sound-dai-cells = <0>;
+            reg = <0x12>;
+            #clock-cells = <0>;
+            clocks = <&audio_clock>;
+            clock-frequency = <12288000>;
+            clock-output-names = "ak4643_mcko";
+        };
+    };

Documentation/devicetree/bindings/sound/everest,es8316.txt

-Everest ES8316 audio CODEC
-
-This device supports both I2C and SPI.
-
-Required properties:
-
-  - compatible  : should be "everest,es8316"
-  - reg : the I2C address of the device for I2C
-
-Optional properties:
-
-  - clocks : a list of phandle, should contain entries for clock-names
-  - clock-names : should include as follows:
-         "mclk" : master clock (MCLK) of the device
-
-Example:
-
-es8316: codec@11 {
-	compatible = "everest,es8316";
-	reg = <0x11>;
-	clocks = <&clks 10>;
-	clock-names = "mclk";
-};

Documentation/devicetree/bindings/sound/everest,es8316.yaml

+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/everest,es8316.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Everest ES8316 audio CODEC
+
+maintainers:
+  - Daniel Drake <drake@endlessm.com>
+  - Katsuhiro Suzuki <katsuhiro@katsuster.net>
+
+properties:
+  compatible:
+    const: everest,es8316
+
+  reg:
+    maxItems: 1
+
+  clocks:
+    items:
+      - description: clock for master clock (MCLK)
+
+  clock-names:
+    items:
+      - const: mclk
+
+  "#sound-dai-cells":
+    const: 0
+
+required:
+  - compatible
+  - reg
+  - "#sound-dai-cells"
+
+additionalProperties: false
+
+examples:
+  - |
+    i2c0 {
+      #address-cells = <1>;
+      #size-cells = <0>;
+      es8316: codec@11 {
+        compatible = "everest,es8316";
+        reg = <0x11>;
+        clocks = <&clks 10>;
+        clock-names = "mclk";
+        #sound-dai-cells = <0>;
+      };
+    };

Documentation/devicetree/bindings/sound/fsl,spdif.txt

@@ -6,7 +6,11 @@ a fibre cable.
 
 Required properties:
 
-  - compatible		: Compatible list, must contain "fsl,imx35-spdif".
+  - compatible		: Compatible list, should contain one of the following
+			  compatibles:
+			  "fsl,imx35-spdif",
+			  "fsl,vf610-spdif",
+			  "fsl,imx6sx-spdif",
 
   - reg			: Offset and length of the register set for the device.
 

Documentation/devicetree/bindings/sound/fsl-asoc-card.txt

@@ -34,6 +34,10 @@ The compatible list for this generic sound card currently:
 
  "fsl,imx-audio-wm8960"
 
+ "fsl,imx-audio-mqs"
+
+ "fsl,imx-audio-wm8524"
+
 Required properties:
 
   - compatible		: Contains one of entries in the compatible list.
@@ -44,6 +48,11 @@ Required properties:
 
   - audio-codec		: The phandle of an audio codec
 
+Optional properties:
+
+  - audio-asrc		: The phandle of ASRC. It can be absent if there's no
+			  need to add ASRC support via DPCM.
+
   - audio-routing	: A list of the connections between audio components.
 			  Each entry is a pair of strings, the first being the
 			  connection's sink, the second being the connection's
@@ -60,10 +69,13 @@ Required properties:
 			        coexisting in order to support the old bindings
 				of wm8962 and sgtl5000.
 
-Optional properties:
-
-  - audio-asrc		: The phandle of ASRC. It can be absent if there's no
-			  need to add ASRC support via DPCM.
+  - hp-det-gpio		: The GPIO that detect headphones are plugged in
+  - mic-det-gpio	: The GPIO that detect microphones are plugged in
+  - bitclock-master	: Indicates dai-link bit clock master; for details see simple-card.yaml.
+  - frame-master	: Indicates dai-link frame master; for details see simple-card.yaml.
+  - dai-format		: audio format, for details see simple-card.yaml.
+  - frame-inversion	: dai-link uses frame clock inversion, for details see simple-card.yaml.
+  - bitclock-inversion	: dai-link uses bit clock inversion, for details see simple-card.yaml.
 
 Optional unless SSI is selected as a CPU DAI:
 

Documentation/devicetree/bindings/sound/intel,keembay-i2s.yaml

+# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause)
+# Copyright 2020 Intel Corporation
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/intel,keembay-i2s.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Intel KeemBay I2S Device Tree Bindings
+
+maintainers:
+  - Sia, Jee Heng <jee.heng.sia@intel.com>
+
+description: |
+ Intel KeemBay I2S
+
+properties:
+  compatible:
+    enum:
+      - intel,keembay-i2s
+
+  "#sound-dai-cells":
+    const: 0
+
+  reg:
+    items:
+      - description: I2S registers
+      - description: I2S gen configuration
+
+  reg-names:
+    items:
+      - const: i2s-regs
+      - const: i2s_gen_cfg
+
+  interrupts:
+    maxItems: 1
+
+  clocks:
+    items:
+      - description: Bus Clock
+      - description: Module Clock
+
+  clock-names:
+    items:
+      - const: osc
+      - const: apb_clk
+
+required:
+  - compatible
+  - "#sound-dai-cells"
+  - reg
+  - clocks
+  - clock-names
+  - interrupts
+
+examples:
+  - |
+     #include <dt-bindings/interrupt-controller/arm-gic.h>
+     #include <dt-bindings/interrupt-controller/irq.h>
+     #define KEEM_BAY_PSS_AUX_I2S3
+     #define KEEM_BAY_PSS_I2S3
+     i2s3: i2s@20140000 {
+         compatible = "intel,keembay-i2s";
+         #sound-dai-cells = <0>;
+         reg = <0x20140000 0x200>, /* I2S registers */
+               <0x202a00a4 0x4>; /* I2S gen configuration */
+         reg-names = "i2s-regs", "i2s_gen_cfg";
+         interrupts = <GIC_SPI 120 IRQ_TYPE_LEVEL_HIGH>;
+         clock-names = "osc", "apb_clk";
+         clocks = <&scmi_clk KEEM_BAY_PSS_AUX_I2S3>, <&scmi_clk KEEM_BAY_PSS_I2S3>;
+     };

Documentation/devicetree/bindings/sound/max98357a.txt

-Maxim MAX98357A audio DAC
+Maxim MAX98357A/MAX98360A audio DAC
 
-This node models the Maxim MAX98357A DAC.
+This node models the Maxim MAX98357A/MAX98360A DAC.
 
 Required properties:
-- compatible   : "maxim,max98357a"
+- compatible   : "maxim,max98357a" for MAX98357A.
+                 "maxim,max98360a" for MAX98360A.
 
 Optional properties:
 - sdmode-gpios : GPIO specifier for the chip's SD_MODE pin.
@@ -20,3 +21,8 @@ max98357a {
 	compatible = "maxim,max98357a";
 	sdmode-gpios = <&qcom_pinmux 25 0>;
 };
+
+max98360a {
+	compatible = "maxim,max98360a";
+	sdmode-gpios = <&qcom_pinmux 25 0>;
+};

Documentation/devicetree/bindings/sound/maxim,max98390.yaml

+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/maxim,max98390.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Maxim Integrated MAX98390 Speaker Amplifier with Integrated Dynamic Speaker Management
+
+maintainers:
+  - Steve Lee <steves.lee@maximintegrated.com>
+
+properties:
+  compatible:
+      const: maxim,max98390
+
+  reg:
+    maxItems: 1
+    description: I2C address of the device.
+
+  maxim,temperature_calib:
+    allOf:
+      - $ref: /schemas/types.yaml#/definitions/uint32
+    description: The calculated temperature data was measured while doing the calibration.
+    minimum: 0
+    maximum: 65535
+
+  maxim,r0_calib:
+    allOf:
+      - $ref: /schemas/types.yaml#/definitions/uint32
+    description: This is r0 calibration data which was measured in factory mode.
+    minimum: 1
+    maximum: 8388607
+
+required:
+  - compatible
+  - reg
+
+additionalProperties: false
+
+examples:
+  - |
+    i2c {
+      #address-cells = <1>;
+      #size-cells = <0>;
+      max98390: amplifier@38 {
+        compatible = "maxim,max98390";
+        reg = <0x38>;
+        maxim,temperature_calib = <1024>;
+        maxim,r0_calib = <100232>;
+      };
+    };

Documentation/devicetree/bindings/sound/mt6358.txt

@@ -10,9 +10,15 @@ Required properties:
 - compatible : "mediatek,mt6358-sound".
 - Avdd-supply : power source of AVDD
 
+Optional properties:
+- mediatek,dmic-mode : Indicates how many data pins are used to transmit two
+	channels of PDM signal. 0 means two wires, 1 means one wire. Default
+	value is 0.
+
 Example:
 
 mt6358_snd {
 	compatible = "mediatek,mt6358-sound";
 	Avdd-supply = <&mt6358_vaud28_reg>;
+	mediatek,dmic-mode = <0>;
 };

Documentation/devicetree/bindings/sound/mt8183-da7219-max98357.txt

-MT8183 with MT6358, DA7219 and MAX98357 CODECS
+MT8183 with MT6358, DA7219, MAX98357, and RT1015 CODECS
 
 Required properties:
-- compatible : "mediatek,mt8183_da7219_max98357"
+- compatible : "mediatek,mt8183_da7219_max98357" for MAX98357A codec
+               "mediatek,mt8183_da7219_rt1015" for RT1015 codec
 - mediatek,headset-codec: the phandles of da7219 codecs
 - mediatek,platform: the phandle of MT8183 ASoC platform
 
+Optional properties:
+- mediatek,hdmi-codec: the phandles of HDMI codec
+
 Example:
 
 	sound {
 		compatible = "mediatek,mt8183_da7219_max98357";
 		mediatek,headset-codec = <&da7219>;
+		mediatek,hdmi-codec = <&it6505dptx>;
 		mediatek,platform = <&afe>;
 	};
 

Documentation/devicetree/bindings/sound/mt8183-mt6358-ts3a227-max98357.txt

-MT8183 with MT6358, TS3A227 and MAX98357 CODECS
+MT8183 with MT6358, TS3A227, MAX98357, and RT1015 CODECS
 
 Required properties:
-- compatible : "mediatek,mt8183_mt6358_ts3a227_max98357"
+- compatible : "mediatek,mt8183_mt6358_ts3a227_max98357" for MAX98357A codec
+               "mediatek,mt8183_mt6358_ts3a227_max98357b" for MAX98357B codec
+               "mediatek,mt8183_mt6358_ts3a227_rt1015" for RT1015 codec
 - mediatek,platform: the phandle of MT8183 ASoC platform
 
 Optional properties:
 - mediatek,headset-codec: the phandles of ts3a227 codecs
 - mediatek,ec-codec: the phandle of EC codecs.
                      See google,cros-ec-codec.txt for more details.
+- mediatek,hdmi-codec: the phandles of HDMI codec
 
 Example:
 
@@ -15,6 +18,7 @@ Example:
 		compatible = "mediatek,mt8183_mt6358_ts3a227_max98357";
 		mediatek,headset-codec = <&ts3a227>;
 		mediatek,ec-codec = <&ec_codec>;
+		mediatek,hdmi-codec = <&it6505dptx>;
 		mediatek,platform = <&afe>;
 	};
 

Documentation/devicetree/bindings/sound/nvidia,tegra186-dspk.yaml

+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/nvidia,tegra186-dspk.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Tegra186 DSPK Controller Device Tree Bindings
+
+description: |
+  The Digital Speaker Controller (DSPK) can be viewed as a Pulse
+  Density Modulation (PDM) transmitter that up-samples the input to
+  the desired sampling rate by interpolation and then converts the
+  over sampled Pulse Code Modulation (PCM) input to the desired 1-bit
+  output via Delta Sigma Modulation (DSM).
+
+maintainers:
+  - Jon Hunter <jonathanh@nvidia.com>
+  - Sameer Pujar <spujar@nvidia.com>
+
+properties:
+  $nodename:
+    pattern: "^dspk@[0-9a-f]*$"
+
+  compatible:
+    oneOf:
+      - const: nvidia,tegra186-dspk
+      - items:
+          - const: nvidia,tegra194-dspk
+          - const: nvidia,tegra186-dspk
+
+  reg:
+    maxItems: 1
+
+  clocks:
+    maxItems: 1
+
+  clock-names:
+    const: dspk
+
+  assigned-clocks:
+    maxItems: 1
+
+  assigned-clock-parents:
+    maxItems: 1
+
+  assigned-clock-rates:
+    maxItems: 1
+
+  sound-name-prefix:
+    pattern: "^DSPK[1-9]$"
+    allOf:
+      - $ref: /schemas/types.yaml#/definitions/string
+    description:
+      Used as prefix for sink/source names of the component. Must be a
+      unique string among multiple instances of the same component.
+      The name can be "DSPK1" or "DSPKx", where x depends on the maximum
+      available instances on a Tegra SoC.
+
+required:
+  - compatible
+  - reg
+  - clocks
+  - clock-names
+  - assigned-clocks
+  - assigned-clock-parents
+  - sound-name-prefix
+
+examples:
+  - |
+    #include<dt-bindings/clock/tegra186-clock.h>
+
+    dspk@2905000 {
+        compatible = "nvidia,tegra186-dspk";
+        reg = <0x2905000 0x100>;
+        clocks = <&bpmp TEGRA186_CLK_DSPK1>;
+        clock-names = "dspk";
+        assigned-clocks = <&bpmp TEGRA186_CLK_DSPK1>;
+        assigned-clock-parents = <&bpmp TEGRA186_CLK_PLL_A_OUT0>;
+        assigned-clock-rates = <12288000>;
+        sound-name-prefix = "DSPK1";
+    };
+
+...

Documentation/devicetree/bindings/sound/nvidia,tegra210-admaif.yaml

+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/nvidia,tegra210-admaif.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Tegra210 ADMAIF Device Tree Bindings
+
+description: |
+  ADMAIF is the interface between ADMA and AHUB. Each ADMA channel
+  that sends/receives data to/from AHUB must interface through an
+  ADMAIF channel. ADMA channel sending data to AHUB pairs with ADMAIF
+  Tx channel and ADMA channel receiving data from AHUB pairs with
+  ADMAIF Rx channel.
+
+maintainers:
+  - Jon Hunter <jonathanh@nvidia.com>
+  - Sameer Pujar <spujar@nvidia.com>
+
+properties:
+  $nodename:
+    pattern: "^admaif@[0-9a-f]*$"
+
+  compatible:
+    oneOf:
+      - enum:
+          - nvidia,tegra210-admaif
+          - nvidia,tegra186-admaif
+      - items:
+          - const: nvidia,tegra194-admaif
+          - const: nvidia,tegra186-admaif
+
+  reg:
+    maxItems: 1
+
+  dmas: true
+
+  dma-names: true
+
+if:
+  properties:
+    compatible:
+      contains:
+        const: nvidia,tegra210-admaif
+
+then:
+  properties:
+    dmas:
+      description:
+        DMA channel specifiers, equally divided for Tx and Rx.
+      minItems: 1
+      maxItems: 20
+    dma-names:
+      items:
+        pattern: "^[rt]x(10|[1-9])$"
+      description:
+        Should be "rx1", "rx2" ... "rx10" for DMA Rx channel
+        Should be "tx1", "tx2" ... "tx10" for DMA Tx channel
+      minItems: 1
+      maxItems: 20
+
+else:
+  properties:
+    dmas:
+      description:
+        DMA channel specifiers, equally divided for Tx and Rx.
+      minItems: 1
+      maxItems: 40
+    dma-names:
+      items:
+        pattern: "^[rt]x(1[0-9]|[1-9]|20)$"
+      description:
+        Should be "rx1", "rx2" ... "rx20" for DMA Rx channel
+        Should be "tx1", "tx2" ... "tx20" for DMA Tx channel
+      minItems: 1
+      maxItems: 40
+
+required:
+  - compatible
+  - reg
+  - dmas
+  - dma-names
+
+examples:
+  - |
+    admaif@702d0000 {
+        compatible = "nvidia,tegra210-admaif";
+        reg = <0x702d0000 0x800>;
+        dmas = <&adma 1>,  <&adma 1>,
+               <&adma 2>,  <&adma 2>,
+               <&adma 3>,  <&adma 3>,
+               <&adma 4>,  <&adma 4>,
+               <&adma 5>,  <&adma 5>,
+               <&adma 6>,  <&adma 6>,
+               <&adma 7>,  <&adma 7>,
+               <&adma 8>,  <&adma 8>,
+               <&adma 9>,  <&adma 9>,
+               <&adma 10>, <&adma 10>;
+        dma-names = "rx1",  "tx1",
+                    "rx2",  "tx2",
+                    "rx3",  "tx3",
+                    "rx4",  "tx4",
+                    "rx5",  "tx5",
+                    "rx6",  "tx6",
+                    "rx7",  "tx7",
+                    "rx8",  "tx8",
+                    "rx9",  "tx9",
+                    "rx10", "tx10";
+    };
+
+...

Documentation/devicetree/bindings/sound/nvidia,tegra210-ahub.yaml

+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/nvidia,tegra210-ahub.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Tegra210 AHUB Device Tree Bindings
+
+description: |
+  The Audio Hub (AHUB) comprises a collection of hardware accelerators
+  for audio pre-processing, post-processing and a programmable full
+  crossbar for routing audio data across these accelerators. It has
+  external interfaces such as I2S, DMIC, DSPK. It interfaces with ADMA
+  engine through ADMAIF.
+
+maintainers:
+  - Jon Hunter <jonathanh@nvidia.com>
+  - Sameer Pujar <spujar@nvidia.com>
+
+properties:
+  $nodename:
+    pattern: "^ahub@[0-9a-f]*$"
+
+  compatible:
+    oneOf:
+      - enum:
+          - nvidia,tegra210-ahub
+          - nvidia,tegra186-ahub
+      - items:
+          - const: nvidia,tegra194-ahub
+          - const: nvidia,tegra186-ahub
+
+  reg:
+    maxItems: 1
+
+  clocks:
+    maxItems: 1
+
+  clock-names:
+    const: ahub
+
+  assigned-clocks:
+    maxItems: 1
+
+  assigned-clock-parents:
+    maxItems: 1
+
+  assigned-clock-rates:
+    maxItems: 1
+
+  "#address-cells":
+    const: 1
+
+  "#size-cells":
+    const: 1
+
+  ranges: true
+
+required:
+  - compatible
+  - reg
+  - clocks
+  - clock-names
+  - assigned-clocks
+  - assigned-clock-parents
+  - "#address-cells"
+  - "#size-cells"
+  - ranges
+
+examples:
+  - |
+    #include<dt-bindings/clock/tegra210-car.h>
+
+    ahub@702d0800 {
+        compatible = "nvidia,tegra210-ahub";
+        reg = <0x702d0800 0x800>;
+        clocks = <&tegra_car TEGRA210_CLK_D_AUDIO>;
+        clock-names = "ahub";
+        assigned-clocks = <&tegra_car TEGRA210_CLK_D_AUDIO>;
+        assigned-clock-parents = <&tegra_car TEGRA210_CLK_PLL_A_OUT0>;
+        #address-cells = <1>;
+        #size-cells = <1>;
+        ranges = <0x702d0000 0x702d0000 0x0000e400>;
+
+        // All AHUB child nodes below
+        admaif@702d0000 {
+            compatible = "nvidia,tegra210-admaif";
+            reg = <0x702d0000 0x800>;
+            dmas = <&adma 1>,  <&adma 1>,
+               <&adma 2>,  <&adma 2>,
+               <&adma 3>,  <&adma 3>,
+               <&adma 4>,  <&adma 4>,
+               <&adma 5>,  <&adma 5>,
+               <&adma 6>,  <&adma 6>,
+               <&adma 7>,  <&adma 7>,
+               <&adma 8>,  <&adma 8>,
+               <&adma 9>,  <&adma 9>,
+               <&adma 10>, <&adma 10>;
+            dma-names = "rx1",  "tx1",
+                    "rx2",  "tx2",
+                    "rx3",  "tx3",
+                    "rx4",  "tx4",
+                    "rx5",  "tx5",
+                    "rx6",  "tx6",
+                    "rx7",  "tx7",
+                    "rx8",  "tx8",
+                    "rx9",  "tx9",
+                    "rx10", "tx10";
+        };
+
+        i2s@702d1000 {
+            compatible = "nvidia,tegra210-i2s";
+            reg = <0x702d1000 0x100>;
+            clocks = <&tegra_car TEGRA210_CLK_I2S0>;
+            clock-names = "i2s";
+            assigned-clocks = <&tegra_car TEGRA210_CLK_I2S0>;
+            assigned-clock-parents = <&tegra_car TEGRA210_CLK_PLL_A_OUT0>;
+            assigned-clock-rates = <1536000>;
+            sound-name-prefix = "I2S1";
+        };
+
+        dmic@702d4000 {
+            compatible = "nvidia,tegra210-dmic";
+            reg = <0x702d4000 0x100>;
+            clocks = <&tegra_car TEGRA210_CLK_DMIC1>;
+            clock-names = "dmic";
+            assigned-clocks = <&tegra_car TEGRA210_CLK_DMIC1>;
+            assigned-clock-parents = <&tegra_car TEGRA210_CLK_PLL_A_OUT0>;
+            assigned-clock-rates = <3072000>;
+            sound-name-prefix = "DMIC1";
+        };
+
+        // More child nodes to follow
+    };
+
+...

Documentation/devicetree/bindings/sound/nvidia,tegra210-dmic.yaml

+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/nvidia,tegra210-dmic.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Tegra210 DMIC Controller Device Tree Bindings
+
+description: |
+  The Digital MIC (DMIC) Controller is used to interface with Pulse
+  Density Modulation (PDM) input devices. It converts PDM signals to
+  Pulse Coded Modulation (PCM) signals. DMIC can be viewed as a PDM
+  receiver.
+
+maintainers:
+  - Jon Hunter <jonathanh@nvidia.com>
+  - Sameer Pujar <spujar@nvidia.com>
+
+properties:
+  $nodename:
+    pattern: "^dmic@[0-9a-f]*$"
+
+  compatible:
+    oneOf:
+      - const: nvidia,tegra210-dmic
+      - items:
+          - enum:
+              - nvidia,tegra194-dmic
+              - nvidia,tegra186-dmic
+          - const: nvidia,tegra210-dmic
+
+  reg:
+    maxItems: 1
+
+  clocks:
+    maxItems: 1
+
+  clock-names:
+    const: dmic
+
+  assigned-clocks:
+    maxItems: 1
+
+  assigned-clock-parents:
+    maxItems: 1
+
+  assigned-clock-rates:
+    maxItems: 1
+
+  sound-name-prefix:
+    pattern: "^DMIC[1-9]$"
+    allOf:
+      - $ref: /schemas/types.yaml#/definitions/string
+    description:
+      used as prefix for sink/source names of the component. Must be a
+      unique string among multiple instances of the same component.
+      The name can be "DMIC1" or "DMIC2" ... "DMICx", where x depends
+      on the maximum available instances on a Tegra SoC.
+
+required:
+  - compatible
+  - reg
+  - clocks
+  - clock-names
+  - assigned-clocks
+  - assigned-clock-parents
+
+examples:
+  - |
+    #include<dt-bindings/clock/tegra210-car.h>
+
+    dmic@702d4000 {
+        compatible = "nvidia,tegra210-dmic";
+        reg = <0x702d4000 0x100>;
+        clocks = <&tegra_car TEGRA210_CLK_DMIC1>;
+        clock-names = "dmic";
+        assigned-clocks = <&tegra_car TEGRA210_CLK_DMIC1>;
+        assigned-clock-parents = <&tegra_car TEGRA210_CLK_PLL_A_OUT0>;
+        assigned-clock-rates = <3072000>;
+        sound-name-prefix = "DMIC1";
+    };
+
+...

Documentation/devicetree/bindings/sound/nvidia,tegra210-i2s.yaml

+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/nvidia,tegra210-i2s.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Tegra210 I2S Controller Device Tree Bindings
+
+description: |
+  The Inter-IC Sound (I2S) controller implements full-duplex,
+  bi-directional and single direction point-to-point serial
+  interfaces. It can interface with I2S compatible devices.
+  I2S controller can operate both in master and slave mode.
+
+maintainers:
+  - Jon Hunter <jonathanh@nvidia.com>
+  - Sameer Pujar <spujar@nvidia.com>
+
+properties:
+  $nodename:
+    pattern: "^i2s@[0-9a-f]*$"
+
+  compatible:
+    oneOf:
+      - const: nvidia,tegra210-i2s
+      - items:
+          - enum:
+              - nvidia,tegra194-i2s
+              - nvidia,tegra186-i2s
+          - const: nvidia,tegra210-i2s
+
+  reg:
+    maxItems: 1
+
+  clocks:
+    minItems: 1
+    maxItems: 2
+    items:
+      - description: I2S bit clock
+      - description:
+          Sync input clock, which can act as clock source to other I/O
+          modules in AHUB. The Tegra I2S driver sets this clock rate as
+          per bit clock rate. I/O module which wants to use this clock
+          as source, can mention this clock as parent in the DT bindings.
+          This is an optional clock entry, since it is only required when
+          some other I/O wants to reference from a particular I2Sx
+          instance.
+
+  clock-names:
+    minItems: 1
+    maxItems: 2
+    items:
+      - const: i2s
+      - const: sync_input
+
+  assigned-clocks:
+    minItems: 1
+    maxItems: 2
+
+  assigned-clock-parents:
+    minItems: 1
+    maxItems: 2
+
+  assigned-clock-rates:
+    minItems: 1
+    maxItems: 2
+
+  sound-name-prefix:
+    pattern: "^I2S[1-9]$"
+    allOf:
+      - $ref: /schemas/types.yaml#/definitions/string
+    description:
+      Used as prefix for sink/source names of the component. Must be a
+      unique string among multiple instances of the same component.
+      The name can be "I2S1" or "I2S2" ... "I2Sx", where x depends
+      on the maximum available instances on a Tegra SoC.
+
+required:
+  - compatible
+  - reg
+  - clocks
+  - clock-names
+  - assigned-clocks
+  - assigned-clock-parents
+
+examples:
+  - |
+    #include<dt-bindings/clock/tegra210-car.h>
+
+    i2s@702d1000 {
+        compatible = "nvidia,tegra210-i2s";
+        reg = <0x702d1000 0x100>;
+        clocks = <&tegra_car TEGRA210_CLK_I2S0>;
+        clock-names = "i2s";
+        assigned-clocks = <&tegra_car TEGRA210_CLK_I2S0>;
+        assigned-clock-parents = <&tegra_car TEGRA210_CLK_PLL_A_OUT0>;
+        assigned-clock-rates = <1536000>;
+        sound-name-prefix = "I2S1";
+    };
+
+...

Documentation/devicetree/bindings/sound/qcom,q6asm.txt

@@ -39,9 +39,9 @@ configuration of each dai. Must contain the following properties.
 	Usage: Required for Compress offload dais
 	Value type: <u32>
 	Definition: Specifies the direction of the dai stream
-			0 for both tx and rx
-			1 for only tx (Capture/Encode)
-			2 for only rx (Playback/Decode)
+			Q6ASM_DAI_TX_RX (0) for both tx and rx
+			Q6ASM_DAI_TX (1) for only tx (Capture/Encode)
+			Q6ASM_DAI_RX (2) for only rx (Playback/Decode)
 
 - is-compress-dai:
 	Usage: Required for Compress offload dais
@@ -50,6 +50,7 @@ configuration of each dai. Must contain the following properties.
 
 
 = EXAMPLE
+#include <dt-bindings/sound/qcom,q6asm.h>
 
 apr-service@7 {
 	compatible = "qcom,q6asm";
@@ -62,7 +63,7 @@ apr-service@7 {
 
 		dai@0 {
 			reg = <0>;
-			direction = <2>;
+			direction = <Q6ASM_DAI_RX>;
 			is-compress-dai;
 		};
 	};

Documentation/devicetree/bindings/sound/renesas,fsi.yaml

@@ -43,30 +43,19 @@ properties:
   '#sound-dai-cells':
     const: 1
 
-  fsia,spdif-connection:
+patternProperties:
+  "^fsi(a|b),spdif-connection$":
     $ref: /schemas/types.yaml#/definitions/flag
     description: FSI is connected by S/PDIF
 
-  fsia,stream-mode-support:
+  "^fsi(a|b),stream-mode-support$":
     $ref: /schemas/types.yaml#/definitions/flag
     description: FSI supports 16bit stream mode
 
-  fsia,use-internal-clock:
+  "^fsi(a|b),use-internal-clock$":
     $ref: /schemas/types.yaml#/definitions/flag
     description: FSI uses internal clock when master mode
 
-  fsib,spdif-connection:
-    $ref: /schemas/types.yaml#/definitions/flag
-    description: same as fsia
-
-  fsib,stream-mode-support:
-    $ref: /schemas/types.yaml#/definitions/flag
-    description: same as fsia
-
-  fsib,use-internal-clock:
-    $ref: /schemas/types.yaml#/definitions/flag
-    description: same as fsia
-
 required:
   - compatible
   - reg

Documentation/devicetree/bindings/sound/renesas,rsnd.txt

@@ -271,6 +271,7 @@ Required properties:
 				    - "renesas,rcar_sound-r8a774a1" (RZ/G2M)
 				    - "renesas,rcar_sound-r8a774b1" (RZ/G2N)
 				    - "renesas,rcar_sound-r8a774c0" (RZ/G2E)
+				    - "renesas,rcar_sound-r8a774e1" (RZ/G2H)
 				    - "renesas,rcar_sound-r8a7778" (R-Car M1A)
 				    - "renesas,rcar_sound-r8a7779" (R-Car H1)
 				    - "renesas,rcar_sound-r8a7790" (R-Car H2)

Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt

-* Rockchip Rk3328 internal codec
-
-Required properties:
-
-- compatible: "rockchip,rk3328-codec"
-- reg: physical base address of the controller and length of memory mapped
-  region.
-- rockchip,grf: the phandle of the syscon node for GRF register.
-- clocks: a list of phandle + clock-specifer pairs, one for each entry in clock-names.
-- clock-names: should be "pclk".
-- spk-depop-time-ms: speak depop time msec.
-
-Optional properties:
-
-- mute-gpios: GPIO specifier for external line driver control (typically the
-              dedicated GPIO_MUTE pin)
-
-Example for rk3328 internal codec:
-
-codec: codec@ff410000 {
-	compatible = "rockchip,rk3328-codec";
-	reg = <0x0 0xff410000 0x0 0x1000>;
-	rockchip,grf = <&grf>;
-	clocks = <&cru PCLK_ACODEC>;
-	clock-names = "pclk";
-	mute-gpios = <&grf_gpio 0 GPIO_ACTIVE_LOW>;
-	spk-depop-time-ms = 100;
-};

Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.yaml

+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/rockchip,rk3328-codec.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Rockchip rk3328 internal codec
+
+maintainers:
+  - Heiko Stuebner <heiko@sntech.de>
+
+properties:
+  compatible:
+    const: rockchip,rk3328-codec
+
+  reg:
+    maxItems: 1
+
+  clocks:
+    items:
+      - description: clock for audio codec
+      - description: clock for I2S master clock
+
+  clock-names:
+    items:
+      - const: pclk
+      - const: mclk
+
+  rockchip,grf:
+    $ref: /schemas/types.yaml#/definitions/phandle
+    description:
+      The phandle of the syscon node for the GRF register.
+
+  spk-depop-time-ms:
+    default: 200
+    description:
+      Speaker depop time in msec.
+
+  mute-gpios:
+    maxItems: 1
+    description:
+      GPIO specifier for external line driver control (typically the
+      dedicated GPIO_MUTE pin)
+
+  "#sound-dai-cells":
+    const: 0
+
+required:
+  - compatible
+  - reg
+  - clocks
+  - clock-names
+  - rockchip,grf
+  - "#sound-dai-cells"
+
+examples:
+  - |
+    #include <dt-bindings/gpio/gpio.h>
+    #include <dt-bindings/clock/rk3328-cru.h>
+    codec: codec@ff410000 {
+      compatible = "rockchip,rk3328-codec";
+      reg = <0xff410000 0x1000>;
+      clocks = <&cru PCLK_ACODECPHY>, <&cru SCLK_I2S1>;
+      clock-names = "pclk", "mclk";
+      rockchip,grf = <&grf>;
+      mute-gpios = <&grf_gpio 0 GPIO_ACTIVE_LOW>;
+      spk-depop-time-ms = <100>;
+      #sound-dai-cells = <0>;
+    };

Documentation/devicetree/bindings/sound/rohm,bd28623.txt

-ROHM BD28623MUV Class D speaker amplifier for digital input
-
-This codec does not have any control buses such as I2C, it detect format and
-rate of I2S signal automatically. It has two signals that can be connected
-to GPIOs: reset and mute.
-
-Required properties:
-- compatible      : should be "rohm,bd28623"
-- #sound-dai-cells: should be 0.
-- VCCA-supply     : regulator phandle for the VCCA supply
-- VCCP1-supply    : regulator phandle for the VCCP1 supply
-- VCCP2-supply    : regulator phandle for the VCCP2 supply
-
-Optional properties:
-- reset-gpios     : GPIO specifier for the active low reset line
-- mute-gpios      : GPIO specifier for the active low mute line
-
-Example:
-
-	codec {
-		compatible = "rohm,bd28623";
-		#sound-dai-cells = <0>;
-
-		VCCA-supply = <&vcc_reg>;
-		VCCP1-supply = <&vcc_reg>;
-		VCCP2-supply = <&vcc_reg>;
-		reset-gpios = <&gpio 0 GPIO_ACTIVE_LOW>;
-		mute-gpios = <&gpio 1 GPIO_ACTIVE_LOW>;
-	};

Documentation/devicetree/bindings/sound/rohm,bd28623.yaml

+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/rohm,bd28623.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: ROHM BD28623MUV Class D speaker amplifier for digital input
+
+description:
+  This codec does not have any control buses such as I2C, it detect
+  format and rate of I2S signal automatically. It has two signals
+  that can be connected to GPIOs reset and mute.
+
+maintainers:
+  - Katsuhiro Suzuki <katsuhiro@katsuster.net>
+
+properties:
+  compatible:
+    const: rohm,bd28623
+
+  "#sound-dai-cells":
+    const: 0
+
+  VCCA-supply:
+    description:
+      regulator phandle for the VCCA (for analog) power supply
+
+  VCCP1-supply:
+    description:
+      regulator phandle for the VCCP1 (for ch1) power supply
+
+  VCCP2-supply:
+    description:
+      regulator phandle for the VCCP2 (for ch2) power supply
+
+  reset-gpios:
+    maxItems: 1
+    description:
+      GPIO specifier for the active low reset line
+
+  mute-gpios:
+    maxItems: 1
+    description:
+      GPIO specifier for the active low mute line
+
+required:
+  - compatible
+  - VCCA-supply
+  - VCCP1-supply
+  - VCCP2-supply
+  - "#sound-dai-cells"
+
+additionalProperties: false
+
+examples:
+  - |
+    #include <dt-bindings/gpio/gpio.h>
+    codec {
+      compatible = "rohm,bd28623";
+      #sound-dai-cells = <0>;
+
+      VCCA-supply = <&vcc_reg>;
+      VCCP1-supply = <&vcc_reg>;
+      VCCP2-supply = <&vcc_reg>;
+      reset-gpios = <&gpio 0 GPIO_ACTIVE_LOW>;
+      mute-gpios = <&gpio 1 GPIO_ACTIVE_LOW>;
+    };

Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml

+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/samsung,aries-wm8994.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Samsung Aries audio complex with WM8994 codec
+
+maintainers:
+  - Jonathan Bakker <xc-racer2@live.ca>
+
+properties:
+  compatible:
+    oneOf:
+      - const: samsung,aries-wm8994
+        description: With FM radio and modem master
+
+      - const: samsung,fascinate4g-wm8994
+        description: Without FM radio and modem slave
+
+  model:
+    $ref: /schemas/types.yaml#/definitions/string
+    description: The user-visible name of this sound complex.
+
+  cpu:
+    type: object
+    properties:
+      sound-dai:
+        minItems: 2
+        maxItems: 2
+        $ref: /schemas/types.yaml#/definitions/phandle-array
+        description: |
+          phandles to the I2S controller and bluetooth codec,
+          in that order
+
+  codec:
+    type: object
+    properties:
+      sound-dai:
+        $ref: /schemas/types.yaml#/definitions/phandle-array
+        description: phandle to the WM8994 CODEC
+
+  samsung,audio-routing:
+    $ref: /schemas/types.yaml#/definitions/non-unique-string-array
+    description: |
+      List of the connections between audio
+      components;  each entry is a pair of strings, the first being the
+      connection's sink, the second being the connection's source;
+      valid names for sources and sinks are the WM8994's pins (as
+      documented in its binding), and the jacks on the board -
+      For samsung,aries-wm8994: HP, SPK, RCV, LINE, Main Mic, Headset Mic,
+      or FM In
+      For samsung,fascinate4g-wm8994: HP, SPK, RCV, LINE, Main Mic,
+      or HeadsetMic
+
+  extcon:
+    description: Extcon phandle for dock detection
+
+  main-micbias-supply:
+    description: Supply for the micbias on the main mic
+
+  headset-micbias-supply:
+    description: Supply for the micbias on the headset mic
+
+  earpath-sel-gpios:
+    description: GPIO for switching between tv-out and mic paths
+
+  headset-detect-gpios:
+    description: GPIO for detection of headset insertion
+
+  headset-key-gpios:
+    description: GPIO for detection of headset key press
+
+  io-channels:
+    maxItems: 1
+    description: IO channel to read micbias voltage for headset detection
+
+  io-channel-names:
+    const: headset-detect
+
+required:
+  - compatible
+  - model
+  - cpu
+  - codec
+  - samsung,audio-routing
+  - extcon
+  - main-micbias-supply
+  - headset-micbias-supply
+  - earpath-sel-gpios
+  - headset-detect-gpios
+  - headset-key-gpios
+
+additionalProperties: false
+
+examples:
+  - |
+    #include <dt-bindings/gpio/gpio.h>
+
+    sound {
+        compatible = "samsung,fascinate4g-wm8994";
+
+        model = "Fascinate4G";
+
+        extcon = <&fsa9480>;
+
+        main-micbias-supply = <&main_micbias_reg>;
+        headset-micbias-supply = <&headset_micbias_reg>;
+
+        earpath-sel-gpios = <&gpj2 6 GPIO_ACTIVE_HIGH>;
+
+        io-channels = <&adc 3>;
+        io-channel-names = "headset-detect";
+        headset-detect-gpios = <&gph0 6 GPIO_ACTIVE_HIGH>;
+        headset-key-gpios = <&gph3 6 GPIO_ACTIVE_HIGH>;
+
+        samsung,audio-routing =
+            "HP", "HPOUT1L",
+            "HP", "HPOUT1R",
+
+            "SPK", "SPKOUTLN",
+            "SPK", "SPKOUTLP",
+
+            "RCV", "HPOUT2N",
+            "RCV", "HPOUT2P",
+
+            "LINE", "LINEOUT2N",
+            "LINE", "LINEOUT2P",
+
+            "IN1LP", "Main Mic",
+            "IN1LN", "Main Mic",
+
+            "IN1RP", "Headset Mic",
+            "IN1RN", "Headset Mic";
+
+        pinctrl-names = "default";
+        pinctrl-0 = <&headset_det &earpath_sel>;
+
+        cpu {
+            sound-dai = <&i2s0>, <&bt_codec>;
+        };
+
+        codec {
+            sound-dai = <&wm8994>;
+        };
+    };
+

Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml

+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/samsung,midas-audio.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Samsung Midas audio complex with WM1811 codec
+
+maintainers:
+  - Sylwester Nawrocki <s.nawrocki@samsung.com>
+
+properties:
+  compatible:
+    const: samsung,midas-audio
+
+  model:
+    $ref: /schemas/types.yaml#/definitions/string
+    description: The user-visible name of this sound complex.
+
+  cpu:
+    type: object
+    properties:
+      sound-dai:
+        $ref: /schemas/types.yaml#/definitions/phandle
+        description: phandle to the I2S controller
+    required:
+      - sound-dai
+
+  codec:
+    type: object
+    properties:
+      sound-dai:
+        $ref: /schemas/types.yaml#/definitions/phandle
+        description: phandle to the WM1811 CODEC
+    required:
+      - sound-dai
+
+  samsung,audio-routing:
+    $ref: /schemas/types.yaml#/definitions/non-unique-string-array
+    description: |
+      List of the connections between audio components; each entry is
+      a pair of strings, the first being the connection's sink, the second
+      being the connection's source; valid names for sources and sinks are
+      the WM1811's pins (as documented in its binding), and the jacks
+      on the board: HP, SPK, Main Mic, Sub Mic, Headset Mic.
+
+  mic-bias-supply:
+    description: Supply for the micbias on the Main microphone
+
+  submic-bias-supply:
+    description: Supply for the micbias on the Sub microphone
+
+  fm-sel-gpios:
+    description: GPIO pin for FM selection
+
+  lineout-sel-gpios:
+    description: GPIO pin for line out selection
+
+required:
+  - compatible
+  - model
+  - cpu
+  - codec
+  - samsung,audio-routing
+  - mic-bias-supply
+  - submic-bias-supply
+
+additionalProperties: false
+
+examples:
+  - |
+    #include <dt-bindings/gpio/gpio.h>
+
+    sound {
+        compatible = "samsung,midas-audio";
+        model = "Midas";
+
+        fm-sel-gpios = <&gpaa0 3 GPIO_ACTIVE_HIGH>;
+
+        mic-bias-supply = <&mic_bias_reg>;
+        submic-bias-supply = <&submic_bias_reg>;
+
+        samsung,audio-routing =
+                "HP", "HPOUT1L",
+                "HP", "HPOUT1R",
+
+                "SPK", "SPKOUTLN",
+                "SPK", "SPKOUTLP",
+                "SPK", "SPKOUTRN",
+                "SPK", "SPKOUTRP",
+
+                "RCV", "HPOUT2N",
+                "RCV", "HPOUT2P",
+
+                "IN1LP", "Main Mic",
+                "IN1LN", "Main Mic",
+                "IN1RP", "Sub Mic",
+                "IN1LP", "Sub Mic";
+
+        cpu {
+            sound-dai = <&i2s0>;
+        };
+
+        codec {
+            sound-dai = <&wm1811>;
+        };
+
+    };

Documentation/devicetree/bindings/sound/sgtl5000.txt

-* Freescale SGTL5000 Stereo Codec
-
-Required properties:
-- compatible : "fsl,sgtl5000".
-
-- reg : the I2C address of the device
-
-- #sound-dai-cells: must be equal to 0
-
-- clocks : the clock provider of SYS_MCLK
-
-- VDDA-supply : the regulator provider of VDDA
-
-- VDDIO-supply: the regulator provider of VDDIO
-
-Optional properties:
-
-- VDDD-supply : the regulator provider of VDDD
-
-- micbias-resistor-k-ohms : the bias resistor to be used in kOhms
-	The resistor can take values of 2k, 4k or 8k.
-	If set to 0 it will be off.
-	If this node is not mentioned or if the value is unknown, then
-	micbias	resistor is set to 4K.
-
-- micbias-voltage-m-volts : the bias voltage to be used in mVolts
-	The voltage can take values from 1.25V to 3V by 250mV steps
-	If this node is not mentioned or the value is unknown, then
-	the value is set to 1.25V.
-
-- lrclk-strength: the LRCLK pad strength. Possible values are:
-0, 1, 2 and 3 as per the table below:
-
-VDDIO		1.8V		2.5V		3.3V
-0 = 		Disable
-1 =		1.66 mA		2.87 mA		4.02  mA
-2 =		3.33 mA		5.74 mA		8.03  mA
-3 =		4.99 mA		8.61 mA		12.05 mA
-
-- sclk-strength: the SCLK pad strength. Possible values are:
-0, 1, 2 and 3 as per the table below:
-
-VDDIO		1.8V		2.5V		3.3V
-0 = 		Disable
-1 =		1.66 mA		2.87 mA		4.02  mA
-2 =		3.33 mA		5.74 mA		8.03  mA
-3 =		4.99 mA		8.61 mA		12.05 mA
-
-Example:
-
-sgtl5000: codec@a {
-	compatible = "fsl,sgtl5000";
-	reg = <0x0a>;
-	#sound-dai-cells = <0>;
-	clocks = <&clks 150>;
-	micbias-resistor-k-ohms = <2>;
-	micbias-voltage-m-volts = <2250>;
-	VDDA-supply = <&reg_3p3v>;
-	VDDIO-supply = <&reg_3p3v>;
-};

Documentation/devicetree/bindings/sound/sgtl5000.yaml

+# SPDX-License-Identifier: GPL-2.0-only
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/sgtl5000.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Freescale SGTL5000 Stereo Codec
+
+maintainers:
+  - Fabio Estevam <festevam@gmail.com>
+
+properties:
+  compatible:
+    const: fsl,sgtl5000
+
+  reg:
+    maxItems: 1
+
+  "#sound-dai-cells":
+    const: 0
+
+  clocks:
+    items:
+      - description: the clock provider of SYS_MCLK
+
+  VDDA-supply:
+    description: the regulator provider of VDDA
+
+  VDDIO-supply:
+    description: the regulator provider of VDDIO
+
+  VDDD-supply:
+    description: the regulator provider of VDDD
+
+  micbias-resistor-k-ohms:
+    description: The bias resistor to be used in kOhms. The resistor can take
+      values of 2k, 4k or 8k. If set to 0 it will be off. If this node is not
+      mentioned or if the value is unknown, then micbias resistor is set to
+      4k.
+    $ref: "/schemas/types.yaml#/definitions/uint32"
+    enum: [ 0, 2, 4, 8 ]
+
+  micbias-voltage-m-volts:
+    description: The bias voltage to be used in mVolts. The voltage can take
+      values from 1.25V to 3V by 250mV steps. If this node is not mentioned
+      or the value is unknown, then the value is set to 1.25V.
+    $ref: "/schemas/types.yaml#/definitions/uint32"
+    enum: [ 1250, 1500, 1750, 2000, 2250, 2500, 2750, 3000 ]
+
+  lrclk-strength:
+    description: |
+      The LRCLK pad strength. Possible values are: 0, 1, 2 and 3 as per the
+      table below:
+
+        VDDIO		1.8V		2.5V		3.3V
+        0 =		Disable
+        1 =		1.66 mA		2.87 mA		4.02  mA
+        2 =		3.33 mA		5.74 mA		8.03  mA
+        3 =		4.99 mA		8.61 mA		12.05 mA
+    $ref: "/schemas/types.yaml#/definitions/uint32"
+    enum: [ 0, 1, 2, 3 ]
+
+  sclk-strength:
+    description: |
+      The SCLK pad strength. Possible values are: 0, 1, 2 and 3 as per the
+      table below:
+
+        VDDIO		1.8V		2.5V		3.3V
+        0 =		Disable
+        1 =		1.66 mA		2.87 mA		4.02  mA
+        2 =		3.33 mA		5.74 mA		8.03  mA
+        3 =		4.99 mA		8.61 mA		12.05 mA
+    $ref: "/schemas/types.yaml#/definitions/uint32"
+    enum: [ 0, 1, 2, 3 ]
+
+required:
+  - compatible
+  - reg
+  - "#sound-dai-cells"
+  - clocks
+  - VDDA-supply
+  - VDDIO-supply
+
+additionalProperties: false
+
+examples:
+  - |
+    i2c {
+        #address-cells = <1>;
+        #size-cells = <0>;
+
+        codec@a {
+            compatible = "fsl,sgtl5000";
+            reg = <0x0a>;
+            #sound-dai-cells = <0>;
+            clocks = <&clks 150>;
+            micbias-resistor-k-ohms = <2>;
+            micbias-voltage-m-volts = <2250>;
+            VDDA-supply = <&reg_3p3v>;
+            VDDIO-supply = <&reg_3p3v>;
+        };
+    };
+...

Documentation/devicetree/bindings/sound/socionext,uniphier-aio.yaml

+# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/socionext,uniphier-aio.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: UniPhier AIO audio system
+
+maintainers:
+  - <alsa-devel@alsa-project.org>
+
+properties:
+  compatible:
+    enum:
+      - socionext,uniphier-ld11-aio
+      - socionext,uniphier-ld20-aio
+      - socionext,uniphier-pxs2-aio
+
+  reg:
+    maxItems: 1
+
+  interrupts:
+    maxItems: 1
+
+  clock-names:
+    const: aio
+
+  clocks:
+    maxItems: 1
+
+  reset-names:
+    const: aio
+
+  resets:
+    maxItems: 1
+
+  socionext,syscon:
+    description: |
+      Specifies a phandle to soc-glue, which is used for changing mode of S/PDIF
+      signal pin to output from Hi-Z. This property is optional if you use I2S
+      signal pins only.
+    $ref: "/schemas/types.yaml#/definitions/phandle"
+
+  "#sound-dai-cells":
+    const: 1
+
+patternProperties:
+  "^port@[0-9]$":
+    type: object
+    properties:
+      endpoint: true
+    required:
+      - endpoint
+
+additionalProperties: false
+
+required:
+  - compatible
+  - reg
+  - interrupts
+  - clock-names
+  - clocks
+  - reset-names
+  - resets
+  - "#sound-dai-cells"
+
+examples:
+  - |
+    audio@56000000 {
+        compatible = "socionext,uniphier-ld20-aio";
+        reg = <0x56000000 0x80000>;
+        interrupts = <0 144 4>;
+        pinctrl-names = "default";
+        pinctrl-0 = <&pinctrl_aout>;
+        clock-names = "aio";
+        clocks = <&sys_clk 40>;
+        reset-names = "aio";
+        resets = <&sys_rst 40>;
+        #sound-dai-cells = <1>;
+        socionext,syscon = <&soc_glue>;
+    };

Documentation/devicetree/bindings/sound/socionext,uniphier-evea.yaml

+# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/socionext,uniphier-evea.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: UniPhier EVEA SoC-internal sound codec
+
+maintainers:
+  - <alsa-devel@alsa-project.org>
+
+properties:
+  compatible:
+    const: socionext,uniphier-evea
+
+  reg:
+    maxItems: 1
+
+  clock-names:
+    items:
+      - const: evea
+      - const: exiv
+
+  clocks:
+    minItems: 2
+    maxItems: 2
+
+  reset-names:
+    items:
+      - const: evea
+      - const: exiv
+      - const: adamv
+
+  resets:
+    minItems: 3
+    maxItems: 3
+
+  "#sound-dai-cells":
+    const: 1
+
+patternProperties:
+  "^port@[0-9]$":
+    type: object
+    properties:
+      endpoint: true
+    required:
+      - endpoint
+
+additionalProperties: false
+
+required:
+  - compatible
+  - reg
+  - clock-names
+  - clocks
+  - reset-names
+  - resets
+  - "#sound-dai-cells"
+
+examples:
+  - |
+    codec@57900000 {
+        compatible = "socionext,uniphier-evea";
+        reg = <0x57900000 0x1000>;
+        clock-names = "evea", "exiv";
+        clocks = <&sys_clk 41>, <&sys_clk 42>;
+        reset-names = "evea", "exiv", "adamv";
+        resets = <&sys_rst 41>, <&sys_rst 42>, <&adamv_rst 0>;
+        #sound-dai-cells = <1>;
+    };

Documentation/devicetree/bindings/sound/tas2552.txt

@@ -33,4 +33,4 @@ tas2552: tas2552@41 {
 };
 
 For more product information please see the link below:
-http://www.ti.com/product/TAS2552
+https://www.ti.com/product/TAS2552

Documentation/devicetree/bindings/sound/tas2562.txt

@@ -11,12 +11,14 @@ Required properties:
  - compatible:	   - Should contain "ti,tas2562", "ti,tas2563".
  - reg:		   - The i2c address. Should be 0x4c, 0x4d, 0x4e or 0x4f.
  - ti,imon-slot-no:- TDM TX current sense time slot.
+ - ti,vmon-slot-no:- TDM TX voltage sense time slot. This slot must always be
+		     greater then ti,imon-slot-no.
 
 Optional properties:
 - interrupt-parent: phandle to the interrupt controller which provides
                     the interrupt.
 - interrupts: (GPIO) interrupt to which the chip is connected.
-- shut-down: GPIO used to control the state of the device.
+- shut-down-gpio: GPIO used to control the state of the device.
 
 Examples:
 tas2562@4c {
@@ -28,7 +30,8 @@ tas2562@4c {
         interrupt-parent = <&gpio1>;
         interrupts = <14>;
 
-	shut-down = <&gpio1 15 0>;
+	shut-down-gpio = <&gpio1 15 0>;
         ti,imon-slot-no = <0>;
+        ti,vmon-slot-no = <1>;
 };
 

Documentation/devicetree/bindings/sound/tas2562.yaml

+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+# Copyright (C) 2019 Texas Instruments Incorporated
+%YAML 1.2
+---
+$id: "http://devicetree.org/schemas/sound/tas2562.yaml#"
+$schema: "http://devicetree.org/meta-schemas/core.yaml#"
+
+title: Texas Instruments TAS2562 Smart PA
+
+maintainers:
+  - Dan Murphy <dmurphy@ti.com>
+
+description: |
+  The TAS2562 is a mono, digital input Class-D audio amplifier optimized for
+  efficiently driving high peak power into small loudspeakers.
+  Integrated speaker voltage and current sense provides for
+  real time monitoring of loudspeaker behavior.
+
+properties:
+  compatible:
+    enum:
+      - ti,tas2562
+      - ti,tas2563
+
+  reg:
+    maxItems: 1
+    description: |
+       I2C address of the device can be one of these 0x4c, 0x4d, 0x4e or 0x4f
+
+  shut-down-gpios:
+    description: GPIO used to control the state of the device.
+    deprecated: true
+
+  shutdown-gpios:
+    description: GPIO used to control the state of the device.
+
+  interrupts:
+    maxItems: 1
+
+  ti,imon-slot-no:
+    $ref: /schemas/types.yaml#/definitions/uint32
+    description: TDM TX current sense time slot.
+
+  '#sound-dai-cells':
+    const: 1
+
+required:
+  - compatible
+  - reg
+
+additionalProperties: false
+
+examples:
+  - |
+   #include <dt-bindings/gpio/gpio.h>
+   i2c0 {
+     #address-cells = <1>;
+     #size-cells = <0>;
+     codec: codec@4c {
+       compatible = "ti,tas2562";
+       reg = <0x4c>;
+       #sound-dai-cells = <1>;
+       interrupt-parent = <&gpio1>;
+       interrupts = <14>;
+       shutdown-gpios = <&gpio1 15 0>;
+       ti,imon-slot-no = <0>;
+     };
+   };
+

Documentation/devicetree/bindings/sound/tas2770.txt

-Texas Instruments TAS2770 Smart PA
-
-The TAS2770 is a mono, digital input Class-D audio amplifier optimized for
-efficiently driving high peak power into small loudspeakers.
-Integrated speaker voltage and current sense provides for
-real time monitoring of loudspeaker behavior.
-
-Required properties:
-
- - compatible:	   - Should contain "ti,tas2770".
- - reg:		       - The i2c address. Should contain <0x4c>, <0x4d>,<0x4e>, or <0x4f>.
- - #address-cells  - Should be <1>.
- - #size-cells     - Should be <0>.
- - ti,asi-format:  - Sets TDM RX capture edge. 0->Rising; 1->Falling.
- - ti,imon-slot-no:- TDM TX current sense time slot.
- - ti,vmon-slot-no:- TDM TX voltage sense time slot.
-
-Optional properties:
-
-- interrupt-parent: the phandle to the interrupt controller which provides
-                     the interrupt.
-- interrupts: interrupt specification for data-ready.
-
-Examples:
-
-    tas2770@4c {
-                compatible = "ti,tas2770";
-                reg = <0x4c>;
-                #address-cells = <1>;
-                #size-cells = <0>;
-                interrupt-parent = <&msm_gpio>;
-                interrupts = <97 0>;
-                ti,asi-format = <0>;
-                ti,imon-slot-no = <0>;
-                ti,vmon-slot-no = <2>;
-        };
-

Documentation/devicetree/bindings/sound/tas2770.yaml

+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+# Copyright (C) 2019-20 Texas Instruments Incorporated
+%YAML 1.2
+---
+$id: "http://devicetree.org/schemas/sound/tas2770.yaml#"
+$schema: "http://devicetree.org/meta-schemas/core.yaml#"
+
+title: Texas Instruments TAS2770 Smart PA
+
+maintainers:
+  - Shi Fu <shifu0704@thundersoft.com>
+
+description: |
+  The TAS2770 is a mono, digital input Class-D audio amplifier optimized for
+  efficiently driving high peak power into small loudspeakers.
+  Integrated speaker voltage and current sense provides for
+  real time monitoring of loudspeaker behavior.
+
+properties:
+  compatible:
+    enum:
+      - ti,tas2770
+
+  reg:
+    maxItems: 1
+    description: |
+       I2C address of the device can be one of these 0x4c, 0x4d, 0x4e or 0x4f
+
+  reset-gpio:
+    description: GPIO used to reset the device.
+
+  interrupts:
+    maxItems: 1
+
+  ti,imon-slot-no:
+    $ref: /schemas/types.yaml#/definitions/uint32
+    description: TDM TX current sense time slot.
+
+  ti,vmon-slot-no:
+    $ref: /schemas/types.yaml#/definitions/uint32
+    description: TDM TX voltage sense time slot.
+
+  ti,asi-format:
+    $ref: /schemas/types.yaml#/definitions/uint32
+    description: Sets TDM RX capture edge.
+    enum:
+          - 0 # Rising edge
+          - 1 # Falling edge
+
+  '#sound-dai-cells':
+    const: 1
+
+required:
+  - compatible
+  - reg
+
+additionalProperties: false
+
+examples:
+  - |
+   #include <dt-bindings/gpio/gpio.h>
+   i2c0 {
+     #address-cells = <1>;
+     #size-cells = <0>;
+     codec: codec@4c {
+       compatible = "ti,tas2770";
+       reg = <0x4c>;
+       #sound-dai-cells = <1>;
+       interrupt-parent = <&gpio1>;
+       interrupts = <14>;
+       reset-gpio = <&gpio1 15 0>;
+       ti,imon-slot-no = <0>;
+       ti,vmon-slot-no = <2>;
+     };
+   };
+

Documentation/devicetree/bindings/sound/tas5720.txt

@@ -4,9 +4,9 @@ The TAS5720 serial control bus communicates through the I2C protocol only. The
 serial bus is also used for periodic codec fault checking/reporting during
 audio playback. For more product information please see the links below:
 
-http://www.ti.com/product/TAS5720L
-http://www.ti.com/product/TAS5720M
-http://www.ti.com/product/TAS5722L
+https://www.ti.com/product/TAS5720L
+https://www.ti.com/product/TAS5720M
+https://www.ti.com/product/TAS5722L
 
 Required properties:
 

Documentation/devicetree/bindings/sound/ti,j721e-cpb-audio.yaml

+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/ti,j721e-cpb-audio.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Texas Instruments J721e Common Processor Board Audio Support
+
+maintainers:
+  - Peter Ujfalusi <peter.ujfalusi@ti.com>
+
+description: |
+  The audio support on the board is using pcm3168a codec connected to McASP10
+  serializers in parallel setup.
+  The pcm3168a SCKI clock is sourced from j721e AUDIO_REFCLK2 pin.
+  In order to support 48KHz and 44.1KHz family of sampling rates the parent
+  clock for AUDIO_REFCLK2 needs to be changed between PLL4 (for 48KHz) and
+  PLL15 (for 44.1KHz). The same PLLs are used for McASP10's AUXCLK clock via
+  different HSDIVIDER.
+
+  Clocking setup for 48KHz family:
+  PLL4 ---> PLL4_HSDIV0 ---> MCASP10_AUXCLK ---> McASP10.auxclk
+        |-> PLL4_HSDIV2 ---> AUDIO_REFCLK2  ---> pcm3168a.SCKI
+
+  Clocking setup for 44.1KHz family:
+  PLL15 ---> PLL15_HSDIV0 ---> MCASP10_AUXCLK ---> McASP10.auxclk
+         |-> PLL15_HSDIV2 ---> AUDIO_REFCLK2  ---> pcm3168a.SCKI
+
+properties:
+  compatible:
+    items:
+      - const: ti,j721e-cpb-audio
+
+  model:
+    $ref: /schemas/types.yaml#/definitions/string
+    description: User specified audio sound card name
+
+  ti,cpb-mcasp:
+    description: phandle to McASP used on CPB
+    allOf:
+      - $ref: /schemas/types.yaml#/definitions/phandle
+
+  ti,cpb-codec:
+    description: phandle to the pcm3168a codec used on the CPB
+    allOf:
+      - $ref: /schemas/types.yaml#/definitions/phandle
+
+  clocks:
+    items:
+      - description: AUXCLK clock for McASP used by CPB audio
+      - description: Parent for CPB_McASP auxclk (for 48KHz)
+      - description: Parent for CPB_McASP auxclk (for 44.1KHz)
+      - description: SCKI clock for the pcm3168a codec on CPB
+      - description: Parent for CPB_SCKI clock (for 48KHz)
+      - description: Parent for CPB_SCKI clock (for 44.1KHz)
+
+  clock-names:
+    items:
+      - const: cpb-mcasp-auxclk
+      - const: cpb-mcasp-auxclk-48000
+      - const: cpb-mcasp-auxclk-44100
+      - const: cpb-codec-scki
+      - const: cpb-codec-scki-48000
+      - const: cpb-codec-scki-44100
+
+required:
+  - compatible
+  - model
+  - ti,cpb-mcasp
+  - ti,cpb-codec
+  - clocks
+  - clock-names
+
+additionalProperties: false
+
+examples:
+  - |+
+    sound {
+        compatible = "ti,j721e-cpb-audio";
+        model = "j721e-cpb";
+
+        status = "okay";
+
+        ti,cpb-mcasp = <&mcasp10>;
+        ti,cpb-codec = <&pcm3168a_1>;
+
+        clocks = <&k3_clks 184 1>,
+                 <&k3_clks 184 2>, <&k3_clks 184 4>,
+                 <&k3_clks 157 371>,
+                 <&k3_clks 157 400>, <&k3_clks 157 401>;
+        clock-names = "cpb-mcasp-auxclk",
+                      "cpb-mcasp-auxclk-48000", "cpb-mcasp-auxclk-44100",
+                      "cpb-codec-scki",
+                      "cpb-codec-scki-48000", "cpb-codec-scki-44100";
+    };

Documentation/devicetree/bindings/sound/ti,j721e-cpb-ivi-audio.yaml

+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/ti,j721e-cpb-ivi-audio.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Texas Instruments J721e Common Processor Board Audio Support
+
+maintainers:
+  - Peter Ujfalusi <peter.ujfalusi@ti.com>
+
+description: |
+  The Infotainment board plugs into the Common Processor Board, the support of the
+  extension board is extending the CPB audio support, decribed in:
+  sound/ti,j721e-cpb-audio.txt
+
+  The audio support on the Infotainment Expansion Board consists of McASP0
+  connected to two pcm3168a codecs with dedicated set of serializers to each.
+  The SCKI for pcm3168a is sourced from j721e AUDIO_REFCLK0 pin.
+
+  In order to support 48KHz and 44.1KHz family of sampling rates the parent clock
+  for AUDIO_REFCLK0 needs to be changed between PLL4 (for 48KHz) and PLL15 (for
+  44.1KHz). The same PLLs are used for McASP0's AUXCLK clock via different
+  HSDIVIDER.
+
+  Note: the same PLL4 and PLL15 is used by the audio support on the CPB!
+
+  Clocking setup for 48KHz family:
+  PLL4 ---> PLL4_HSDIV0 ---> MCASP10_AUXCLK ---> McASP10.auxclk
+        |                |-> MCASP0_AUXCLK  ---> McASP0.auxclk
+        |
+        |-> PLL4_HSDIV2 ---> AUDIO_REFCLK2  ---> pcm3168a.SCKI
+                         |-> AUDIO_REFCLK0  ---> pcm3168a_a/b.SCKI
+
+  Clocking setup for 44.1KHz family:
+  PLL15 ---> PLL15_HSDIV0 ---> MCASP10_AUXCLK ---> McASP10.auxclk
+        |                  |-> MCASP0_AUXCLK  ---> McASP0.auxclk
+        |
+        |-> PLL15_HSDIV2 ---> AUDIO_REFCLK2   ---> pcm3168a.SCKI
+                          |-> AUDIO_REFCLK0   ---> pcm3168a_a/b.SCKI
+
+properties:
+  compatible:
+    items:
+      - const: ti,j721e-cpb-ivi-audio
+
+  model:
+    $ref: /schemas/types.yaml#/definitions/string
+    description: User specified audio sound card name
+
+  ti,cpb-mcasp:
+    description: phandle to McASP used on CPB
+    allOf:
+      - $ref: /schemas/types.yaml#/definitions/phandle
+
+  ti,cpb-codec:
+    description: phandle to the pcm3168a codec used on the CPB
+    allOf:
+      - $ref: /schemas/types.yaml#/definitions/phandle
+
+  ti,ivi-mcasp:
+    description: phandle to McASP used on IVI
+    allOf:
+      - $ref: /schemas/types.yaml#/definitions/phandle
+
+  ti,ivi-codec-a:
+    description: phandle to the pcm3168a-A codec on the expansion board
+    allOf:
+      - $ref: /schemas/types.yaml#/definitions/phandle
+
+  ti,ivi-codec-b:
+    description: phandle to the pcm3168a-B codec on the expansion board
+    allOf:
+      - $ref: /schemas/types.yaml#/definitions/phandle
+
+  clocks:
+    items:
+      - description: AUXCLK clock for McASP used by CPB audio
+      - description: Parent for CPB_McASP auxclk (for 48KHz)
+      - description: Parent for CPB_McASP auxclk (for 44.1KHz)
+      - description: SCKI clock for the pcm3168a codec on CPB
+      - description: Parent for CPB_SCKI clock (for 48KHz)
+      - description: Parent for CPB_SCKI clock (for 44.1KHz)
+      - description: AUXCLK clock for McASP used by IVI audio
+      - description: Parent for IVI_McASP auxclk (for 48KHz)
+      - description: Parent for IVI_McASP auxclk (for 44.1KHz)
+      - description: SCKI clock for the pcm3168a codec on IVI
+      - description: Parent for IVI_SCKI clock (for 48KHz)
+      - description: Parent for IVI_SCKI clock (for 44.1KHz)
+
+  clock-names:
+    items:
+      - const: cpb-mcasp-auxclk
+      - const: cpb-mcasp-auxclk-48000
+      - const: cpb-mcasp-auxclk-44100
+      - const: cpb-codec-scki
+      - const: cpb-codec-scki-48000
+      - const: cpb-codec-scki-44100
+      - const: ivi-mcasp-auxclk
+      - const: ivi-mcasp-auxclk-48000
+      - const: ivi-mcasp-auxclk-44100
+      - const: ivi-codec-scki
+      - const: ivi-codec-scki-48000
+      - const: ivi-codec-scki-44100
+
+required:
+  - compatible
+  - model
+  - ti,cpb-mcasp
+  - ti,cpb-codec
+  - ti,ivi-mcasp
+  - ti,ivi-codec-a
+  - ti,ivi-codec-b
+  - clocks
+  - clock-names
+
+additionalProperties: false
+
+examples:
+  - |+
+    sound {
+        compatible = "ti,j721e-cpb-ivi-audio";
+        model = "j721e-cpb-ivi";
+
+        status = "okay";
+
+        ti,cpb-mcasp = <&mcasp10>;
+        ti,cpb-codec = <&pcm3168a_1>;
+
+        ti,ivi-mcasp = <&mcasp0>;
+        ti,ivi-codec-a = <&pcm3168a_a>;
+        ti,ivi-codec-b = <&pcm3168a_b>;
+
+        clocks = <&k3_clks 184 1>,
+                 <&k3_clks 184 2>, <&k3_clks 184 4>,
+                 <&k3_clks 157 371>,
+                 <&k3_clks 157 400>, <&k3_clks 157 401>,
+                 <&k3_clks 174 1>,
+                 <&k3_clks 174 2>, <&k3_clks 174 4>,
+                 <&k3_clks 157 301>,
+                 <&k3_clks 157 330>, <&k3_clks 157 331>;
+        clock-names = "cpb-mcasp-auxclk",
+                      "cpb-mcasp-auxclk-48000", "cpb-mcasp-auxclk-44100",
+                      "cpb-codec-scki",
+                      "cpb-codec-scki-48000", "cpb-codec-scki-44100",
+                      "ivi-mcasp-auxclk",
+                      "ivi-mcasp-auxclk-48000", "ivi-mcasp-auxclk-44100",
+                      "ivi-codec-scki",
+                      "ivi-codec-scki-48000", "ivi-codec-scki-44100";
+    };

Documentation/devicetree/bindings/sound/ti,tas6424.txt

@@ -19,4 +19,4 @@ tas6424: tas6424@6a {
 };
 
 For more product information please see the link below:
-http://www.ti.com/product/TAS6424-Q1
+https://www.ti.com/product/TAS6424-Q1

Documentation/devicetree/bindings/sound/tlv320adcx140.yaml

@@ -18,9 +18,9 @@ description: |
   microphone bias or supply voltage generation.
 
   Specifications can be found at:
-    http://www.ti.com/lit/ds/symlink/tlv320adc3140.pdf
-    http://www.ti.com/lit/ds/symlink/tlv320adc5140.pdf
-    http://www.ti.com/lit/ds/symlink/tlv320adc6140.pdf
+    https://www.ti.com/lit/ds/symlink/tlv320adc3140.pdf
+    https://www.ti.com/lit/ds/symlink/tlv320adc5140.pdf
+    https://www.ti.com/lit/ds/symlink/tlv320adc6140.pdf
 
 properties:
   compatible:
@@ -108,6 +108,32 @@ properties:
       maximum: 7
     default: [0, 0, 0, 0]
 
+patternProperties:
+ '^ti,gpo-config-[1-4]$':
+    $ref: /schemas/types.yaml#/definitions/uint32-array
+    description: |
+       Defines the configuration and output driver for the general purpose
+       output pins (GPO).  These values are pairs, the first value is for the
+       configuration type and the second value is for the output drive type.
+       The array is defined as <GPO_CFG GPO_DRV>
+
+       GPO output configuration can be one of the following:
+
+       0 - (default) disabled
+       1 - GPOX is configured as a general-purpose output (GPO)
+       2 - GPOX is configured as a device interrupt output (IRQ)
+       3 - GPOX is configured as a secondary ASI output (SDOUT2)
+       4 - GPOX is configured as a PDM clock output (PDMCLK)
+
+       GPO output drive configuration for the GPO pins can be one of the following:
+
+       0d - (default) Hi-Z output
+       1d - Drive active low and active high
+       2d - Drive active low and weak high
+       3d - Drive active low and Hi-Z
+       4d - Drive weak low and active high
+       5d - Drive Hi-Z and active high
+
 required:
   - compatible
   - reg
@@ -124,6 +150,8 @@ examples:
         ti,mic-bias-source = <6>;
         ti,pdm-edge-select = <0 1 0 1>;
         ti,gpi-config = <4 5 6 7>;
+        ti,gpo-config-1 = <0 0>;
+        ti,gpo-config-2 = <0 0>;
         reset-gpios = <&gpio0 14 GPIO_ACTIVE_HIGH>;
       };
     };

Documentation/devicetree/bindings/sound/uniphier,aio.txt

-Socionext UniPhier SoC audio driver
-
-The Socionext UniPhier audio subsystem consists of I2S and S/PDIF blocks in
-the same register space.
-
-Required properties:
-- compatible      : should be one of the following:
-		    "socionext,uniphier-ld11-aio"
-		    "socionext,uniphier-ld20-aio"
-		    "socionext,uniphier-pxs2-aio"
-- reg             : offset and length of the register set for the device.
-- interrupts      : should contain I2S or S/PDIF interrupt.
-- pinctrl-names   : should be "default".
-- pinctrl-0       : defined I2S signal pins for an external codec chip.
-- clock-names     : should include following entries:
-                    "aio"
-- clocks          : a list of phandle, should contain an entry for each
-                    entry in clock-names.
-- reset-names     : should include following entries:
-                    "aio"
-- resets          : a list of phandle, should contain an entry for each
-                    entry in reset-names.
-- #sound-dai-cells: should be 1.
-
-Optional properties:
-- socionext,syscon: a phandle, should contain soc-glue.
-                    The soc-glue is used for changing mode of S/PDIF signal pin
-                    to Output from Hi-Z. This property is optional if you use
-                    I2S signal pins only.
-
-Example:
-	audio {
-		compatible = "socionext,uniphier-ld20-aio";
-		reg = <0x56000000 0x80000>;
-		interrupts = <0 144 4>;
-		pinctrl-names = "default";
-		pinctrl-0 = <&pinctrl_aout>;
-		clock-names = "aio";
-		clocks = <&sys_clk 40>;
-		reset-names = "aio";
-		resets = <&sys_rst 40>;
-		#sound-dai-cells = <1>;
-
-		socionext,syscon = <&sg>;
-	};

Documentation/devicetree/bindings/sound/uniphier,evea.txt

-Socionext EVEA - UniPhier SoC internal codec driver
-
-Required properties:
-- compatible      : should be "socionext,uniphier-evea".
-- reg             : offset and length of the register set for the device.
-- clock-names     : should include following entries:
-                    "evea", "exiv"
-- clocks          : a list of phandle, should contain an entry for each
-                    entries in clock-names.
-- reset-names     : should include following entries:
-                    "evea", "exiv", "adamv"
-- resets          : a list of phandle, should contain reset entries of
-                    reset-names.
-- #sound-dai-cells: should be 1.
-
-Example:
-
-	codec {
-		compatible = "socionext,uniphier-evea";
-		reg = <0x57900000 0x1000>;
-		clock-names = "evea", "exiv";
-		clocks = <&sys_clk 41>, <&sys_clk 42>;
-		reset-names = "evea", "exiv", "adamv";
-		resets = <&sys_rst 41>, <&sys_rst 42>, <&adamv_rst 0>;
-		#sound-dai-cells = <1>;
-	};

Documentation/devicetree/bindings/sound/wm8960.txt

@@ -21,6 +21,17 @@ Optional properties:
 	enabled and disabled together with HP_L and HP_R pins in response to jack
 	detect events.
 
+  - wlf,hp-cfg: A list of headphone jack detect configuration register values.
+		The list must be 3 entries long.
+		hp-cfg[0]: HPSEL[1:0] of R48 (Additional Control 4).
+		hp-cfg[1]: {HPSWEN:HPSWPOL} of R24 (Additional Control 2).
+		hp-cfg[2]: {TOCLKSEL:TOEN} of R23 (Additional Control 1).
+
+  - wlf,gpio-cfg: A list of GPIO configuration register values.
+		  The list must be 2 entries long.
+		  gpio-cfg[0]: ALRCGPIO of R9 (Audio interface)
+		  gpio-cfg[1]: {GPIOPOL:GPIOSEL[2:0]} of R48 (Additional Control 4).
+
 Example:
 
 wm8960: codec@1a {

Documentation/devicetree/bindings/sound/wm8994.txt

@@ -68,6 +68,29 @@ Optional properties:
   - wlf,csnaddr-pd : If present enable the internal pull-down resistor on
     the CS/ADDR pin.
 
+Pins on the device (for linking into audio routes):
+
+  * IN1LN
+  * IN1LP
+  * IN2LN
+  * IN2LP:VXRN
+  * IN1RN
+  * IN1RP
+  * IN2RN
+  * IN2RP:VXRP
+  * SPKOUTLP
+  * SPKOUTLN
+  * SPKOUTRP
+  * SPKOUTRN
+  * HPOUT1L
+  * HPOUT1R
+  * HPOUT2P
+  * HPOUT2N
+  * LINEOUT1P
+  * LINEOUT1N
+  * LINEOUT2P
+  * LINEOUT2N
+
 Example:
 
 wm8994: codec@1a {

Documentation/devicetree/bindings/trivial-devices.yaml

@@ -80,8 +80,6 @@ properties:
           - fsl,mpl3115
             # MPR121: Proximity Capacitive Touch Sensor Controller
           - fsl,mpr121
-            # SGTL5000: Ultra Low-Power Audio Codec
-          - fsl,sgtl5000
             # G751: Digital Temperature Sensor and Thermal Watchdog with Two-Wire Interface
           - gmt,g751
             # Infineon IR38064 Voltage Regulator

Documentation/devicetree/bindings/vendor-prefixes.yaml

@@ -20,7 +20,7 @@ patternProperties:
   "^(keypad|m25p|max8952|max8997|max8998|mpmc),.*": true
   "^(pinctrl-single|#pinctrl-single|PowerPC),.*": true
   "^(pl022|pxa-mmc|rcar_sound|rotary-encoder|s5m8767|sdhci),.*": true
-  "^(simple-audio-card|simple-graph-card|st-plgpio|st-spics|ts),.*": true
+  "^(simple-audio-card|st-plgpio|st-spics|ts),.*": true
 
   # Keep list in alphabetical order.
   "^abilis,.*":

Documentation/sound/kernel-api/alsa-driver-api.rst

@@ -99,7 +99,7 @@ ASoC Core API
 .. kernel-doc:: include/sound/soc.h
 .. kernel-doc:: sound/soc/soc-core.c
 .. kernel-doc:: sound/soc/soc-devres.c
-.. kernel-doc:: sound/soc/soc-io.c
+.. kernel-doc:: sound/soc/soc-component.c
 .. kernel-doc:: sound/soc/soc-pcm.c
 .. kernel-doc:: sound/soc/soc-ops.c
 .. kernel-doc:: sound/soc/soc-compress.c

Documentation/sound/soc/dai.rst

@@ -17,7 +17,7 @@ frame (FRAME) (usually 48kHz) is always driven by the controller. Each AC97
 frame is 21uS long and is divided into 13 time slots.
 
 The AC97 specification can be found at :
-http://www.intel.com/p/en_US/business/design
+https://www.intel.com/p/en_US/business/design
 
 
 I2S

arch/arm/boot/dts/motorola-mapphone-common.dtsi

@@ -140,13 +140,13 @@
 		compatible = "audio-graph-card";
 		label = "Droid 4 Audio";
 
-		simple-graph-card,widgets =
+		widgets =
 			"Speaker", "Earpiece",
 			"Speaker", "Loudspeaker",
 			"Headphone", "Headphone Jack",
 			"Microphone", "Internal Mic";
 
-		simple-graph-card,routing =
+		routing =
 			"Earpiece", "EP",
 			"Loudspeaker", "SPKR",
 			"Headphone Jack", "HSL",

drivers/gpu/drm/bridge/sii902x.c

@@ -672,8 +672,8 @@ static void sii902x_audio_shutdown(struct device *dev, void *data)
 	clk_disable_unprepare(sii902x->audio.mclk);
 }
 
-static int sii902x_audio_digital_mute(struct device *dev,
-				      void *data, bool enable)
+static int sii902x_audio_mute(struct device *dev, void *data,
+			      bool enable, int direction)
 {
 	struct sii902x *sii902x = dev_get_drvdata(dev);
 
@@ -724,9 +724,10 @@ static int sii902x_audio_get_dai_id(struct snd_soc_component *component,
 static const struct hdmi_codec_ops sii902x_audio_codec_ops = {
 	.hw_params = sii902x_audio_hw_params,
 	.audio_shutdown = sii902x_audio_shutdown,
-	.digital_mute = sii902x_audio_digital_mute,
+	.mute_stream = sii902x_audio_mute,
 	.get_eld = sii902x_audio_get_eld,
 	.get_dai_id = sii902x_audio_get_dai_id,
+	.no_capture_mute = 1,
 };
 
 static int sii902x_audio_codec_init(struct sii902x *sii902x,

drivers/gpu/drm/exynos/exynos_hdmi.c

@@ -1604,7 +1604,8 @@ static int hdmi_audio_hw_params(struct device *dev, void *data,
 	return 0;
 }
 
-static int hdmi_audio_digital_mute(struct device *dev, void *data, bool mute)
+static int hdmi_audio_mute(struct device *dev, void *data,
+			   bool mute, int direction)
 {
 	struct hdmi_context *hdata = dev_get_drvdata(dev);
 
@@ -1634,8 +1635,9 @@ static int hdmi_audio_get_eld(struct device *dev, void *data, uint8_t *buf,
 static const struct hdmi_codec_ops audio_codec_ops = {
 	.hw_params = hdmi_audio_hw_params,
 	.audio_shutdown = hdmi_audio_shutdown,
-	.digital_mute = hdmi_audio_digital_mute,
+	.mute_stream = hdmi_audio_mute,
 	.get_eld = hdmi_audio_get_eld,
+	.no_capture_mute = 1,
 };
 
 static int hdmi_register_audio_device(struct hdmi_context *hdata)

drivers/gpu/drm/i2c/tda998x_drv.c

@@ -1133,8 +1133,8 @@ static void tda998x_audio_shutdown(struct device *dev, void *data)
 	mutex_unlock(&priv->audio_mutex);
 }
 
-static int tda998x_audio_digital_mute(struct device *dev, void *data,
-				      bool enable)
+static int tda998x_audio_mute_stream(struct device *dev, void *data,
+				     bool enable, int direction)
 {
 	struct tda998x_priv *priv = dev_get_drvdata(dev);
 
@@ -1162,8 +1162,9 @@ static int tda998x_audio_get_eld(struct device *dev, void *data,
 static const struct hdmi_codec_ops audio_codec_ops = {
 	.hw_params = tda998x_audio_hw_params,
 	.audio_shutdown = tda998x_audio_shutdown,
-	.digital_mute = tda998x_audio_digital_mute,
+	.mute_stream = tda998x_audio_mute_stream,
 	.get_eld = tda998x_audio_get_eld,
+	.no_capture_mute = 1,
 };
 
 static int tda998x_audio_codec_init(struct tda998x_priv *priv,

drivers/gpu/drm/mediatek/mtk_hdmi.c

@@ -1643,7 +1643,8 @@ static void mtk_hdmi_audio_shutdown(struct device *dev, void *data)
 }
 
 static int
-mtk_hdmi_audio_digital_mute(struct device *dev, void *data, bool enable)
+mtk_hdmi_audio_mute(struct device *dev, void *data,
+		    bool enable, int direction)
 {
 	struct mtk_hdmi *hdmi = dev_get_drvdata(dev);
 
@@ -1684,9 +1685,10 @@ static const struct hdmi_codec_ops mtk_hdmi_audio_codec_ops = {
 	.hw_params = mtk_hdmi_audio_hw_params,
 	.audio_startup = mtk_hdmi_audio_startup,
 	.audio_shutdown = mtk_hdmi_audio_shutdown,
-	.digital_mute = mtk_hdmi_audio_digital_mute,
+	.mute_stream = mtk_hdmi_audio_mute,
 	.get_eld = mtk_hdmi_audio_get_eld,
 	.hook_plugged_cb = mtk_hdmi_audio_hook_plugged_cb,
+	.no_capture_mute = 1,
 };
 
 static int mtk_hdmi_register_audio_driver(struct device *dev)

drivers/gpu/drm/rockchip/cdn-dp-core.c

@@ -817,8 +817,8 @@ static void cdn_dp_audio_shutdown(struct device *dev, void *data)
 	mutex_unlock(&dp->lock);
 }
 
-static int cdn_dp_audio_digital_mute(struct device *dev, void *data,
-				     bool enable)
+static int cdn_dp_audio_mute_stream(struct device *dev, void *data,
+				    bool enable, int direction)
 {
 	struct cdn_dp_device *dp = dev_get_drvdata(dev);
 	int ret;
@@ -849,8 +849,9 @@ static int cdn_dp_audio_get_eld(struct device *dev, void *data,
 static const struct hdmi_codec_ops audio_codec_ops = {
 	.hw_params = cdn_dp_audio_hw_params,
 	.audio_shutdown = cdn_dp_audio_shutdown,
-	.digital_mute = cdn_dp_audio_digital_mute,
+	.mute_stream = cdn_dp_audio_mute_stream,
 	.get_eld = cdn_dp_audio_get_eld,
+	.no_capture_mute = 1,
 };
 
 static int cdn_dp_audio_codec_init(struct cdn_dp_device *dp,

drivers/gpu/drm/sti/sti_hdmi.c

@@ -1191,7 +1191,8 @@ static int hdmi_audio_hw_params(struct device *dev,
 	return 0;
 }
 
-static int hdmi_audio_digital_mute(struct device *dev, void *data, bool enable)
+static int hdmi_audio_mute(struct device *dev, void *data,
+			   bool enable, int direction)
 {
 	struct sti_hdmi *hdmi = dev_get_drvdata(dev);
 
@@ -1219,8 +1220,9 @@ static int hdmi_audio_get_eld(struct device *dev, void *data, uint8_t *buf, size
 static const struct hdmi_codec_ops audio_codec_ops = {
 	.hw_params = hdmi_audio_hw_params,
 	.audio_shutdown = hdmi_audio_shutdown,
-	.digital_mute = hdmi_audio_digital_mute,
+	.mute_stream = hdmi_audio_mute,
 	.get_eld = hdmi_audio_get_eld,
+	.no_capture_mute = 1,
 };
 
 static int sti_hdmi_register_audio_driver(struct device *dev,

drivers/gpu/drm/zte/zx_hdmi.c

@@ -439,8 +439,8 @@ static int zx_hdmi_audio_hw_params(struct device *dev,
 	return zx_hdmi_infoframe_trans(hdmi, &frame, FSEL_AUDIO);
 }
 
-static int zx_hdmi_audio_digital_mute(struct device *dev, void *data,
-				      bool enable)
+static int zx_hdmi_audio_mute(struct device *dev, void *data,
+			      bool enable, int direction)
 {
 	struct zx_hdmi *hdmi = dev_get_drvdata(dev);
 
@@ -468,8 +468,9 @@ static const struct hdmi_codec_ops zx_hdmi_codec_ops = {
 	.audio_startup = zx_hdmi_audio_startup,
 	.hw_params = zx_hdmi_audio_hw_params,
 	.audio_shutdown = zx_hdmi_audio_shutdown,
-	.digital_mute = zx_hdmi_audio_digital_mute,
+	.mute_stream = zx_hdmi_audio_mute,
 	.get_eld = zx_hdmi_audio_get_eld,
+	.no_capture_mute = 1,
 };
 
 static struct hdmi_codec_pdata zx_hdmi_codec_pdata = {

include/dt-bindings/sound/qcom,q6asm.h

@@ -19,4 +19,8 @@
 #define	MSM_FRONTEND_DAI_MULTIMEDIA15	14
 #define	MSM_FRONTEND_DAI_MULTIMEDIA16	15
 
+#define Q6ASM_DAI_TX_RX	0
+#define Q6ASM_DAI_TX	1
+#define Q6ASM_DAI_RX	2
+
 #endif /* __DT_BINDINGS_Q6_ASM_H__ */

include/sound/hda_codec.h

@@ -415,6 +415,8 @@ __printf(2, 3)
 struct hda_pcm *snd_hda_codec_pcm_new(struct hda_codec *codec,
 				      const char *fmt, ...);
 
+void snd_hda_codec_cleanup_for_unbind(struct hda_codec *codec);
+
 static inline void snd_hda_codec_pcm_get(struct hda_pcm *pcm)
 {
 	kref_get(&pcm->kref);

include/sound/hdmi-codec.h

@@ -76,7 +76,8 @@ struct hdmi_codec_ops {
 	 * Mute/unmute HDMI audio stream.
 	 * Optional
 	 */
-	int (*digital_mute)(struct device *dev, void *data, bool enable);
+	int (*mute_stream)(struct device *dev, void *data,
+			   bool enable, int direction);
 
 	/*
 	 * Provides EDID-Like-Data from connected HDMI device.
@@ -99,6 +100,9 @@ struct hdmi_codec_ops {
 	int (*hook_plugged_cb)(struct device *dev, void *data,
 			       hdmi_codec_plugged_cb fn,
 			       struct device *codec_dev);
+
+	/* bit field */
+	unsigned int no_capture_mute:1;
 };
 
 /* HDMI codec initalization data */

include/sound/rt5670.h

-/* SPDX-License-Identifier: GPL-2.0-only */
-/*
- * linux/sound/rt5670.h -- Platform data for RT5670
- *
- * Copyright 2014 Realtek Microelectronics
- */
-
-#ifndef __LINUX_SND_RT5670_H
-#define __LINUX_SND_RT5670_H
-
-struct rt5670_platform_data {
-	int jd_mode;
-	bool in2_diff;
-	bool dev_gpio;
-	bool gpio1_is_ext_spk_en;
-
-	bool dmic_en;
-	unsigned int dmic1_data_pin;
-	/* 0 = GPIO6; 1 = IN2P; 3 = GPIO7*/
-	unsigned int dmic2_data_pin;
-	/* 0 = GPIO8; 1 = IN3N; */
-	unsigned int dmic3_data_pin;
-	/* 0 = GPIO9; 1 = GPIO10; 2 = GPIO5*/
-};
-
-#endif

include/sound/simple_card_utils.h

@@ -12,9 +12,9 @@
 #include <sound/soc.h>
 
 #define asoc_simple_init_hp(card, sjack, prefix) \
-	asoc_simple_init_jack(card, sjack, 1, prefix)
+	asoc_simple_init_jack(card, sjack, 1, prefix, NULL)
 #define asoc_simple_init_mic(card, sjack, prefix) \
-	asoc_simple_init_jack(card, sjack, 0, prefix)
+	asoc_simple_init_jack(card, sjack, 0, prefix, NULL)
 
 struct asoc_simple_dai {
 	const char *name;
@@ -131,7 +131,7 @@ int asoc_simple_parse_pin_switches(struct snd_soc_card *card,
 
 int asoc_simple_init_jack(struct snd_soc_card *card,
 			       struct asoc_simple_jack *sjack,
-			       int is_hp, char *prefix);
+			       int is_hp, char *prefix, char *pin);
 int asoc_simple_init_priv(struct asoc_simple_priv *priv,
 			       struct link_info *li);
 

include/sound/soc-component.h

@@ -2,7 +2,8 @@
  *
  * soc-component.h
  *
- * Copyright (c) 2019 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+ * Copyright (C) 2019 Renesas Electronics Corp.
+ * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
  */
 #ifndef __SOC_COMPONENT_H
 #define __SOC_COMPONENT_H
@@ -324,10 +325,12 @@ static inline int snd_soc_component_cache_sync(
 	return regcache_sync(component->regmap);
 }
 
+void snd_soc_component_set_aux(struct snd_soc_component *component,
+			       struct snd_soc_aux_dev *aux);
+int snd_soc_component_init(struct snd_soc_component *component);
+
 /* component IO */
-int snd_soc_component_read(struct snd_soc_component *component,
-			   unsigned int reg, unsigned int *val);
-unsigned int snd_soc_component_read32(struct snd_soc_component *component,
+unsigned int snd_soc_component_read(struct snd_soc_component *component,
 				      unsigned int reg);
 int snd_soc_component_write(struct snd_soc_component *component,
 			    unsigned int reg, unsigned int val);
@@ -359,6 +362,7 @@ int snd_soc_component_stream_event(struct snd_soc_component *component,
 int snd_soc_component_set_bias_level(struct snd_soc_component *component,
 				     enum snd_soc_bias_level level);
 
+void snd_soc_component_setup_regmap(struct snd_soc_component *component);
 #ifdef CONFIG_REGMAP
 void snd_soc_component_init_regmap(struct snd_soc_component *component,
 				   struct regmap *regmap);
@@ -421,16 +425,6 @@ int snd_soc_component_open(struct snd_soc_component *component,
 			   struct snd_pcm_substream *substream);
 int snd_soc_component_close(struct snd_soc_component *component,
 			    struct snd_pcm_substream *substream);
-int snd_soc_component_prepare(struct snd_soc_component *component,
-			      struct snd_pcm_substream *substream);
-int snd_soc_component_hw_params(struct snd_soc_component *component,
-				struct snd_pcm_substream *substream,
-				struct snd_pcm_hw_params *params);
-int snd_soc_component_hw_free(struct snd_soc_component *component,
-			      struct snd_pcm_substream *substream);
-int snd_soc_component_trigger(struct snd_soc_component *component,
-			      struct snd_pcm_substream *substream,
-			      int cmd);
 void snd_soc_component_suspend(struct snd_soc_component *component);
 void snd_soc_component_resume(struct snd_soc_component *component);
 int snd_soc_component_is_suspended(struct snd_soc_component *component);
@@ -455,5 +449,13 @@ int snd_soc_pcm_component_mmap(struct snd_pcm_substream *substream,
 			       struct vm_area_struct *vma);
 int snd_soc_pcm_component_new(struct snd_soc_pcm_runtime *rtd);
 void snd_soc_pcm_component_free(struct snd_soc_pcm_runtime *rtd);
+int snd_soc_pcm_component_prepare(struct snd_pcm_substream *substream);
+int snd_soc_pcm_component_hw_params(struct snd_pcm_substream *substream,
+				    struct snd_pcm_hw_params *params,
+				    struct snd_soc_component **last);
+void snd_soc_pcm_component_hw_free(struct snd_pcm_substream *substream,
+				   struct snd_soc_component *last);
+int snd_soc_pcm_component_trigger(struct snd_pcm_substream *substream,
+				  int cmd);
 
 #endif /* __SOC_COMPONENT_H */

include/sound/soc-dai.h

@@ -39,7 +39,7 @@ struct snd_compr_stream;
 /*
  * DAI Clock gating.
  *
- * DAI bit clocks can be be gated (disabled) when the DAI is not
+ * DAI bit clocks can be gated (disabled) when the DAI is not
  * sending or receiving PCM data in a frame. This can be used to save power.
  */
 #define SND_SOC_DAIFMT_CONT		(1 << 4) /* continuous clock */
@@ -76,12 +76,12 @@ struct snd_compr_stream;
  *
  * This is wrt the codec, the inverse is true for the interface
  * i.e. if the codec is clk and FRM master then the interface is
- * clk and frame slave.
+ * clk and frame secondary.
  */
 #define SND_SOC_DAIFMT_CBM_CFM		(1 << 12) /* codec clk & FRM master */
-#define SND_SOC_DAIFMT_CBS_CFM		(2 << 12) /* codec clk slave & FRM master */
-#define SND_SOC_DAIFMT_CBM_CFS		(3 << 12) /* codec clk master & frame slave */
-#define SND_SOC_DAIFMT_CBS_CFS		(4 << 12) /* codec clk & FRM slave */
+#define SND_SOC_DAIFMT_CBS_CFM		(2 << 12) /* codec clk secondary & FRM master */
+#define SND_SOC_DAIFMT_CBM_CFS		(3 << 12) /* codec clk master & frame secondary */
+#define SND_SOC_DAIFMT_CBS_CFS		(4 << 12) /* codec clk & FRM secondary */
 
 #define SND_SOC_DAIFMT_FORMAT_MASK	0x000f
 #define SND_SOC_DAIFMT_CLOCK_MASK	0x00f0
@@ -247,7 +247,6 @@ struct snd_soc_dai_ops {
 	 * DAI digital mute - optional.
 	 * Called by soc-core to minimise any pops.
 	 */
-	int (*digital_mute)(struct snd_soc_dai *dai, int mute);
 	int (*mute_stream)(struct snd_soc_dai *dai, int mute, int stream);
 
 	/*
@@ -281,6 +280,9 @@ struct snd_soc_dai_ops {
 	 */
 	snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *,
 		struct snd_soc_dai *);
+
+	/* bit field */
+	unsigned int no_capture_mute:1;
 };
 
 struct snd_soc_cdai_ops {

include/sound/soc-dapm.h

@@ -16,6 +16,8 @@
 #include <sound/asoc.h>
 
 struct device;
+struct snd_soc_pcm_runtime;
+struct soc_enum;
 
 /* widget has no PM register bit */
 #define SND_SOC_NOPM	-1
@@ -376,6 +378,24 @@ struct snd_soc_dapm_widget_list;
 struct snd_soc_dapm_update;
 enum snd_soc_dapm_direction;
 
+/*
+ * Bias levels
+ *
+ * @ON:      Bias is fully on for audio playback and capture operations.
+ * @PREPARE: Prepare for audio operations. Called before DAPM switching for
+ *           stream start and stop operations.
+ * @STANDBY: Low power standby state when no playback/capture operations are
+ *           in progress. NOTE: The transition time between STANDBY and ON
+ *           should be as fast as possible and no longer than 10ms.
+ * @OFF:     Power Off. No restrictions on transition times.
+ */
+enum snd_soc_bias_level {
+	SND_SOC_BIAS_OFF = 0,
+	SND_SOC_BIAS_STANDBY = 1,
+	SND_SOC_BIAS_PREPARE = 2,
+	SND_SOC_BIAS_ON = 3,
+};
+
 int dapm_regulator_event(struct snd_soc_dapm_widget *w,
 			 struct snd_kcontrol *kcontrol, int event);
 int dapm_clock_event(struct snd_soc_dapm_widget *w,

include/sound/soc-link.h

@@ -9,6 +9,7 @@
 #define __SOC_LINK_H
 
 int snd_soc_link_init(struct snd_soc_pcm_runtime *rtd);
+void snd_soc_link_exit(struct snd_soc_pcm_runtime *rtd);
 int snd_soc_link_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
 				    struct snd_pcm_hw_params *params);
 

include/sound/soc.h

@@ -368,24 +368,6 @@
 #define SOC_ENUM_SINGLE_VIRT_DECL(name, xtexts) \
 	const struct soc_enum name = SOC_ENUM_SINGLE_VIRT(ARRAY_SIZE(xtexts), xtexts)
 
-/*
- * Bias levels
- *
- * @ON:      Bias is fully on for audio playback and capture operations.
- * @PREPARE: Prepare for audio operations. Called before DAPM switching for
- *           stream start and stop operations.
- * @STANDBY: Low power standby state when no playback/capture operations are
- *           in progress. NOTE: The transition time between STANDBY and ON
- *           should be as fast as possible and no longer than 10ms.
- * @OFF:     Power Off. No restrictions on transition times.
- */
-enum snd_soc_bias_level {
-	SND_SOC_BIAS_OFF = 0,
-	SND_SOC_BIAS_STANDBY = 1,
-	SND_SOC_BIAS_PREPARE = 2,
-	SND_SOC_BIAS_ON = 3,
-};
-
 struct device_node;
 struct snd_jack;
 struct snd_soc_card;
@@ -432,11 +414,12 @@ static inline int snd_soc_resume(struct device *dev)
 }
 #endif
 int snd_soc_poweroff(struct device *dev);
-int snd_soc_add_component(struct device *dev,
-		struct snd_soc_component *component,
-		const struct snd_soc_component_driver *component_driver,
-		struct snd_soc_dai_driver *dai_drv,
-		int num_dai);
+int snd_soc_component_initialize(struct snd_soc_component *component,
+				 const struct snd_soc_component_driver *driver,
+				 struct device *dev);
+int snd_soc_add_component(struct snd_soc_component *component,
+			  struct snd_soc_dai_driver *dai_drv,
+			  int num_dai);
 int snd_soc_register_component(struct device *dev,
 			 const struct snd_soc_component_driver *component_driver,
 			 struct snd_soc_dai_driver *dai_drv, int num_dai);
@@ -801,6 +784,9 @@ struct snd_soc_dai_link {
 	/* codec/machine specific init - e.g. add machine controls */
 	int (*init)(struct snd_soc_pcm_runtime *rtd);
 
+	/* codec/machine specific exit - dual of init() */
+	void (*exit)(struct snd_soc_pcm_runtime *rtd);
+
 	/* optional hw_params re-writing for BE and FE sync */
 	int (*be_hw_params_fixup)(struct snd_soc_pcm_runtime *rtd,
 			struct snd_pcm_hw_params *params);
@@ -1183,6 +1169,8 @@ struct snd_soc_pcm_runtime {
 /* see soc_new_pcm_runtime()  */
 #define asoc_rtd_to_cpu(rtd, n)   (rtd)->dais[n]
 #define asoc_rtd_to_codec(rtd, n) (rtd)->dais[n + (rtd)->num_cpus]
+#define asoc_substream_to_rtd(substream) \
+	(struct snd_soc_pcm_runtime *)snd_pcm_substream_chip(substream)
 
 #define for_each_rtd_components(rtd, i, component)			\
 	for ((i) = 0, component = NULL;					\

include/sound/wm8960.h

@@ -16,6 +16,23 @@ struct wm8960_data {
 	bool capless;  /* Headphone outputs configured in capless mode */
 
 	bool shared_lrclk;  /* DAC and ADC LRCLKs are wired together */
+
+	/*
+	 * Setup for headphone detection
+	 *
+	 * hp_cfg[0]: HPSEL[1:0] of R48 (Additional Control 4)
+	 * hp_cfg[1]: {HPSWEN:HPSWPOL} of R24 (Additional Control 2).
+	 * hp_cfg[2]: {TOCLKSEL:TOEN} of R23 (Additional Control 1).
+	 */
+	u32 hp_cfg[3];
+
+	/*
+	 * Setup for gpio configuration
+	 *
+	 * gpio_cfg[0]: ALRCGPIO of R9 (Audio interface)
+	 * gpio_cfg[1]: {GPIOPOL:GPIOSEL[2:0]} of R48 (Additional Control 4).
+	 */
+	u32 gpio_cfg[2];
 };
 
 #endif

sound/pci/hda/hda_codec.c

@@ -792,6 +792,7 @@ void snd_hda_codec_cleanup_for_unbind(struct hda_codec *codec)
 	remove_conn_list(codec);
 	snd_hdac_regmap_exit(&codec->core);
 }
+EXPORT_SYMBOL_GPL(snd_hda_codec_cleanup_for_unbind);
 
 static unsigned int hda_set_power_state(struct hda_codec *codec,
 				unsigned int power_state);
@@ -3178,7 +3179,7 @@ int snd_hda_codec_prepare(struct hda_codec *codec,
 EXPORT_SYMBOL_GPL(snd_hda_codec_prepare);
 
 /**
- * snd_hda_codec_cleanup - Prepare a stream
+ * snd_hda_codec_cleanup - Clean up stream resources
  * @codec: the HDA codec
  * @hinfo: PCM information
  * @substream: PCM substream

sound/soc/Makefile

 # SPDX-License-Identifier: GPL-2.0
 snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-utils.o soc-dai.o soc-component.o
-snd-soc-core-objs += soc-pcm.o soc-io.o soc-devres.o soc-ops.o soc-link.o soc-card.o
+snd-soc-core-objs += soc-pcm.o soc-devres.o soc-ops.o soc-link.o soc-card.o
 snd-soc-core-$(CONFIG_SND_SOC_COMPRESS) += soc-compress.o
 
 ifneq ($(CONFIG_SND_SOC_TOPOLOGY),)

sound/soc/amd/Kconfig

@@ -33,6 +33,7 @@ config SND_SOC_AMD_RV_RT5682_MACH
 	select SND_SOC_MAX98357A
 	select SND_SOC_CROS_EC_CODEC
 	select I2C_CROS_EC_TUNNEL
+	select SND_SOC_RT1015
 	depends on SND_SOC_AMD_ACP3x && I2C && CROS_EC
 	help
 	 This option enables machine driver for RT5682 and MAX9835.

sound/soc/amd/acp-da7219-max98357a.c

@@ -99,7 +99,7 @@ static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd)
 static int da7219_clk_enable(struct snd_pcm_substream *substream)
 {
 	int ret = 0;
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 
 	/*
 	 * Set wclk to 48000 because the rate constraint of this driver is
@@ -146,7 +146,7 @@ static const struct snd_pcm_hw_constraint_list constraints_channels = {
 static int cz_da7219_play_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct snd_soc_card *card = rtd->card;
 	struct acp_platform_info *machine = snd_soc_card_get_drvdata(card);
 
@@ -167,7 +167,7 @@ static int cz_da7219_play_startup(struct snd_pcm_substream *substream)
 static int cz_da7219_cap_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct snd_soc_card *card = rtd->card;
 	struct acp_platform_info *machine = snd_soc_card_get_drvdata(card);
 
@@ -189,7 +189,7 @@ static int cz_da7219_cap_startup(struct snd_pcm_substream *substream)
 static int cz_max_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct snd_soc_card *card = rtd->card;
 	struct acp_platform_info *machine = snd_soc_card_get_drvdata(card);
 
@@ -210,7 +210,7 @@ static int cz_max_startup(struct snd_pcm_substream *substream)
 static int cz_dmic0_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct snd_soc_card *card = rtd->card;
 	struct acp_platform_info *machine = snd_soc_card_get_drvdata(card);
 
@@ -231,7 +231,7 @@ static int cz_dmic0_startup(struct snd_pcm_substream *substream)
 static int cz_dmic1_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct snd_soc_card *card = rtd->card;
 	struct acp_platform_info *machine = snd_soc_card_get_drvdata(card);
 
@@ -450,11 +450,13 @@ static int cz_probe(struct platform_device *pdev)
 	return 0;
 }
 
+#ifdef CONFIG_ACPI
 static const struct acpi_device_id cz_audio_acpi_match[] = {
 	{ "AMD7219", 0 },
 	{},
 };
 MODULE_DEVICE_TABLE(acpi, cz_audio_acpi_match);
+#endif
 
 static struct platform_driver cz_pcm_driver = {
 	.driver = {

sound/soc/amd/acp-pcm-dma.c

@@ -840,7 +840,7 @@ static int acp_dma_hw_params(struct snd_soc_component *component,
 	u32 val = 0;
 	struct snd_pcm_runtime *runtime;
 	struct audio_substream_data *rtd;
-	struct snd_soc_pcm_runtime *prtd = substream->private_data;
+	struct snd_soc_pcm_runtime *prtd = asoc_substream_to_rtd(substream);
 	struct audio_drv_data *adata = dev_get_drvdata(component->dev);
 	struct snd_soc_card *card = prtd->card;
 	struct acp_platform_info *pinfo = snd_soc_card_get_drvdata(card);

sound/soc/amd/acp-rt5645.c

@@ -47,7 +47,7 @@ static int cz_aif1_hw_params(struct snd_pcm_substream *substream,
 			     struct snd_pcm_hw_params *params)
 {
 	int ret = 0;
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 
 	ret = snd_soc_dai_set_pll(codec_dai, 0, RT5645_PLL1_S_MCLK,
@@ -182,11 +182,13 @@ static int cz_probe(struct platform_device *pdev)
 	return 0;
 }
 
+#ifdef CONFIG_ACPI
 static const struct acpi_device_id cz_audio_acpi_match[] = {
 	{ "AMDI1002", 0 },
 	{},
 };
 MODULE_DEVICE_TABLE(acpi, cz_audio_acpi_match);
+#endif
 
 static struct platform_driver cz_pcm_driver = {
 	.driver = {

sound/soc/amd/acp3x-rt5682-max9836.c

@@ -21,6 +21,7 @@
 
 #include "raven/acp3x.h"
 #include "../codecs/rt5682.h"
+#include "../codecs/rt1015.h"
 
 #define PCO_PLAT_CLK 48000000
 #define RT5682_PLL_FREQ (48000 * 512)
@@ -30,6 +31,13 @@ static struct snd_soc_jack pco_jack;
 static struct clk *rt5682_dai_wclk;
 static struct clk *rt5682_dai_bclk;
 static struct gpio_desc *dmic_sel;
+void *soc_is_rltk_max(struct device *dev);
+
+enum {
+	RT5682 = 0,
+	MAX,
+	EC,
+};
 
 static int acp3x_5682_init(struct snd_soc_pcm_runtime *rtd)
 {
@@ -105,7 +113,7 @@ static int acp3x_5682_init(struct snd_soc_pcm_runtime *rtd)
 static int rt5682_clk_enable(struct snd_pcm_substream *substream)
 {
 	int ret = 0;
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 
 	/* RT5682 will support only 48K output with 48M mclk */
 	clk_set_rate(rt5682_dai_wclk, 48000);
@@ -119,6 +127,34 @@ static int rt5682_clk_enable(struct snd_pcm_substream *substream)
 	return ret;
 }
 
+static int acp3x_1015_hw_params(struct snd_pcm_substream *substream,
+					struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai;
+	int srate, i, ret;
+
+	ret = 0;
+	srate = params_rate(params);
+
+	for_each_rtd_codec_dais(rtd, i, codec_dai) {
+		if (strcmp(codec_dai->component->name, "rt1015-aif"))
+			continue;
+		ret = snd_soc_dai_set_bclk_ratio(codec_dai, 64);
+		if (ret < 0)
+			return ret;
+		ret = snd_soc_dai_set_pll(codec_dai, 0, RT1015_PLL_S_BCLK,
+						64 * srate, 256 * srate);
+		if (ret < 0)
+			return ret;
+		ret = snd_soc_dai_set_sysclk(codec_dai, RT1015_SCLK_S_PLL,
+					256 * srate, SND_SOC_CLOCK_IN);
+		if (ret < 0)
+			return ret;
+	}
+	return ret;
+}
+
 static void rt5682_clk_disable(void)
 {
 	clk_disable_unprepare(rt5682_dai_wclk);
@@ -147,7 +183,7 @@ static const struct snd_pcm_hw_constraint_list constraints_channels = {
 static int acp3x_5682_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct snd_soc_card *card = rtd->card;
 	struct acp3x_platform_info *machine = snd_soc_card_get_drvdata(card);
 
@@ -165,7 +201,7 @@ static int acp3x_5682_startup(struct snd_pcm_substream *substream)
 static int acp3x_max_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct snd_soc_card *card = rtd->card;
 	struct acp3x_platform_info *machine = snd_soc_card_get_drvdata(card);
 
@@ -181,32 +217,34 @@ static int acp3x_max_startup(struct snd_pcm_substream *substream)
 
 static int acp3x_ec_dmic0_startup(struct snd_pcm_substream *substream)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct snd_soc_card *card = rtd->card;
 	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	struct acp3x_platform_info *machine = snd_soc_card_get_drvdata(card);
 
 	machine->cap_i2s_instance = I2S_BT_INSTANCE;
 	snd_soc_dai_set_bclk_ratio(codec_dai, 64);
-	if (dmic_sel)
-		gpiod_set_value(dmic_sel, 0);
 
 	return rt5682_clk_enable(substream);
 }
 
-static int acp3x_ec_dmic1_startup(struct snd_pcm_substream *substream)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_card *card = rtd->card;
-	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
-	struct acp3x_platform_info *machine = snd_soc_card_get_drvdata(card);
+static int dmic_switch;
 
-	machine->cap_i2s_instance = I2S_BT_INSTANCE;
-	snd_soc_dai_set_bclk_ratio(codec_dai, 64);
-	if (dmic_sel)
-		gpiod_set_value(dmic_sel, 1);
+static int dmic_get(struct snd_kcontrol *kcontrol,
+			 struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = dmic_switch;
+	return 0;
+}
 
-	return rt5682_clk_enable(substream);
+static int dmic_set(struct snd_kcontrol *kcontrol,
+			 struct snd_ctl_elem_value *ucontrol)
+{
+	if (dmic_sel) {
+		dmic_switch = ucontrol->value.integer.value[0];
+		gpiod_set_value(dmic_sel, dmic_switch);
+	}
+	return 0;
 }
 
 static void rt5682_shutdown(struct snd_pcm_substream *substream)
@@ -222,6 +260,7 @@ static const struct snd_soc_ops acp3x_5682_ops = {
 static const struct snd_soc_ops acp3x_max_play_ops = {
 	.startup = acp3x_max_startup,
 	.shutdown = rt5682_shutdown,
+	.hw_params = acp3x_1015_hw_params,
 };
 
 static const struct snd_soc_ops acp3x_ec_cap0_ops = {
@@ -229,11 +268,6 @@ static const struct snd_soc_ops acp3x_ec_cap0_ops = {
 	.shutdown = rt5682_shutdown,
 };
 
-static const struct snd_soc_ops acp3x_ec_cap1_ops = {
-	.startup = acp3x_ec_dmic1_startup,
-	.shutdown = rt5682_shutdown,
-};
-
 SND_SOC_DAILINK_DEF(acp3x_i2s,
 	DAILINK_COMP_ARRAY(COMP_CPU("acp3x_i2s_playcap.0")));
 SND_SOC_DAILINK_DEF(acp3x_bt,
@@ -243,14 +277,28 @@ SND_SOC_DAILINK_DEF(rt5682,
 	DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10EC5682:00", "rt5682-aif1")));
 SND_SOC_DAILINK_DEF(max,
 	DAILINK_COMP_ARRAY(COMP_CODEC("MX98357A:00", "HiFi")));
+SND_SOC_DAILINK_DEF(rt1015,
+	DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10EC1015:00", "rt1015-aif"),
+			COMP_CODEC("i2c-10EC1015:01", "rt1015-aif")));
 SND_SOC_DAILINK_DEF(cros_ec,
 	DAILINK_COMP_ARRAY(COMP_CODEC("GOOG0013:00", "EC Codec I2S RX")));
 
 SND_SOC_DAILINK_DEF(platform,
 	DAILINK_COMP_ARRAY(COMP_PLATFORM("acp3x_rv_i2s_dma.0")));
 
-static struct snd_soc_dai_link acp3x_dai_5682_98357[] = {
+static struct snd_soc_codec_conf rt1015_conf[] = {
+	{
+		.dlc = COMP_CODEC_CONF("i2c-10EC1015:00"),
+		.name_prefix = "Left",
+	},
 	{
+		.dlc = COMP_CODEC_CONF("i2c-10EC1015:01"),
+		.name_prefix = "Right",
+	},
+};
+
+static struct snd_soc_dai_link acp3x_dai[] = {
+	[RT5682] = {
 		.name = "acp3x-5682-play",
 		.stream_name = "Playback",
 		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
@@ -261,16 +309,19 @@ static struct snd_soc_dai_link acp3x_dai_5682_98357[] = {
 		.ops = &acp3x_5682_ops,
 		SND_SOC_DAILINK_REG(acp3x_i2s, rt5682, platform),
 	},
-	{
+	[MAX] = {
 		.name = "acp3x-max98357-play",
 		.stream_name = "HiFi Playback",
 		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
-				| SND_SOC_DAIFMT_CBM_CFM,
+				| SND_SOC_DAIFMT_CBS_CFS,
 		.dpcm_playback = 1,
 		.ops = &acp3x_max_play_ops,
-		SND_SOC_DAILINK_REG(acp3x_bt, max, platform),
+		.cpus = acp3x_bt,
+		.num_cpus = ARRAY_SIZE(acp3x_bt),
+		.platforms = platform,
+		.num_platforms = ARRAY_SIZE(platform),
 	},
-	{
+	[EC] = {
 		.name = "acp3x-ec-dmic0-capture",
 		.stream_name = "Capture DMIC0",
 		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
@@ -279,61 +330,136 @@ static struct snd_soc_dai_link acp3x_dai_5682_98357[] = {
 		.ops = &acp3x_ec_cap0_ops,
 		SND_SOC_DAILINK_REG(acp3x_bt, cros_ec, platform),
 	},
-	{
-		.name = "acp3x-ec-dmic1-capture",
-		.stream_name = "Capture DMIC1",
-		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
-				| SND_SOC_DAIFMT_CBS_CFS,
-		.dpcm_capture = 1,
-		.ops = &acp3x_ec_cap1_ops,
-		SND_SOC_DAILINK_REG(acp3x_bt, cros_ec, platform),
-	},
 };
 
-static const struct snd_soc_dapm_widget acp3x_widgets[] = {
+static const char * const dmic_mux_text[] = {
+	"Front Mic",
+	"Rear Mic",
+};
+
+static SOC_ENUM_SINGLE_DECL(
+		acp3x_dmic_enum, SND_SOC_NOPM, 0, dmic_mux_text);
+
+static const struct snd_kcontrol_new acp3x_dmic_mux_control =
+	SOC_DAPM_ENUM_EXT("DMIC Select Mux", acp3x_dmic_enum,
+			  dmic_get, dmic_set);
+
+static const struct snd_soc_dapm_widget acp3x_5682_widgets[] = {
 	SND_SOC_DAPM_HP("Headphone Jack", NULL),
 	SND_SOC_DAPM_SPK("Spk", NULL),
 	SND_SOC_DAPM_MIC("Headset Mic", NULL),
+	SND_SOC_DAPM_MUX("Dmic Mux", SND_SOC_NOPM, 0, 0,
+			 &acp3x_dmic_mux_control),
 };
 
-static const struct snd_soc_dapm_route acp3x_audio_route[] = {
+static const struct snd_soc_dapm_route acp3x_5682_audio_route[] = {
 	{"Headphone Jack", NULL, "HPOL"},
 	{"Headphone Jack", NULL, "HPOR"},
 	{"IN1P", NULL, "Headset Mic"},
 	{"Spk", NULL, "Speaker"},
+	{"Dmic Mux", "Front Mic", "DMIC"},
+	{"Dmic Mux", "Rear Mic", "DMIC"},
 };
 
-static const struct snd_kcontrol_new acp3x_mc_controls[] = {
+static const struct snd_kcontrol_new acp3x_5682_mc_controls[] = {
 	SOC_DAPM_PIN_SWITCH("Headphone Jack"),
 	SOC_DAPM_PIN_SWITCH("Spk"),
 	SOC_DAPM_PIN_SWITCH("Headset Mic"),
 };
 
-static struct snd_soc_card acp3x_card = {
+static struct snd_soc_card acp3x_5682 = {
 	.name = "acp3xalc5682m98357",
 	.owner = THIS_MODULE,
-	.dai_link = acp3x_dai_5682_98357,
-	.num_links = ARRAY_SIZE(acp3x_dai_5682_98357),
-	.dapm_widgets = acp3x_widgets,
-	.num_dapm_widgets = ARRAY_SIZE(acp3x_widgets),
-	.dapm_routes = acp3x_audio_route,
-	.num_dapm_routes = ARRAY_SIZE(acp3x_audio_route),
-	.controls = acp3x_mc_controls,
-	.num_controls = ARRAY_SIZE(acp3x_mc_controls),
+	.dai_link = acp3x_dai,
+	.num_links = ARRAY_SIZE(acp3x_dai),
+	.dapm_widgets = acp3x_5682_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(acp3x_5682_widgets),
+	.dapm_routes = acp3x_5682_audio_route,
+	.num_dapm_routes = ARRAY_SIZE(acp3x_5682_audio_route),
+	.controls = acp3x_5682_mc_controls,
+	.num_controls = ARRAY_SIZE(acp3x_5682_mc_controls),
+};
+
+static const struct snd_soc_dapm_widget acp3x_1015_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_MIC("Headset Mic", NULL),
+	SND_SOC_DAPM_MUX("Dmic Mux", SND_SOC_NOPM, 0, 0,
+			 &acp3x_dmic_mux_control),
+	SND_SOC_DAPM_SPK("Left Spk", NULL),
+	SND_SOC_DAPM_SPK("Right Spk", NULL),
+};
+
+static const struct snd_soc_dapm_route acp3x_1015_route[] = {
+	{"Headphone Jack", NULL, "HPOL"},
+	{"Headphone Jack", NULL, "HPOR"},
+	{"IN1P", NULL, "Headset Mic"},
+	{"Dmic Mux", "Front Mic", "DMIC"},
+	{"Dmic Mux", "Rear Mic", "DMIC"},
+	{"Left Spk", NULL, "Left SPO"},
+	{"Right Spk", NULL, "Right SPO"},
+};
+
+static const struct snd_kcontrol_new acp3x_mc_1015_controls[] = {
+	SOC_DAPM_PIN_SWITCH("Headphone Jack"),
+	SOC_DAPM_PIN_SWITCH("Headset Mic"),
+	SOC_DAPM_PIN_SWITCH("Left Spk"),
+	SOC_DAPM_PIN_SWITCH("Right Spk"),
+};
+
+static struct snd_soc_card acp3x_1015 = {
+	.name = "acp3xalc56821015",
+	.owner = THIS_MODULE,
+	.dai_link = acp3x_dai,
+	.num_links = ARRAY_SIZE(acp3x_dai),
+	.dapm_widgets = acp3x_1015_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(acp3x_1015_widgets),
+	.dapm_routes = acp3x_1015_route,
+	.num_dapm_routes = ARRAY_SIZE(acp3x_1015_route),
+	.codec_conf = rt1015_conf,
+	.num_configs = ARRAY_SIZE(rt1015_conf),
+	.controls = acp3x_mc_1015_controls,
+	.num_controls = ARRAY_SIZE(acp3x_mc_1015_controls),
 };
 
+void *soc_is_rltk_max(struct device *dev)
+{
+	const struct acpi_device_id *match;
+
+	match = acpi_match_device(dev->driver->acpi_match_table, dev);
+	if (!match)
+		return NULL;
+	return (void *)match->driver_data;
+}
+
+static void card_spk_dai_link_present(struct snd_soc_dai_link *links,
+						const char *card_name)
+{
+	if (!strcmp(card_name, "acp3xalc56821015")) {
+		links[1].codecs = rt1015;
+		links[1].num_codecs = ARRAY_SIZE(rt1015);
+	} else {
+		links[1].codecs = max;
+		links[1].num_codecs = ARRAY_SIZE(max);
+	}
+}
+
 static int acp3x_probe(struct platform_device *pdev)
 {
 	int ret;
 	struct snd_soc_card *card;
 	struct acp3x_platform_info *machine;
+	struct device *dev = &pdev->dev;
+
+	card = (struct snd_soc_card *)soc_is_rltk_max(dev);
+	if (!card)
+		return -ENODEV;
 
 	machine = devm_kzalloc(&pdev->dev, sizeof(*machine), GFP_KERNEL);
 	if (!machine)
 		return -ENOMEM;
 
-	card = &acp3x_card;
-	acp3x_card.dev = &pdev->dev;
+	card_spk_dai_link_present(card->dai_link, card->name);
+	card->dev = &pdev->dev;
 	platform_set_drvdata(pdev, card);
 	snd_soc_card_set_drvdata(card, machine);
 
@@ -344,18 +470,19 @@ static int acp3x_probe(struct platform_device *pdev)
 		return PTR_ERR(dmic_sel);
 	}
 
-	ret = devm_snd_soc_register_card(&pdev->dev, &acp3x_card);
+	ret = devm_snd_soc_register_card(&pdev->dev, card);
 	if (ret) {
 		dev_err(&pdev->dev,
 				"devm_snd_soc_register_card(%s) failed: %d\n",
-				acp3x_card.name, ret);
+				card->name, ret);
 		return ret;
 	}
 	return 0;
 }
 
 static const struct acpi_device_id acp3x_audio_acpi_match[] = {
-	{ "AMDI5682", 0 },
+	{ "AMDI5682", (unsigned long)&acp3x_5682},
+	{ "AMDI1015", (unsigned long)&acp3x_1015},
 	{},
 };
 MODULE_DEVICE_TABLE(acpi, acp3x_audio_acpi_match);
@@ -372,5 +499,6 @@ static struct platform_driver acp3x_audio = {
 module_platform_driver(acp3x_audio);
 
 MODULE_AUTHOR("akshu.agrawal@amd.com");
-MODULE_DESCRIPTION("ALC5682 & MAX98357 audio support");
+MODULE_AUTHOR("Vishnuvardhanrao.Ravulapati@amd.com");
+MODULE_DESCRIPTION("ALC5682 ALC1015 & MAX98357 audio support");
 MODULE_LICENSE("GPL v2");

sound/soc/amd/raven/acp3x-i2s.c

@@ -80,7 +80,7 @@ static int acp3x_i2s_hwparams(struct snd_pcm_substream *substream,
 	u32 val;
 	u32 reg_val, frmt_reg;
 
-	prtd = substream->private_data;
+	prtd = asoc_substream_to_rtd(substream);
 	rtd = substream->runtime->private_data;
 	card = prtd->card;
 	adata = snd_soc_dai_get_drvdata(dai);
@@ -149,22 +149,10 @@ static int acp3x_i2s_trigger(struct snd_pcm_substream *substream,
 				int cmd, struct snd_soc_dai *dai)
 {
 	struct i2s_stream_instance *rtd;
-	struct snd_soc_pcm_runtime *prtd;
-	struct snd_soc_card *card;
-	struct acp3x_platform_info *pinfo;
 	u32 ret, val, period_bytes, reg_val, ier_val, water_val;
 	u32 buf_size, buf_reg;
 
-	prtd = substream->private_data;
 	rtd = substream->runtime->private_data;
-	card = prtd->card;
-	pinfo = snd_soc_card_get_drvdata(card);
-	if (pinfo) {
-		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-			rtd->i2s_instance = pinfo->play_i2s_instance;
-		else
-			rtd->i2s_instance = pinfo->cap_i2s_instance;
-	}
 	period_bytes = frames_to_bytes(substream->runtime,
 			substream->runtime->period_size);
 	buf_size = frames_to_bytes(substream->runtime,

sound/soc/amd/raven/acp3x-pcm-dma.c

@@ -217,7 +217,7 @@ static int acp3x_dma_open(struct snd_soc_component *component,
 	int ret;
 
 	runtime = substream->runtime;
-	prtd = substream->private_data;
+	prtd = asoc_substream_to_rtd(substream);
 	component = snd_soc_rtdcom_lookup(prtd, DRV_NAME);
 	adata = dev_get_drvdata(component->dev);
 	i2s_data = kzalloc(sizeof(*i2s_data), GFP_KERNEL);
@@ -238,7 +238,7 @@ static int acp3x_dma_open(struct snd_soc_component *component,
 	}
 
 	if (!adata->play_stream && !adata->capture_stream &&
-		adata->i2ssp_play_stream && !adata->i2ssp_capture_stream)
+	    !adata->i2ssp_play_stream && !adata->i2ssp_capture_stream)
 		rv_writel(1, adata->acp3x_base + mmACP_EXTERNAL_INTR_ENB);
 
 	i2s_data->acp3x_base = adata->acp3x_base;
@@ -258,7 +258,7 @@ static int acp3x_dma_hw_params(struct snd_soc_component *component,
 	struct i2s_dev_data *adata;
 	u64 size;
 
-	prtd = substream->private_data;
+	prtd = asoc_substream_to_rtd(substream);
 	card = prtd->card;
 	pinfo = snd_soc_card_get_drvdata(card);
 	adata = dev_get_drvdata(component->dev);
@@ -301,15 +301,11 @@ static int acp3x_dma_hw_params(struct snd_soc_component *component,
 static snd_pcm_uframes_t acp3x_dma_pointer(struct snd_soc_component *component,
 					   struct snd_pcm_substream *substream)
 {
-	struct snd_soc_pcm_runtime *prtd;
-	struct snd_soc_card *card;
 	struct i2s_stream_instance *rtd;
 	u32 pos;
 	u32 buffersize;
 	u64 bytescount;
 
-	prtd = substream->private_data;
-	card = prtd->card;
 	rtd = substream->runtime->private_data;
 
 	buffersize = frames_to_bytes(substream->runtime,
@@ -344,7 +340,7 @@ static int acp3x_dma_close(struct snd_soc_component *component,
 	struct i2s_dev_data *adata;
 	struct i2s_stream_instance *ins;
 
-	prtd = substream->private_data;
+	prtd = asoc_substream_to_rtd(substream);
 	component = snd_soc_rtdcom_lookup(prtd, DRV_NAME);
 	adata = dev_get_drvdata(component->dev);
 	ins = substream->runtime->private_data;

sound/soc/amd/raven/pci-acp3x.c

@@ -19,10 +19,12 @@ struct acp3x_dev_data {
 	bool acp3x_audio_mode;
 	struct resource *res;
 	struct platform_device *pdev[ACP3x_DEVS];
+	u32 pme_en;
 };
 
-static int acp3x_power_on(void __iomem *acp3x_base)
+static int acp3x_power_on(struct acp3x_dev_data *adata)
 {
+	void __iomem *acp3x_base = adata->acp3x_base;
 	u32 val;
 	int timeout;
 
@@ -39,10 +41,10 @@ static int acp3x_power_on(void __iomem *acp3x_base)
 	while (++timeout < 500) {
 		val = rv_readl(acp3x_base + mmACP_PGFSM_STATUS);
 		if (!val) {
-			/* Set PME_EN as after ACP power On,
-			 * PME_EN gets cleared
+			/* ACP power On clears PME_EN.
+			 * Restore the value to its prior state
 			 */
-			rv_writel(0x1, acp3x_base + mmACP_PME_EN);
+			rv_writel(adata->pme_en, acp3x_base + mmACP_PME_EN);
 			return 0;
 		}
 		udelay(1);
@@ -74,12 +76,13 @@ static int acp3x_reset(void __iomem *acp3x_base)
 	return -ETIMEDOUT;
 }
 
-static int acp3x_init(void __iomem *acp3x_base)
+static int acp3x_init(struct acp3x_dev_data *adata)
 {
+	void __iomem *acp3x_base = adata->acp3x_base;
 	int ret;
 
 	/* power on */
-	ret = acp3x_power_on(acp3x_base);
+	ret = acp3x_power_on(adata);
 	if (ret) {
 		pr_err("ACP3x power on failed\n");
 		return ret;
@@ -151,7 +154,9 @@ static int snd_acp3x_probe(struct pci_dev *pci,
 	}
 	pci_set_master(pci);
 	pci_set_drvdata(pci, adata);
-	ret = acp3x_init(adata->acp3x_base);
+	/* Save ACP_PME_EN state */
+	adata->pme_en = rv_readl(adata->acp3x_base + mmACP_PME_EN);
+	ret = acp3x_init(adata);
 	if (ret)
 		goto disable_msi;
 
@@ -274,7 +279,7 @@ static int snd_acp3x_resume(struct device *dev)
 	struct acp3x_dev_data *adata;
 
 	adata = dev_get_drvdata(dev);
-	ret = acp3x_init(adata->acp3x_base);
+	ret = acp3x_init(adata);
 	if (ret) {
 		dev_err(dev, "ACP init failed\n");
 		return ret;

sound/soc/amd/renoir/rn-pci-acp3x.c

@@ -5,6 +5,7 @@
 //Copyright 2020 Advanced Micro Devices, Inc.
 
 #include <linux/pci.h>
+#include <linux/acpi.h>
 #include <linux/module.h>
 #include <linux/io.h>
 #include <linux/delay.h>
@@ -18,6 +19,16 @@ static int acp_power_gating;
 module_param(acp_power_gating, int, 0644);
 MODULE_PARM_DESC(acp_power_gating, "Enable acp power gating");
 
+/**
+ * dmic_acpi_check = -1 - Checks ACPI method to know DMIC hardware status runtime
+ *                 = 0 - Skips the DMIC device creation and returns probe failure
+ *                 = 1 - Assumes that platform has DMIC support and skips ACPI
+ *                       method check
+ */
+static int dmic_acpi_check = ACP_DMIC_AUTO;
+module_param(dmic_acpi_check, bint, 0644);
+MODULE_PARM_DESC(dmic_acpi_check, "checks Dmic hardware runtime");
+
 struct acp_dev_data {
 	void __iomem *acp_base;
 	struct resource *res;
@@ -157,6 +168,10 @@ static int snd_rn_acp_probe(struct pci_dev *pci,
 {
 	struct acp_dev_data *adata;
 	struct platform_device_info pdevinfo[ACP_DEVS];
+#if defined(CONFIG_ACPI)
+	acpi_handle handle;
+	acpi_integer dmic_status;
+#endif
 	unsigned int irqflags;
 	int ret, index;
 	u32 addr;
@@ -201,6 +216,24 @@ static int snd_rn_acp_probe(struct pci_dev *pci,
 	if (ret)
 		goto disable_msi;
 
+	if (!dmic_acpi_check) {
+		ret = -ENODEV;
+		goto de_init;
+	} else if (dmic_acpi_check == ACP_DMIC_AUTO) {
+#if defined(CONFIG_ACPI)
+		handle = ACPI_HANDLE(&pci->dev);
+		ret = acpi_evaluate_integer(handle, "_WOV", NULL, &dmic_status);
+		if (ACPI_FAILURE(ret)) {
+			ret = -EINVAL;
+			goto de_init;
+		}
+		if (!dmic_status) {
+			ret = -ENODEV;
+			goto de_init;
+		}
+#endif
+	}
+
 	adata->res = devm_kzalloc(&pci->dev,
 				  sizeof(struct resource) * 2,
 				  GFP_KERNEL);

sound/soc/amd/renoir/rn_acp3x.h

@@ -55,6 +55,8 @@
 
 #define MAX_BUFFER (CAPTURE_MAX_PERIOD_SIZE * CAPTURE_MAX_NUM_PERIODS)
 #define MIN_BUFFER MAX_BUFFER
+#define	ACP_DMIC_AUTO   -1
+
 struct pdm_dev_data {
 	u32 pdm_irq;
 	void __iomem *acp_base;

sound/soc/atmel/atmel-classd.c

@@ -118,48 +118,30 @@ static const struct snd_pcm_hardware atmel_classd_hw = {
 static int atmel_classd_cpu_dai_startup(struct snd_pcm_substream *substream,
 					struct snd_soc_dai *cpu_dai)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card);
+	int err;
 
 	regmap_write(dd->regmap, CLASSD_THR, 0x0);
 
-	return clk_prepare_enable(dd->pclk);
-}
-
-static void atmel_classd_cpu_dai_shutdown(struct snd_pcm_substream *substream,
-					struct snd_soc_dai *cpu_dai)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card);
-
-	clk_disable_unprepare(dd->pclk);
+	err = clk_prepare_enable(dd->pclk);
+	if (err)
+		return err;
+	err = clk_prepare_enable(dd->gclk);
+	if (err) {
+		clk_disable_unprepare(dd->pclk);
+		return err;
+	}
+	return 0;
 }
 
-static const struct snd_soc_dai_ops atmel_classd_cpu_dai_ops = {
-	.startup	= atmel_classd_cpu_dai_startup,
-	.shutdown	= atmel_classd_cpu_dai_shutdown,
-};
-
-static struct snd_soc_dai_driver atmel_classd_cpu_dai = {
-	.playback = {
-		.channels_min	= 1,
-		.channels_max	= 2,
-		.rates		= ATMEL_CLASSD_RATES,
-		.formats	= SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = &atmel_classd_cpu_dai_ops,
-};
-
-static const struct snd_soc_component_driver atmel_classd_cpu_dai_component = {
-	.name = "atmel-classd",
-};
-
 /* platform */
 static int
 atmel_classd_platform_configure_dma(struct snd_pcm_substream *substream,
 	struct snd_pcm_hw_params *params,
 	struct dma_slave_config *slave_config)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card);
 
 	if (params_physical_width(params) != 16) {
@@ -306,31 +288,10 @@ static int atmel_classd_component_resume(struct snd_soc_component *component)
 	return regcache_sync(dd->regmap);
 }
 
-static struct snd_soc_component_driver soc_component_dev_classd = {
-	.probe			= atmel_classd_component_probe,
-	.resume			= atmel_classd_component_resume,
-	.controls		= atmel_classd_snd_controls,
-	.num_controls		= ARRAY_SIZE(atmel_classd_snd_controls),
-	.idle_bias_on		= 1,
-	.use_pmdown_time	= 1,
-	.endianness		= 1,
-	.non_legacy_dai_naming	= 1,
-};
-
-/* codec dai component */
-static int atmel_classd_codec_dai_startup(struct snd_pcm_substream *substream,
-				struct snd_soc_dai *codec_dai)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card);
-
-	return clk_prepare_enable(dd->gclk);
-}
-
-static int atmel_classd_codec_dai_digital_mute(struct snd_soc_dai *codec_dai,
-	int mute)
+static int atmel_classd_cpu_dai_mute_stream(struct snd_soc_dai *cpu_dai,
+					    int mute, int direction)
 {
-	struct snd_soc_component *component = codec_dai->component;
+	struct snd_soc_component *component = cpu_dai->component;
 	u32 mask, val;
 
 	mask = CLASSD_MR_LMUTE_MASK | CLASSD_MR_RMUTE_MASK;
@@ -373,13 +334,13 @@ static struct {
 };
 
 static int
-atmel_classd_codec_dai_hw_params(struct snd_pcm_substream *substream,
-			    struct snd_pcm_hw_params *params,
-			    struct snd_soc_dai *codec_dai)
+atmel_classd_cpu_dai_hw_params(struct snd_pcm_substream *substream,
+			       struct snd_pcm_hw_params *params,
+			       struct snd_soc_dai *cpu_dai)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card);
-	struct snd_soc_component *component = codec_dai->component;
+	struct snd_soc_component *component = cpu_dai->component;
 	int fs;
 	int i, best, best_val, cur_val, ret;
 	u32 mask, val;
@@ -417,19 +378,19 @@ atmel_classd_codec_dai_hw_params(struct snd_pcm_substream *substream,
 }
 
 static void
-atmel_classd_codec_dai_shutdown(struct snd_pcm_substream *substream,
-			    struct snd_soc_dai *codec_dai)
+atmel_classd_cpu_dai_shutdown(struct snd_pcm_substream *substream,
+			      struct snd_soc_dai *cpu_dai)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card);
 
 	clk_disable_unprepare(dd->gclk);
 }
 
-static int atmel_classd_codec_dai_prepare(struct snd_pcm_substream *substream,
-					struct snd_soc_dai *codec_dai)
+static int atmel_classd_cpu_dai_prepare(struct snd_pcm_substream *substream,
+					struct snd_soc_dai *cpu_dai)
 {
-	struct snd_soc_component *component = codec_dai->component;
+	struct snd_soc_component *component = cpu_dai->component;
 
 	snd_soc_component_update_bits(component, CLASSD_MR,
 				CLASSD_MR_LEN_MASK | CLASSD_MR_REN_MASK,
@@ -439,10 +400,10 @@ static int atmel_classd_codec_dai_prepare(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int atmel_classd_codec_dai_trigger(struct snd_pcm_substream *substream,
-					int cmd, struct snd_soc_dai *codec_dai)
+static int atmel_classd_cpu_dai_trigger(struct snd_pcm_substream *substream,
+					int cmd, struct snd_soc_dai *cpu_dai)
 {
-	struct snd_soc_component *component = codec_dai->component;
+	struct snd_soc_component *component = cpu_dai->component;
 	u32 mask, val;
 
 	mask = CLASSD_MR_LEN_MASK | CLASSD_MR_REN_MASK;
@@ -468,19 +429,17 @@ static int atmel_classd_codec_dai_trigger(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static const struct snd_soc_dai_ops atmel_classd_codec_dai_ops = {
-	.digital_mute	= atmel_classd_codec_dai_digital_mute,
-	.startup	= atmel_classd_codec_dai_startup,
-	.shutdown	= atmel_classd_codec_dai_shutdown,
-	.hw_params	= atmel_classd_codec_dai_hw_params,
-	.prepare	= atmel_classd_codec_dai_prepare,
-	.trigger	= atmel_classd_codec_dai_trigger,
+static const struct snd_soc_dai_ops atmel_classd_cpu_dai_ops = {
+	.startup        = atmel_classd_cpu_dai_startup,
+	.shutdown       = atmel_classd_cpu_dai_shutdown,
+	.mute_stream	= atmel_classd_cpu_dai_mute_stream,
+	.hw_params	= atmel_classd_cpu_dai_hw_params,
+	.prepare	= atmel_classd_cpu_dai_prepare,
+	.trigger	= atmel_classd_cpu_dai_trigger,
+	.no_capture_mute = 1,
 };
 
-#define ATMEL_CLASSD_CODEC_DAI_NAME  "atmel-classd-hifi"
-
-static struct snd_soc_dai_driver atmel_classd_codec_dai = {
-	.name = ATMEL_CLASSD_CODEC_DAI_NAME,
+static struct snd_soc_dai_driver atmel_classd_cpu_dai = {
 	.playback = {
 		.stream_name	= "Playback",
 		.channels_min	= 1,
@@ -488,7 +447,18 @@ static struct snd_soc_dai_driver atmel_classd_codec_dai = {
 		.rates		= ATMEL_CLASSD_RATES,
 		.formats	= SNDRV_PCM_FMTBIT_S16_LE,
 	},
-	.ops = &atmel_classd_codec_dai_ops,
+	.ops = &atmel_classd_cpu_dai_ops,
+};
+
+static const struct snd_soc_component_driver atmel_classd_cpu_dai_component = {
+	.name			= "atmel-classd",
+	.probe			= atmel_classd_component_probe,
+	.resume			= atmel_classd_component_resume,
+	.controls		= atmel_classd_snd_controls,
+	.num_controls		= ARRAY_SIZE(atmel_classd_snd_controls),
+	.idle_bias_on		= 1,
+	.use_pmdown_time	= 1,
+	.endianness		= 1,
 };
 
 /* ASoC sound card */
@@ -517,9 +487,9 @@ static int atmel_classd_asoc_card_init(struct device *dev,
 
 	dai_link->name			= "CLASSD";
 	dai_link->stream_name		= "CLASSD PCM";
-	dai_link->codecs->dai_name	= ATMEL_CLASSD_CODEC_DAI_NAME;
+	dai_link->codecs->dai_name	= "snd-soc-dummy-dai";
 	dai_link->cpus->dai_name	= dev_name(dev);
-	dai_link->codecs->name		= dev_name(dev);
+	dai_link->codecs->name		= "snd-soc-dummy";
 	dai_link->platforms->name	= dev_name(dev);
 
 	card->dai_link	= dai_link;
@@ -620,13 +590,6 @@ static int atmel_classd_probe(struct platform_device *pdev)
 		return ret;
 	}
 
-	ret = devm_snd_soc_register_component(dev, &soc_component_dev_classd,
-					&atmel_classd_codec_dai, 1);
-	if (ret) {
-		dev_err(dev, "could not register component: %d\n", ret);
-		return ret;
-	}
-
 	/* register sound card */
 	card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL);
 	if (!card) {

sound/soc/atmel/atmel-pcm-dma.c

@@ -44,7 +44,7 @@ static const struct snd_pcm_hardware atmel_pcm_dma_hardware = {
 	.buffer_bytes_max	= 512 * 1024,
 };
 
-/**
+/*
  * atmel_pcm_dma_irq: SSC interrupt handler for DMAENGINE enabled SSC
  *
  * We use DMAENGINE to send/receive data to/from SSC so this ISR is only to
@@ -53,7 +53,7 @@ static const struct snd_pcm_hardware atmel_pcm_dma_hardware = {
 static void atmel_pcm_dma_irq(u32 ssc_sr,
 	struct snd_pcm_substream *substream)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct atmel_pcm_dma_params *prtd;
 
 	prtd = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream);
@@ -78,7 +78,7 @@ static void atmel_pcm_dma_irq(u32 ssc_sr,
 static int atmel_pcm_configure_dma(struct snd_pcm_substream *substream,
 	struct snd_pcm_hw_params *params, struct dma_slave_config *slave_config)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct atmel_pcm_dma_params *prtd;
 	struct ssc_device *ssc;
 	int ret;

sound/soc/atmel/atmel-pcm-pdc.c

@@ -205,7 +205,7 @@ static int atmel_pcm_hw_params(struct snd_soc_component *component,
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct atmel_runtime_data *prtd = runtime->private_data;
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 
 	/* this may get called several times by oss emulation
 	 * with different params */

sound/soc/atmel/atmel-pdmic.c

@@ -104,7 +104,7 @@ static struct atmel_pdmic_pdata *atmel_pdmic_dt_init(struct device *dev)
 static int atmel_pdmic_cpu_dai_startup(struct snd_pcm_substream *substream,
 					struct snd_soc_dai *cpu_dai)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct atmel_pdmic *dd = snd_soc_card_get_drvdata(rtd->card);
 	int ret;
 
@@ -132,7 +132,7 @@ static int atmel_pdmic_cpu_dai_startup(struct snd_pcm_substream *substream,
 static void atmel_pdmic_cpu_dai_shutdown(struct snd_pcm_substream *substream,
 					struct snd_soc_dai *cpu_dai)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct atmel_pdmic *dd = snd_soc_card_get_drvdata(rtd->card);
 
 	/* Disable the overrun error interrupt */
@@ -145,34 +145,28 @@ static void atmel_pdmic_cpu_dai_shutdown(struct snd_pcm_substream *substream,
 static int atmel_pdmic_cpu_dai_prepare(struct snd_pcm_substream *substream,
 					struct snd_soc_dai *cpu_dai)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct atmel_pdmic *dd = snd_soc_card_get_drvdata(rtd->card);
+	struct snd_soc_component *component = cpu_dai->component;
 	u32 val;
+	int ret;
 
 	/* Clean the PDMIC Converted Data Register */
-	return regmap_read(dd->regmap, PDMIC_CDR, &val);
-}
-
-static const struct snd_soc_dai_ops atmel_pdmic_cpu_dai_ops = {
-	.startup	= atmel_pdmic_cpu_dai_startup,
-	.shutdown	= atmel_pdmic_cpu_dai_shutdown,
-	.prepare	= atmel_pdmic_cpu_dai_prepare,
-};
+	ret = regmap_read(dd->regmap, PDMIC_CDR, &val);
+	if (ret < 0)
+		return 0;
 
-#define ATMEL_PDMIC_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE)
+	ret = snd_soc_component_update_bits(component, PDMIC_CR,
+					    PDMIC_CR_ENPDM_MASK,
+					    PDMIC_CR_ENPDM_DIS <<
+					    PDMIC_CR_ENPDM_SHIFT);
+	if (ret < 0)
+		return ret;
 
-static struct snd_soc_dai_driver atmel_pdmic_cpu_dai = {
-	.capture = {
-		.channels_min	= 1,
-		.channels_max	= 1,
-		.rates		= SNDRV_PCM_RATE_KNOT,
-		.formats	= ATMEL_PDMIC_FORMATS,},
-	.ops = &atmel_pdmic_cpu_dai_ops,
-};
+	return 0;
+}
 
-static const struct snd_soc_component_driver atmel_pdmic_cpu_dai_component = {
-	.name = "atmel-pdmic",
-};
+#define ATMEL_PDMIC_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE)
 
 /* platform */
 #define ATMEL_PDMIC_MAX_BUF_SIZE  (64 * 1024)
@@ -197,7 +191,7 @@ atmel_pdmic_platform_configure_dma(struct snd_pcm_substream *substream,
 				struct snd_pcm_hw_params *params,
 				struct dma_slave_config *slave_config)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct atmel_pdmic *dd = snd_soc_card_get_drvdata(rtd->card);
 	int ret;
 
@@ -290,10 +284,10 @@ static int pdmic_get_mic_volsw(struct snd_kcontrol *kcontrol,
 	unsigned int dgain_val, scale_val;
 	int i;
 
-	dgain_val = (snd_soc_component_read32(component, PDMIC_DSPR1) & PDMIC_DSPR1_DGAIN_MASK)
+	dgain_val = (snd_soc_component_read(component, PDMIC_DSPR1) & PDMIC_DSPR1_DGAIN_MASK)
 		    >> PDMIC_DSPR1_DGAIN_SHIFT;
 
-	scale_val = (snd_soc_component_read32(component, PDMIC_DSPR0) & PDMIC_DSPR0_SCALE_MASK)
+	scale_val = (snd_soc_component_read(component, PDMIC_DSPR0) & PDMIC_DSPR0_SCALE_MASK)
 		    >> PDMIC_DSPR0_SCALE_SHIFT;
 
 	for (i = 0; i < ARRAY_SIZE(mic_gain_table); i++) {
@@ -355,27 +349,16 @@ static int atmel_pdmic_component_probe(struct snd_soc_component *component)
 	return 0;
 }
 
-static struct snd_soc_component_driver soc_component_dev_pdmic = {
-	.probe			= atmel_pdmic_component_probe,
-	.controls		= atmel_pdmic_snd_controls,
-	.num_controls		= ARRAY_SIZE(atmel_pdmic_snd_controls),
-	.idle_bias_on		= 1,
-	.use_pmdown_time	= 1,
-	.endianness		= 1,
-	.non_legacy_dai_naming	= 1,
-};
-
-/* codec dai component */
 #define PDMIC_MR_PRESCAL_MAX_VAL 127
 
 static int
-atmel_pdmic_codec_dai_hw_params(struct snd_pcm_substream *substream,
-			    struct snd_pcm_hw_params *params,
-			    struct snd_soc_dai *codec_dai)
+atmel_pdmic_cpu_dai_hw_params(struct snd_pcm_substream *substream,
+			      struct snd_pcm_hw_params *params,
+			      struct snd_soc_dai *cpu_dai)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct atmel_pdmic *dd = snd_soc_card_get_drvdata(rtd->card);
-	struct snd_soc_component *component = codec_dai->component;
+	struct snd_soc_component *component = cpu_dai->component;
 	unsigned int rate_min = substream->runtime->hw.rate_min;
 	unsigned int rate_max = substream->runtime->hw.rate_max;
 	int fs = params_rate(params);
@@ -445,21 +428,10 @@ atmel_pdmic_codec_dai_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int atmel_pdmic_codec_dai_prepare(struct snd_pcm_substream *substream,
-					struct snd_soc_dai *codec_dai)
-{
-	struct snd_soc_component *component = codec_dai->component;
-
-	snd_soc_component_update_bits(component, PDMIC_CR, PDMIC_CR_ENPDM_MASK,
-			    PDMIC_CR_ENPDM_DIS << PDMIC_CR_ENPDM_SHIFT);
-
-	return 0;
-}
-
-static int atmel_pdmic_codec_dai_trigger(struct snd_pcm_substream *substream,
-					int cmd, struct snd_soc_dai *codec_dai)
+static int atmel_pdmic_cpu_dai_trigger(struct snd_pcm_substream *substream,
+				       int cmd, struct snd_soc_dai *cpu_dai)
 {
-	struct snd_soc_component *component = codec_dai->component;
+	struct snd_soc_component *component = cpu_dai->component;
 	u32 val;
 
 	switch (cmd) {
@@ -482,16 +454,16 @@ static int atmel_pdmic_codec_dai_trigger(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static const struct snd_soc_dai_ops atmel_pdmic_codec_dai_ops = {
-	.hw_params	= atmel_pdmic_codec_dai_hw_params,
-	.prepare	= atmel_pdmic_codec_dai_prepare,
-	.trigger	= atmel_pdmic_codec_dai_trigger,
+static const struct snd_soc_dai_ops atmel_pdmic_cpu_dai_ops = {
+	.startup	= atmel_pdmic_cpu_dai_startup,
+	.shutdown	= atmel_pdmic_cpu_dai_shutdown,
+	.prepare	= atmel_pdmic_cpu_dai_prepare,
+	.hw_params	= atmel_pdmic_cpu_dai_hw_params,
+	.trigger	= atmel_pdmic_cpu_dai_trigger,
 };
 
-#define ATMEL_PDMIC_CODEC_DAI_NAME  "atmel-pdmic-hifi"
 
-static struct snd_soc_dai_driver atmel_pdmic_codec_dai = {
-	.name = ATMEL_PDMIC_CODEC_DAI_NAME,
+static struct snd_soc_dai_driver atmel_pdmic_cpu_dai = {
 	.capture = {
 		.stream_name	= "Capture",
 		.channels_min	= 1,
@@ -499,7 +471,17 @@ static struct snd_soc_dai_driver atmel_pdmic_codec_dai = {
 		.rates		= SNDRV_PCM_RATE_KNOT,
 		.formats	= ATMEL_PDMIC_FORMATS,
 	},
-	.ops = &atmel_pdmic_codec_dai_ops,
+	.ops = &atmel_pdmic_cpu_dai_ops,
+};
+
+static const struct snd_soc_component_driver atmel_pdmic_cpu_dai_component = {
+	.name			= "atmel-pdmic",
+	.probe			= atmel_pdmic_component_probe,
+	.controls		= atmel_pdmic_snd_controls,
+	.num_controls		= ARRAY_SIZE(atmel_pdmic_snd_controls),
+	.idle_bias_on		= 1,
+	.use_pmdown_time	= 1,
+	.endianness		= 1,
 };
 
 /* ASoC sound card */
@@ -528,9 +510,9 @@ static int atmel_pdmic_asoc_card_init(struct device *dev,
 
 	dai_link->name			= "PDMIC";
 	dai_link->stream_name		= "PDMIC PCM";
-	dai_link->codecs->dai_name	= ATMEL_PDMIC_CODEC_DAI_NAME;
+	dai_link->codecs->dai_name	= "snd-soc-dummy-dai";
 	dai_link->cpus->dai_name	= dev_name(dev);
-	dai_link->codecs->name		= dev_name(dev);
+	dai_link->codecs->name		= "snd-soc-dummy";
 	dai_link->platforms->name	= dev_name(dev);
 
 	card->dai_link	= dai_link;
@@ -684,16 +666,6 @@ static int atmel_pdmic_probe(struct platform_device *pdev)
 		return ret;
 	}
 
-	/* register codec and codec dai */
-	atmel_pdmic_codec_dai.capture.rate_min = rate_min;
-	atmel_pdmic_codec_dai.capture.rate_max = rate_max;
-	ret = devm_snd_soc_register_component(dev, &soc_component_dev_pdmic,
-				     &atmel_pdmic_codec_dai, 1);
-	if (ret) {
-		dev_err(dev, "could not register component: %d\n", ret);
-		return ret;
-	}
-
 	/* register sound card */
 	card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL);
 	if (!card) {

sound/soc/atmel/atmel_ssc_dai.c

@@ -887,6 +887,7 @@ static int asoc_ssc_init(struct device *dev)
 
 /**
  * atmel_ssc_set_audio - Allocate the specified SSC for audio use.
+ * @ssc_id: SSD ID in [0, NUM_SSC_DEVICES[
  */
 int atmel_ssc_set_audio(int ssc_id)
 {

sound/soc/atmel/atmel_wm8904.c

@@ -26,7 +26,7 @@ static const struct snd_soc_dapm_widget atmel_asoc_wm8904_dapm_widgets[] = {
 static int atmel_asoc_wm8904_hw_params(struct snd_pcm_substream *substream,
 		struct snd_pcm_hw_params *params)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	int ret;
 

sound/soc/au1x/db1200.c

@@ -94,7 +94,7 @@ static struct snd_soc_card db1550_ac97_machine = {
 
 static int db1200_i2s_startup(struct snd_pcm_substream *substream)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 
 	/* WM8731 has its own 12MHz crystal */

sound/soc/au1x/dbdma2.c

@@ -278,7 +278,7 @@ static int au1xpsc_pcm_open(struct snd_soc_component *component,
 			    struct snd_pcm_substream *substream)
 {
 	struct au1xpsc_audio_dmadata *pcd = to_dmadata(substream, component);
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	int stype = substream->stream, *dmaids;
 
 	dmaids = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream);

sound/soc/au1x/dma.c

@@ -191,7 +191,7 @@ static int alchemy_pcm_open(struct snd_soc_component *component,
 			    struct snd_pcm_substream *substream)
 {
 	struct alchemy_pcm_ctx *ctx = ss_to_ctx(substream, component);
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	int *dmaids, s = substream->stream;
 	char *name;
 

sound/soc/bcm/bcm2835-i2s.c

@@ -841,9 +841,12 @@ static int bcm2835_i2s_probe(struct platform_device *pdev)
 	dev->clk_prepared = false;
 	dev->clk = devm_clk_get(&pdev->dev, NULL);
 	if (IS_ERR(dev->clk)) {
-		dev_err(&pdev->dev, "could not get clk: %ld\n",
-			PTR_ERR(dev->clk));
-		return PTR_ERR(dev->clk);
+		ret = PTR_ERR(dev->clk);
+		if (ret == -EPROBE_DEFER)
+			dev_dbg(&pdev->dev, "could not get clk: %d\n", ret);
+		else
+			dev_err(&pdev->dev, "could not get clk: %d\n", ret);
+		return ret;
 	}
 
 	/* Request ioarea */

sound/soc/bcm/bcm63xx-pcm-whistler.c

@@ -45,7 +45,7 @@ static int bcm63xx_pcm_hw_params(struct snd_soc_component *component,
 				 struct snd_pcm_hw_params *params)
 {
 	struct i2s_dma_desc *dma_desc;
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct snd_pcm_runtime *runtime = substream->runtime;
 
 	snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
@@ -64,7 +64,7 @@ static int bcm63xx_pcm_hw_free(struct snd_soc_component *component,
 			struct snd_pcm_substream *substream)
 {
 	struct i2s_dma_desc	*dma_desc;
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 
 	dma_desc = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream);
 	kfree(dma_desc);
@@ -81,7 +81,7 @@ static int bcm63xx_pcm_trigger(struct snd_soc_component *component,
 	struct bcm_i2s_priv *i2s_priv;
 	struct regmap   *regmap_i2s;
 
-	rtd = substream->private_data;
+	rtd = asoc_substream_to_rtd(substream);
 	i2s_priv = dev_get_drvdata(asoc_rtd_to_cpu(rtd, 0)->dev);
 	regmap_i2s = i2s_priv->regmap_i2s;
 
@@ -148,7 +148,7 @@ static int bcm63xx_pcm_prepare(struct snd_soc_component *component,
 	struct i2s_dma_desc	*dma_desc;
 	struct regmap		*regmap_i2s;
 	struct bcm_i2s_priv	*i2s_priv;
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	uint32_t regaddr_desclen, regaddr_descaddr;
 
@@ -267,7 +267,7 @@ static irqreturn_t i2s_dma_isr(int irq, void *bcm_i2s_priv)
 	if (int_status & I2S_RX_DESC_OFF_INTR_EN_MSK) {
 		substream = i2s_priv->capture_substream;
 		runtime = substream->runtime;
-		rtd = substream->private_data;
+		rtd = asoc_substream_to_rtd(substream);
 		prtd = runtime->private_data;
 		dma_desc = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream);
 
@@ -315,7 +315,7 @@ static irqreturn_t i2s_dma_isr(int irq, void *bcm_i2s_priv)
 	if (int_status & I2S_TX_DESC_OFF_INTR_EN_MSK) {
 		substream = i2s_priv->play_substream;
 		runtime = substream->runtime;
-		rtd = substream->private_data;
+		rtd = asoc_substream_to_rtd(substream);
 		prtd = runtime->private_data;
 		dma_desc = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream);
 

sound/soc/bcm/cygnus-pcm.c

@@ -207,7 +207,7 @@ static u64 cygnus_dma_dmamask = DMA_BIT_MASK(32);
 static struct cygnus_aio_port *cygnus_dai_get_dma_data(
 				struct snd_pcm_substream *substream)
 {
-	struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
+	struct snd_soc_pcm_runtime *soc_runtime = asoc_substream_to_rtd(substream);
 
 	return snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(soc_runtime, 0), substream);
 }
@@ -353,7 +353,7 @@ static void enable_intr(struct snd_pcm_substream *substream)
 
 static void disable_intr(struct snd_pcm_substream *substream)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct cygnus_aio_port *aio;
 	u32 set_mask;
 
@@ -581,7 +581,7 @@ static irqreturn_t cygnus_dma_irq(int irq, void *data)
 static int cygnus_pcm_open(struct snd_soc_component *component,
 			   struct snd_pcm_substream *substream)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct cygnus_aio_port *aio;
 	int ret;
@@ -618,7 +618,7 @@ static int cygnus_pcm_open(struct snd_soc_component *component,
 static int cygnus_pcm_close(struct snd_soc_component *component,
 			    struct snd_pcm_substream *substream)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct cygnus_aio_port *aio;
 
 	aio = cygnus_dai_get_dma_data(substream);
@@ -640,7 +640,7 @@ static int cygnus_pcm_hw_params(struct snd_soc_component *component,
 				struct snd_pcm_substream *substream,
 				struct snd_pcm_hw_params *params)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct cygnus_aio_port *aio;
 
@@ -656,7 +656,7 @@ static int cygnus_pcm_hw_params(struct snd_soc_component *component,
 static int cygnus_pcm_hw_free(struct snd_soc_component *component,
 			      struct snd_pcm_substream *substream)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct cygnus_aio_port *aio;
 
 	aio = cygnus_dai_get_dma_data(substream);
@@ -669,7 +669,7 @@ static int cygnus_pcm_hw_free(struct snd_soc_component *component,
 static int cygnus_pcm_prepare(struct snd_soc_component *component,
 			      struct snd_pcm_substream *substream)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct cygnus_aio_port *aio;
 	unsigned long bufsize, periodsize;
@@ -733,7 +733,7 @@ static snd_pcm_uframes_t cygnus_pcm_pointer(struct snd_soc_component *component,
 static int cygnus_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
 {
 	struct snd_pcm_substream *substream = pcm->streams[stream].substream;
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct snd_dma_buffer *buf = &substream->dma_buffer;
 	size_t size;
 

sound/soc/cirrus/edb93xx.c

@@ -22,7 +22,7 @@
 static int edb93xx_hw_params(struct snd_pcm_substream *substream,
 			     struct snd_pcm_hw_params *params)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	int err;

sound/soc/cirrus/ep93xx-ac97.c

@@ -285,7 +285,7 @@ static int ep93xx_ac97_trigger(struct snd_pcm_substream *substream,
 			/*
 			 * As per Cirrus EP93xx errata described below:
 			 *
-			 * http://www.cirrus.com/en/pubs/errata/ER667E2B.pdf
+			 * https://www.cirrus.com/en/pubs/errata/ER667E2B.pdf
 			 *
 			 * we will wait for the TX FIFO to be empty before
 			 * clearing the TEN bit.

sound/soc/cirrus/snappercl15.c

@@ -22,7 +22,7 @@
 static int snappercl15_hw_params(struct snd_pcm_substream *substream,
 				 struct snd_pcm_hw_params *params)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	int err;

sound/soc/codecs/88pm860x-codec.c

@@ -274,10 +274,10 @@ static int snd_soc_get_volsw_2r_st(struct snd_kcontrol *kcontrol,
 	unsigned int reg2 = mc->rreg;
 	int val[2], val2[2], i;
 
-	val[0] = snd_soc_component_read32(component, reg) & 0x3f;
-	val[1] = (snd_soc_component_read32(component, PM860X_SIDETONE_SHIFT) >> 4) & 0xf;
-	val2[0] = snd_soc_component_read32(component, reg2) & 0x3f;
-	val2[1] = (snd_soc_component_read32(component, PM860X_SIDETONE_SHIFT)) & 0xf;
+	val[0] = snd_soc_component_read(component, reg) & 0x3f;
+	val[1] = (snd_soc_component_read(component, PM860X_SIDETONE_SHIFT) >> 4) & 0xf;
+	val2[0] = snd_soc_component_read(component, reg2) & 0x3f;
+	val2[1] = (snd_soc_component_read(component, PM860X_SIDETONE_SHIFT)) & 0xf;
 
 	for (i = 0; i < ARRAY_SIZE(st_table); i++) {
 		if ((st_table[i].m == val[0]) && (st_table[i].n == val[1]))
@@ -333,8 +333,8 @@ static int snd_soc_get_volsw_2r_out(struct snd_kcontrol *kcontrol,
 	int max = mc->max, val, val2;
 	unsigned int mask = (1 << fls(max)) - 1;
 
-	val = snd_soc_component_read32(component, reg) >> shift;
-	val2 = snd_soc_component_read32(component, reg2) >> shift;
+	val = snd_soc_component_read(component, reg) >> shift;
+	val2 = snd_soc_component_read(component, reg2) >> shift;
 	ucontrol->value.integer.value[0] = (max - val) & mask;
 	ucontrol->value.integer.value[1] = (max - val2) & mask;
 
@@ -426,7 +426,7 @@ static int pm860x_dac_event(struct snd_soc_dapm_widget *w,
 			snd_soc_component_update_bits(component, PM860X_EAR_CTRL_2,
 					    RSYNC_CHANGE, RSYNC_CHANGE);
 			/* update dac */
-			data = snd_soc_component_read32(component, PM860X_DAC_EN_2);
+			data = snd_soc_component_read(component, PM860X_DAC_EN_2);
 			data &= ~dac;
 			if (!(data & (DAC_LEFT | DAC_RIGHT)))
 				data &= ~MODULATOR;
@@ -902,7 +902,7 @@ static const struct snd_soc_dapm_route pm860x_dapm_routes[] = {
  * Use MUTE_LEFT & MUTE_RIGHT to implement digital mute.
  * These bits can also be used to mute.
  */
-static int pm860x_digital_mute(struct snd_soc_dai *codec_dai, int mute)
+static int pm860x_mute_stream(struct snd_soc_dai *codec_dai, int mute, int direction)
 {
 	struct snd_soc_component *component = codec_dai->component;
 	int data = 0, mask = MUTE_LEFT | MUTE_RIGHT;
@@ -1136,17 +1136,19 @@ static int pm860x_set_bias_level(struct snd_soc_component *component,
 }
 
 static const struct snd_soc_dai_ops pm860x_pcm_dai_ops = {
-	.digital_mute	= pm860x_digital_mute,
+	.mute_stream	= pm860x_mute_stream,
 	.hw_params	= pm860x_pcm_hw_params,
 	.set_fmt	= pm860x_pcm_set_dai_fmt,
 	.set_sysclk	= pm860x_set_dai_sysclk,
+	.no_capture_mute = 1,
 };
 
 static const struct snd_soc_dai_ops pm860x_i2s_dai_ops = {
-	.digital_mute	= pm860x_digital_mute,
+	.mute_stream	= pm860x_mute_stream,
 	.hw_params	= pm860x_i2s_hw_params,
 	.set_fmt	= pm860x_i2s_set_dai_fmt,
 	.set_sysclk	= pm860x_set_dai_sysclk,
+	.no_capture_mute = 1,
 };
 
 #define PM860X_RATES	(SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |	\

sound/soc/codecs/Kconfig

@@ -115,7 +115,8 @@ config SND_SOC_ALL_CODECS
 	imply SND_SOC_MAX98925
 	imply SND_SOC_MAX98926
 	imply SND_SOC_MAX98927
-	imply SND_SOC_MAX98373
+	imply SND_SOC_MAX98373_I2C
+	imply SND_SOC_MAX98373_SDW
 	imply SND_SOC_MAX98390
 	imply SND_SOC_MAX9850
 	imply SND_SOC_MAX9860
@@ -868,8 +869,25 @@ config SND_SOC_MAX98927
 	depends on I2C
 
 config SND_SOC_MAX98373
+	tristate
+
+config SND_SOC_MAX98373_I2C
 	tristate "Maxim Integrated MAX98373 Speaker Amplifier"
 	depends on I2C
+	select SND_SOC_MAX98373
+
+config SND_SOC_MAX98373_SDW
+	tristate "Maxim Integrated MAX98373 Speaker Amplifier - SDW"
+	depends on SOUNDWIRE
+	select SND_SOC_MAX98373
+	select REGMAP_SOUNDWIRE
+	help
+	  Enable support for Maxim Integrated MAX98373 Soundwire
+	  amplifier. MAX98373 supports either the MIPI SoundWire
+	  compatible interface for audio and control data, or
+	  the PCM interface for audio data and a standard I2C
+	  interface for control data. Select this if MAX98373 is
+	  connected via soundwire.
 
 config SND_SOC_MAX98390
 	tristate "Maxim Integrated MAX98390 Speaker Amplifier"

sound/soc/codecs/Makefile

@@ -115,6 +115,8 @@ snd-soc-max98925-objs := max98925.o
 snd-soc-max98926-objs := max98926.o
 snd-soc-max98927-objs := max98927.o
 snd-soc-max98373-objs := max98373.o
+snd-soc-max98373-i2c-objs := max98373-i2c.o
+snd-soc-max98373-sdw-objs := max98373-sdw.o
 snd-soc-max98390-objs := max98390.o
 snd-soc-max9850-objs := max9850.o
 snd-soc-max9860-objs := max9860.o
@@ -418,6 +420,8 @@ obj-$(CONFIG_SND_SOC_MAX98925)	+= snd-soc-max98925.o
 obj-$(CONFIG_SND_SOC_MAX98926)	+= snd-soc-max98926.o
 obj-$(CONFIG_SND_SOC_MAX98927)	+= snd-soc-max98927.o
 obj-$(CONFIG_SND_SOC_MAX98373)	+= snd-soc-max98373.o
+obj-$(CONFIG_SND_SOC_MAX98373_I2C)   += snd-soc-max98373-i2c.o
+obj-$(CONFIG_SND_SOC_MAX98373_SDW)   += snd-soc-max98373-sdw.o
 obj-$(CONFIG_SND_SOC_MAX98390)	+= snd-soc-max98390.o
 obj-$(CONFIG_SND_SOC_MAX9850)	+= snd-soc-max9850.o
 obj-$(CONFIG_SND_SOC_MAX9860)	+= snd-soc-max9860.o