Merge tag 'asoc-v6.3' of...

Merge tag 'asoc-v6.3' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next

ASoC: Updates for v6.3

There's been quite a lot of activity this release, but not really
one big feature - lots of new devices, plus a lot of cleanup and
modernisation work spread throughout the subsystem:

 - More factoring out of common operations into helper functions
   by Morimoto-san.
 - DT schema conversons and stylistic nits.
 - Continued work on building out the new SOF IPC4 scheme.
 - Support for Awinc AT88395, Infineon PEB2466, Iron Device
   SMA1303, Mediatek MT8188, Realtek RT712, Renesas IDT821034,
   Samsung/Tesla FSD SoC I2S, and TI TAS5720A-Q1.

Documentation/devicetree/bindings/dsp/mediatek,mt8186-dsp.yaml

@@ -15,7 +15,9 @@ description: |
 
 properties:
   compatible:
-    const: mediatek,mt8186-dsp
+    enum:
+      - mediatek,mt8186-dsp
+      - mediatek,mt8188-dsp
 
   reg:
     items:

Documentation/devicetree/bindings/sound/adi,adau7002.txt

-Analog Devices ADAU7002 Stereo PDM-to-I2S/TDM Converter
-
-Required properties:
-
- - compatible: Must be "adi,adau7002"
-
-Optional properties:
-
- - IOVDD-supply: Phandle and specifier for the power supply providing the IOVDD
-	supply as covered in Documentation/devicetree/bindings/regulator/regulator.txt
-
-	If this property is not present it is assumed that the supply pin is
-	hardwired to always on.
-
-Example:
-	adau7002: pdm-to-i2s {
-		compatible = "adi,adau7002";
-		IOVDD-supply = <&supply>;
-	};

Documentation/devicetree/bindings/sound/adi,adau7002.yaml

+# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/adi,adau7002.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Analog Devices ADAU7002 Stereo PDM-to-I2S/TDM Converter
+
+maintainers:
+  - Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
+
+allOf:
+  - $ref: dai-common.yaml#
+
+properties:
+  compatible:
+    const: adi,adau7002
+
+  IOVDD-supply:
+    description:
+      IOVDD power supply, if skipped then it is assumed that the supply pin is
+      hardwired to always on.
+
+  wakeup-delay-ms:
+    description:
+      Delay after power up needed for device to settle.
+
+required:
+  - compatible
+
+unevaluatedProperties: false
+
+examples:
+  - |
+    audio-codec {
+        compatible = "adi,adau7002";
+        IOVDD-supply = <&pp1800_l15a>;
+        #sound-dai-cells = <0>;
+        wakeup-delay-ms = <80>;
+    };

Documentation/devicetree/bindings/sound/ak4613.yaml

@@ -25,6 +25,13 @@ properties:
   "#sound-dai-cells":
     const: 0
 
+  ports:
+    $ref: audio-graph-port.yaml#/definitions/ports
+
+  port:
+    $ref: audio-graph-port.yaml#
+    unevaluatedProperties: false
+
 patternProperties:
   "^asahi-kasei,in[1-2]-single-end$":
     description: Input Pin 1 - 2.

Documentation/devicetree/bindings/sound/amlogic,axg-fifo.txt

-* Amlogic Audio FIFO controllers
-
-Required properties:
-- compatible: 'amlogic,axg-toddr' or
-	      'amlogic,axg-toddr' or
-	      'amlogic,g12a-frddr' or
-	      'amlogic,g12a-toddr' or
-	      'amlogic,sm1-frddr' or
-	      'amlogic,sm1-toddr'
-- reg: physical base address of the controller and length of memory
-       mapped region.
-- interrupts: interrupt specifier for the fifo.
-- clocks: phandle to the fifo peripheral clock provided by the audio
-	  clock controller.
-- resets: list of reset phandle, one for each entry reset-names.
-- reset-names: should contain the following:
-  * "arb" : memory ARB line (required)
-  * "rst" : dedicated device reset line (optional)
-- #sound-dai-cells: must be 0.
-- amlogic,fifo-depth: The size of the controller's fifo in bytes. This
-  		      is useful for determining certain configuration such
-		      as the flush threshold of the fifo
-
-Example of FRDDR A on the A113 SoC:
-
-frddr_a: audio-controller@1c0 {
-	compatible = "amlogic,axg-frddr";
-	reg = <0x0 0x1c0 0x0 0x1c>;
-	#sound-dai-cells = <0>;
-	interrupts = <GIC_SPI 88 IRQ_TYPE_EDGE_RISING>;
-	clocks = <&clkc_audio AUD_CLKID_FRDDR_A>;
-	resets = <&arb AXG_ARB_FRDDR_A>;
-	fifo-depth = <512>;
-};

Documentation/devicetree/bindings/sound/amlogic,axg-fifo.yaml

+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/amlogic,axg-fifo.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Amlogic AXG Audio FIFO controllers
+
+maintainers:
+  - Jerome Brunet <jbrunet@baylibre.com>
+
+properties:
+  compatible:
+    oneOf:
+      - enum:
+          - amlogic,axg-toddr
+          - amlogic,axg-frddr
+      - items:
+          - enum:
+              - amlogic,g12a-toddr
+              - amlogic,sm1-toddr
+          - const: amlogic,axg-toddr
+      - items:
+          - enum:
+              - amlogic,g12a-frddr
+              - amlogic,sm1-frddr
+          - const: amlogic,axg-frddr
+
+  reg:
+    maxItems: 1
+
+  "#sound-dai-cells":
+    const: 0
+
+  clocks:
+    items:
+      - description: Peripheral clock
+
+  interrupts:
+    maxItems: 1
+
+  resets:
+    minItems: 1
+    maxItems: 2
+
+  reset-names:
+    minItems: 1
+    maxItems: 2
+
+  amlogic,fifo-depth:
+    $ref: /schemas/types.yaml#/definitions/uint32
+    description: Size of the controller's fifo in bytes
+
+required:
+  - compatible
+  - reg
+  - "#sound-dai-cells"
+  - clocks
+  - interrupts
+  - resets
+  - amlogic,fifo-depth
+
+allOf:
+  - $ref: dai-common.yaml#
+  - if:
+      properties:
+        compatible:
+          contains:
+            enum:
+              - amlogic,g12a-toddr
+              - amlogic,sm1-toddr
+              - amlogic,g12a-frddr
+              - amlogic,sm1-frddr
+
+    then:
+      properties:
+        resets:
+          minItems: 2
+        reset-names:
+          items:
+            - const: arb
+            - const: rst
+      required:
+        - reset-names
+
+    else:
+      properties:
+        resets:
+          maxItems: 1
+        reset-names:
+          const: arb
+
+unevaluatedProperties: false
+
+examples:
+  - |
+    #include <dt-bindings/clock/axg-audio-clkc.h>
+    #include <dt-bindings/interrupt-controller/irq.h>
+    #include <dt-bindings/interrupt-controller/arm-gic.h>
+    #include <dt-bindings/reset/amlogic,meson-axg-audio-arb.h>
+    #include <dt-bindings/reset/amlogic,meson-g12a-audio-reset.h>
+
+    audio-controller@1c0 {
+        compatible = "amlogic,g12a-frddr", "amlogic,axg-frddr";
+        reg = <0x1c0 0x1c>;
+        #sound-dai-cells = <0>;
+        clocks = <&clkc_audio AUD_CLKID_FRDDR_A>;
+        interrupts = <GIC_SPI 88 IRQ_TYPE_EDGE_RISING>;
+        resets = <&arb>, <&clkc_audio AUD_RESET_FRDDR_A>;
+        reset-names = "arb", "rst";
+        amlogic,fifo-depth = <512>;
+    };

Documentation/devicetree/bindings/sound/amlogic,axg-pdm.txt

-* Amlogic Audio PDM input
-
-Required properties:
-- compatible: 'amlogic,axg-pdm' or
-	      'amlogic,g12a-pdm' or
-	      'amlogic,sm1-pdm'
-- reg: physical base address of the controller and length of memory
-       mapped region.
-- clocks: list of clock phandle, one for each entry clock-names.
-- clock-names: should contain the following:
-  * "pclk"   : peripheral clock.
-  * "dclk"   : pdm digital clock
-  * "sysclk" : dsp system clock
-- #sound-dai-cells: must be 0.
-
-Optional property:
-- resets: phandle to the dedicated reset line of the pdm input.
-
-Example of PDM on the A113 SoC:
-
-pdm: audio-controller@ff632000 {
-	compatible = "amlogic,axg-pdm";
-	reg = <0x0 0xff632000 0x0 0x34>;
-	#sound-dai-cells = <0>;
-	clocks = <&clkc_audio AUD_CLKID_PDM>,
-		 <&clkc_audio AUD_CLKID_PDM_DCLK>,
-		 <&clkc_audio AUD_CLKID_PDM_SYSCLK>;
-	clock-names = "pclk", "dclk", "sysclk";
-};

Documentation/devicetree/bindings/sound/amlogic,axg-pdm.yaml

+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/amlogic,axg-pdm.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Amlogic Audio AXG PDM input
+
+maintainers:
+  - Jerome Brunet <jbrunet@baylibre.com>
+
+properties:
+  compatible:
+    oneOf:
+      - items:
+          - enum:
+              - amlogic,g12a-pdm
+              - amlogic,sm1-pdm
+          - const: amlogic,axg-pdm
+      - const: amlogic,axg-pdm
+
+  reg:
+    maxItems: 1
+
+  "#sound-dai-cells":
+    const: 0
+
+  clocks:
+    items:
+      - description: Peripheral clock
+      - description: PDM digital clock
+      - description: DSP system clock
+
+  clock-names:
+    items:
+      - const: pclk
+      - const: dclk
+      - const: sysclk
+
+  resets:
+    maxItems: 1
+
+required:
+  - compatible
+  - reg
+  - "#sound-dai-cells"
+  - clocks
+  - clock-names
+
+allOf:
+  - $ref: dai-common.yaml#
+
+  - if:
+      properties:
+        compatible:
+          contains:
+            enum:
+              - amlogic,g12a-pdm
+              - amlogic,sm1-pdm
+    then:
+      required:
+        - resets
+
+    else:
+      properties:
+        resets: false
+
+unevaluatedProperties: false
+
+examples:
+  - |
+    #include <dt-bindings/clock/axg-audio-clkc.h>
+
+    audio-controller@ff632000 {
+        compatible = "amlogic,axg-pdm";
+        reg = <0xff632000 0x34>;
+        #sound-dai-cells = <0>;
+        clocks = <&clkc_audio AUD_CLKID_PDM>,
+                 <&clkc_audio AUD_CLKID_PDM_DCLK>,
+                 <&clkc_audio AUD_CLKID_PDM_SYSCLK>;
+        clock-names = "pclk", "dclk", "sysclk";
+    };

Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.txt

-Amlogic AXG sound card:
-
-Required properties:
-
-- compatible: "amlogic,axg-sound-card"
-- model : User specified audio sound card name, one string
-
-Optional properties:
-
-- audio-aux-devs : List of phandles pointing to auxiliary devices
-- audio-widgets : Please refer to widgets.txt.
-- audio-routing : A list of the connections between audio components.
-
-Subnodes:
-
-- dai-link: Container for dai-link level properties and the CODEC
-	    sub-nodes. There should be at least one (and probably more)
-	    subnode of this type.
-
-Required dai-link properties:
-
-- sound-dai: phandle and port of the CPU DAI.
-
-Required TDM Backend dai-link properties:
-- dai-format : CPU/CODEC common audio format
-
-Optional TDM Backend dai-link properties:
-- dai-tdm-slot-rx-mask-{0,1,2,3}: Receive direction slot masks
-- dai-tdm-slot-tx-mask-{0,1,2,3}: Transmit direction slot masks
-				  When omitted, mask is assumed to have to no
-				  slots. A valid must have at one slot, so at
-				  least one these mask should be provided with
-				  an enabled slot.
-- dai-tdm-slot-num : Please refer to tdm-slot.txt.
-		     If omitted, slot number is set to accommodate the largest
-		     mask provided.
-- dai-tdm-slot-width : Please refer to tdm-slot.txt. default to 32 if omitted.
-- mclk-fs : Multiplication factor between stream rate and mclk
-
-Backend dai-link subnodes:
-
-- codec: dai-link representing backend links should have at least one subnode.
-	 One subnode for each codec of the dai-link.
-	 dai-link representing frontend links have no codec, therefore have no
-	 subnodes
-
-Required codec subnodes properties:
-
-- sound-dai: phandle and port of the CODEC DAI.
-
-Optional codec subnodes properties:
-
-- dai-tdm-slot-tx-mask : Please refer to tdm-slot.txt.
-- dai-tdm-slot-rx-mask : Please refer to tdm-slot.txt.
-
-Example:
-
-sound {
-	compatible = "amlogic,axg-sound-card";
-	model = "AXG-S420";
-	audio-aux-devs = <&tdmin_a>, <&tdmout_c>;
-	audio-widgets = "Line", "Lineout",
-			"Line", "Linein",
-			"Speaker", "Speaker1 Left",
-			"Speaker", "Speaker1 Right";
-			"Speaker", "Speaker2 Left",
-			"Speaker", "Speaker2 Right";
-	audio-routing = "TDMOUT_C IN 0", "FRDDR_A OUT 2",
-			"SPDIFOUT IN 0", "FRDDR_A OUT 3",
-			"TDM_C Playback", "TDMOUT_C OUT",
-			"TDMIN_A IN 2", "TDM_C Capture",
-			"TDMIN_A IN 5", "TDM_C Loopback",
-			"TODDR_A IN 0", "TDMIN_A OUT",
-			"Lineout", "Lineout AOUTL",
-			"Lineout", "Lineout AOUTR",
-			"Speaker1 Left", "SPK1 OUT_A",
-			"Speaker2 Left", "SPK2 OUT_A",
-			"Speaker1 Right", "SPK1 OUT_B",
-			"Speaker2 Right", "SPK2 OUT_B",
-			"Linein AINL", "Linein",
-			"Linein AINR", "Linein";
-
-	dai-link@0 {
-		sound-dai = <&frddr_a>;
-	};
-
-	dai-link@1 {
-		sound-dai = <&toddr_a>;
-	};
-
-	dai-link@2 {
-		sound-dai = <&tdmif_c>;
-		dai-format = "i2s";
-		dai-tdm-slot-tx-mask-2 = <1 1>;
-		dai-tdm-slot-tx-mask-3 = <1 1>;
-		dai-tdm-slot-rx-mask-1 = <1 1>;
-		mclk-fs = <256>;
-
-		codec@0 {
-			sound-dai = <&lineout>;
-		};
-
-		codec@1 {
-			sound-dai = <&speaker_amp1>;
-		};
-
-		codec@2 {
-			sound-dai = <&speaker_amp2>;
-		};
-
-		codec@3 {
-			sound-dai = <&linein>;
-		};
-
-	};
-
-	dai-link@3 {
-		sound-dai = <&spdifout>;
-
-		codec {
-			sound-dai = <&spdif_dit>;
-		};
-	};
-};

Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.yaml

+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/amlogic,axg-sound-card.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Amlogic AXG sound card
+
+maintainers:
+  - Jerome Brunet <jbrunet@baylibre.com>
+
+properties:
+  compatible:
+    const: amlogic,axg-sound-card
+
+  audio-aux-devs:
+    $ref: /schemas/types.yaml#/definitions/phandle-array
+    description: list of auxiliary devices
+
+  audio-routing:
+    $ref: /schemas/types.yaml#/definitions/non-unique-string-array
+    description:
+      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 source.
+
+  audio-widgets:
+    $ref: /schemas/types.yaml#/definitions/non-unique-string-array
+    description:
+      A list off component DAPM widget. Each entry is a pair of strings,
+      the first being the widget type, the second being the widget name
+
+  model:
+    $ref: /schemas/types.yaml#/definitions/string
+    description: User specified audio sound card name
+
+patternProperties:
+  "^dai-link-[0-9]+$":
+    type: object
+    additionalProperties: false
+    description:
+      Container for dai-link level properties and the CODEC sub-nodes.
+      There should be at least one (and probably more) subnode of this type
+
+    properties:
+      dai-format:
+        $ref: /schemas/types.yaml#/definitions/string
+        enum: [ i2s, left-j, dsp_a ]
+
+      dai-tdm-slot-num:
+        $ref: /schemas/types.yaml#/definitions/uint32
+        description:
+          Number of slots in use. If omitted, slot number is set to
+          accommodate the largest mask provided.
+        maximum: 32
+
+      dai-tdm-slot-width:
+        $ref: /schemas/types.yaml#/definitions/uint32
+        description: Width in bits for each slot
+        enum: [ 8, 16, 20, 24, 32 ]
+        default: 32
+
+      mclk-fs:
+        $ref: /schemas/types.yaml#/definitions/uint32
+        description:
+          Multiplication factor between the frame rate and master clock
+          rate
+
+      sound-dai:
+        maxItems: 1
+        description: phandle of the CPU DAI
+
+    patternProperties:
+      "^dai-tdm-slot-(t|r)x-mask-[0-3]$":
+        $ref: /schemas/types.yaml#/definitions/uint32-array
+        minItems: 1
+        maxItems: 32
+        description:
+          Transmit and receive cpu slot masks of each TDM lane
+          When omitted, mask is assumed to have to no slots. A valid
+          interface must have at least one slot, so at least one these
+          mask should be provided with an enabled slot.
+
+      "^codec(-[0-9]+)?$":
+        type: object
+        additionalProperties: false
+        description:
+          dai-link representing backend links should have at least one subnode.
+          One subnode for each codec of the dai-link. dai-link representing
+          frontend links have no codec, therefore have no subnodes
+
+        properties:
+          sound-dai:
+            maxItems: 1
+            description: phandle of the codec DAI
+
+        patternProperties:
+          "^dai-tdm-slot-(t|r)x-mask$":
+            $ref: /schemas/types.yaml#/definitions/uint32-array
+            minItems: 1
+            maxItems: 32
+            description: Transmit and receive codec slot masks
+
+        required:
+          - sound-dai
+
+    required:
+      - sound-dai
+
+required:
+  - model
+  - dai-link-0
+
+unevaluatedProperties: false
+
+examples:
+  - |
+    sound {
+        compatible = "amlogic,axg-sound-card";
+        model = "AXG-S420";
+        audio-aux-devs = <&tdmin_a>, <&tdmout_c>;
+        audio-widgets = "Line", "Lineout",
+                        "Line", "Linein",
+                        "Speaker", "Speaker1 Left",
+                        "Speaker", "Speaker1 Right",
+                        "Speaker", "Speaker2 Left",
+                        "Speaker", "Speaker2 Right";
+        audio-routing = "TDMOUT_C IN 0", "FRDDR_A OUT 2",
+                        "SPDIFOUT IN 0", "FRDDR_A OUT 3",
+                        "TDM_C Playback", "TDMOUT_C OUT",
+                        "TDMIN_A IN 2", "TDM_C Capture",
+                        "TDMIN_A IN 5", "TDM_C Loopback",
+                        "TODDR_A IN 0", "TDMIN_A OUT",
+                        "Lineout", "Lineout AOUTL",
+                        "Lineout", "Lineout AOUTR",
+                        "Speaker1 Left", "SPK1 OUT_A",
+                        "Speaker2 Left", "SPK2 OUT_A",
+                        "Speaker1 Right", "SPK1 OUT_B",
+                        "Speaker2 Right", "SPK2 OUT_B",
+                        "Linein AINL", "Linein",
+                        "Linein AINR", "Linein";
+
+        dai-link-0 {
+            sound-dai = <&frddr_a>;
+        };
+
+        dai-link-1 {
+            sound-dai = <&toddr_a>;
+        };
+
+        dai-link-2 {
+            sound-dai = <&tdmif_c>;
+            dai-format = "i2s";
+            dai-tdm-slot-tx-mask-2 = <1 1>;
+            dai-tdm-slot-tx-mask-3 = <1 1>;
+            dai-tdm-slot-rx-mask-1 = <1 1>;
+            mclk-fs = <256>;
+
+            codec-0 {
+                sound-dai = <&lineout>;
+            };
+
+            codec-1 {
+                sound-dai = <&speaker_amp1>;
+            };
+
+            codec-2 {
+                sound-dai = <&speaker_amp2>;
+            };
+
+            codec-3 {
+                sound-dai = <&linein>;
+            };
+        };
+
+        dai-link-3 {
+            sound-dai = <&spdifout>;
+
+            codec {
+                sound-dai = <&spdif_dit>;
+            };
+        };
+    };

Documentation/devicetree/bindings/sound/amlogic,axg-spdifin.txt

-* Amlogic Audio SPDIF Input
-
-Required properties:
-- compatible: 'amlogic,axg-spdifin' or
-	      'amlogic,g12a-spdifin' or
-	      'amlogic,sm1-spdifin'
-- interrupts: interrupt specifier for the spdif input.
-- clocks: list of clock phandle, one for each entry clock-names.
-- clock-names: should contain the following:
-  * "pclk" : peripheral clock.
-  * "refclk" : spdif input reference clock
-- #sound-dai-cells: must be 0.
-
-Optional property:
-- resets: phandle to the dedicated reset line of the spdif input.
-
-Example on the A113 SoC:
-
-spdifin: audio-controller@400 {
-	compatible = "amlogic,axg-spdifin";
-	reg = <0x0 0x400 0x0 0x30>;
-	#sound-dai-cells = <0>;
-	interrupts = <GIC_SPI 87 IRQ_TYPE_EDGE_RISING>;
-	clocks = <&clkc_audio AUD_CLKID_SPDIFIN>,
-		 <&clkc_audio AUD_CLKID_SPDIFIN_CLK>;
-	clock-names = "pclk", "refclk";
-};

Documentation/devicetree/bindings/sound/amlogic,axg-spdifin.yaml

+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/amlogic,axg-spdifin.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Amlogic Audio AXG SPDIF Input
+
+maintainers:
+  - Jerome Brunet <jbrunet@baylibre.com>
+
+properties:
+  compatible:
+    oneOf:
+      - const: amlogic,axg-spdifin
+      - items:
+          - enum:
+              - amlogic,g12a-spdifin
+              - amlogic,sm1-spdifin
+          - const: amlogic,axg-spdifin
+
+  reg:
+    maxItems: 1
+
+  "#sound-dai-cells":
+    const: 0
+
+  clocks:
+    items:
+      - description: Peripheral clock
+      - description: SPDIF input reference clock
+
+  clock-names:
+    items:
+      - const: pclk
+      - const: refclk
+
+  interrupts:
+    maxItems: 1
+
+  resets:
+    maxItems: 1
+
+required:
+  - compatible
+  - reg
+  - "#sound-dai-cells"
+  - clocks
+  - clock-names
+  - interrupts
+
+allOf:
+  - $ref: dai-common.yaml#
+
+  - if:
+      properties:
+        compatible:
+          contains:
+            enum:
+              - amlogic,g12a-spdifin
+              - amlogic,sm1-spdifin
+    then:
+      required:
+        - resets
+
+    else:
+      properties:
+        resets: false
+
+unevaluatedProperties: false
+
+examples:
+  - |
+    #include <dt-bindings/clock/axg-audio-clkc.h>
+    #include <dt-bindings/interrupt-controller/irq.h>
+    #include <dt-bindings/interrupt-controller/arm-gic.h>
+
+    audio-controller@400 {
+        compatible = "amlogic,axg-spdifin";
+        reg = <0x400 0x30>;
+        #sound-dai-cells = <0>;
+        interrupts = <GIC_SPI 87 IRQ_TYPE_EDGE_RISING>;
+        clocks = <&clkc_audio AUD_CLKID_SPDIFIN>,
+                 <&clkc_audio AUD_CLKID_SPDIFIN_CLK>;
+        clock-names = "pclk", "refclk";
+    };

Documentation/devicetree/bindings/sound/amlogic,axg-spdifout.txt

-* Amlogic Audio SPDIF Output
-
-Required properties:
-- compatible: 'amlogic,axg-spdifout' or
-	      'amlogic,g12a-spdifout' or
-	      'amlogic,sm1-spdifout'
-- clocks: list of clock phandle, one for each entry clock-names.
-- clock-names: should contain the following:
-  * "pclk" : peripheral clock.
-  * "mclk" : master clock
-- #sound-dai-cells: must be 0.
-
-Optional property:
-- resets: phandle to the dedicated reset line of the spdif output.
-
-Example on the A113 SoC:
-
-spdifout: audio-controller@480 {
-	compatible = "amlogic,axg-spdifout";
-	reg = <0x0 0x480 0x0 0x50>;
-	#sound-dai-cells = <0>;
-	clocks = <&clkc_audio AUD_CLKID_SPDIFOUT>,
-		 <&clkc_audio AUD_CLKID_SPDIFOUT_CLK>;
-	clock-names = "pclk", "mclk";
-};

Documentation/devicetree/bindings/sound/amlogic,axg-spdifout.yaml

+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/amlogic,axg-spdifout.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Amlogic Audio AXG SPDIF Output
+
+maintainers:
+  - Jerome Brunet <jbrunet@baylibre.com>
+
+properties:
+  compatible:
+    oneOf:
+      - const: amlogic,axg-spdifout
+      - items:
+          - enum:
+              - amlogic,g12a-spdifout
+              - amlogic,sm1-spdifout
+          - const: amlogic,axg-spdifout
+
+  reg:
+    maxItems: 1
+
+  "#sound-dai-cells":
+    const: 0
+
+  clocks:
+    items:
+      - description: Peripheral clock
+      - description: SPDIF output master clock
+
+  clock-names:
+    items:
+      - const: pclk
+      - const: mclk
+
+  resets:
+    maxItems: 1
+
+required:
+  - compatible
+  - reg
+  - "#sound-dai-cells"
+  - clocks
+  - clock-names
+
+allOf:
+  - $ref: dai-common.yaml#
+
+  - if:
+      properties:
+        compatible:
+          contains:
+            enum:
+              - amlogic,g12a-spdifout
+              - amlogic,sm1-spdifout
+    then:
+      required:
+        - resets
+
+    else:
+      properties:
+        resets: false
+
+unevaluatedProperties: false
+
+examples:
+  - |
+    #include <dt-bindings/clock/axg-audio-clkc.h>
+
+    audio-controller@480 {
+        compatible = "amlogic,axg-spdifout";
+        reg = <0x480 0x50>;
+        #sound-dai-cells = <0>;
+        clocks = <&clkc_audio AUD_CLKID_SPDIFOUT>,
+                 <&clkc_audio AUD_CLKID_SPDIFOUT_CLK>;
+        clock-names = "pclk", "mclk";
+    };

Documentation/devicetree/bindings/sound/amlogic,axg-tdm-formatters.txt

-* Amlogic Audio TDM formatters
-
-Required properties:
-- compatible: 'amlogic,axg-tdmin' or
-	      'amlogic,axg-tdmout' or
-	      'amlogic,g12a-tdmin' or
-	      'amlogic,g12a-tdmout' or
-	      'amlogic,sm1-tdmin' or
-	      'amlogic,sm1-tdmout
-- reg: physical base address of the controller and length of memory
-       mapped region.
-- clocks: list of clock phandle, one for each entry clock-names.
-- clock-names: should contain the following:
-  * "pclk"     : peripheral clock.
-  * "sclk"     : bit clock.
-  * "sclk_sel" : bit clock input multiplexer.
-  * "lrclk"    : sample clock
-  * "lrclk_sel": sample clock input multiplexer
-
-Optional property:
-- resets: phandle to the dedicated reset line of the tdm formatter.
-
-Example of TDMOUT_A on the S905X2 SoC:
-
-tdmout_a: audio-controller@500 {
-	compatible = "amlogic,axg-tdmout";
-	reg = <0x0 0x500 0x0 0x40>;
-	resets = <&clkc_audio AUD_RESET_TDMOUT_A>;
-	clocks = <&clkc_audio AUD_CLKID_TDMOUT_A>,
-		 <&clkc_audio AUD_CLKID_TDMOUT_A_SCLK>,
-		 <&clkc_audio AUD_CLKID_TDMOUT_A_SCLK_SEL>,
-		 <&clkc_audio AUD_CLKID_TDMOUT_A_LRCLK>,
-		 <&clkc_audio AUD_CLKID_TDMOUT_A_LRCLK>;
-	clock-names = "pclk", "sclk", "sclk_sel",
-		      "lrclk", "lrclk_sel";
-};

Documentation/devicetree/bindings/sound/amlogic,axg-tdm-formatters.yaml

+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/amlogic,axg-tdm-formatters.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Amlogic Audio AXG TDM formatters
+
+maintainers:
+  - Jerome Brunet <jbrunet@baylibre.com>
+
+properties:
+  compatible:
+    enum:
+      - amlogic,g12a-tdmout
+      - amlogic,sm1-tdmout
+      - amlogic,axg-tdmout
+      - amlogic,g12a-tdmin
+      - amlogic,sm1-tdmin
+      - amlogic,axg-tdmin
+
+  clocks:
+    items:
+      - description: Peripheral clock
+      - description: Bit clock
+      - description: Bit clock input multiplexer
+      - description: Sample clock
+      - description: Sample clock input multiplexer
+
+  clock-names:
+    items:
+      - const: pclk
+      - const: sclk
+      - const: sclk_sel
+      - const: lrclk
+      - const: lrclk_sel
+
+  reg:
+    maxItems: 1
+
+  resets:
+    maxItems: 1
+
+required:
+  - compatible
+  - reg
+  - clocks
+  - clock-names
+
+allOf:
+  - $ref: component-common.yaml#
+
+  - if:
+      properties:
+        compatible:
+          contains:
+            enum:
+              - amlogic,g12a-tdmin
+              - amlogic,sm1-tdmin
+              - amlogic,g12a-tdmout
+              - amlogic,sm1-tdmout
+    then:
+      required:
+        - resets
+
+    else:
+      properties:
+        resets: false
+
+unevaluatedProperties: false
+
+examples:
+  - |
+    #include <dt-bindings/clock/axg-audio-clkc.h>
+    #include <dt-bindings/reset/amlogic,meson-g12a-audio-reset.h>
+
+    audio-controller@500 {
+        compatible = "amlogic,g12a-tdmout";
+        reg = <0x500 0x40>;
+        resets = <&clkc_audio AUD_RESET_TDMOUT_A>;
+        clocks = <&clkc_audio AUD_CLKID_TDMOUT_A>,
+                 <&clkc_audio AUD_CLKID_TDMOUT_A_SCLK>,
+                 <&clkc_audio AUD_CLKID_TDMOUT_A_SCLK_SEL>,
+                 <&clkc_audio AUD_CLKID_TDMOUT_A_LRCLK>,
+                 <&clkc_audio AUD_CLKID_TDMOUT_A_LRCLK>;
+        clock-names = "pclk", "sclk", "sclk_sel",
+                      "lrclk", "lrclk_sel";
+    };

Documentation/devicetree/bindings/sound/amlogic,axg-tdm-iface.txt

-* Amlogic Audio TDM Interfaces
-
-Required properties:
-- compatible: 'amlogic,axg-tdm-iface'
-- clocks: list of clock phandle, one for each entry clock-names.
-- clock-names: should contain the following:
-  * "sclk" : bit clock.
-  * "lrclk": sample clock
-  * "mclk" : master clock
-	     -> optional if the interface is in clock slave mode.
-- #sound-dai-cells: must be 0.
-
-Example of TDM_A on the A113 SoC:
-
-tdmif_a: audio-controller@0 {
-	compatible = "amlogic,axg-tdm-iface";
-	#sound-dai-cells = <0>;
-	clocks = <&clkc_audio AUD_CLKID_MST_A_MCLK>,
-		 <&clkc_audio AUD_CLKID_MST_A_SCLK>,
-		 <&clkc_audio AUD_CLKID_MST_A_LRCLK>;
-	clock-names = "mclk", "sclk", "lrclk";
-};

Documentation/devicetree/bindings/sound/amlogic,axg-tdm-iface.yaml

+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/amlogic,axg-tdm-iface.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Amlogic Audio TDM Interfaces
+
+maintainers:
+  - Jerome Brunet <jbrunet@baylibre.com>
+
+allOf:
+  - $ref: dai-common.yaml#
+
+properties:
+  compatible:
+    const: amlogic,axg-tdm-iface
+
+  "#sound-dai-cells":
+    const: 0
+
+  clocks:
+    minItems: 2
+    items:
+      - description: Bit clock
+      - description: Sample clock
+      - description: Master clock #optional
+
+  clock-names:
+    minItems: 2
+    items:
+      - const: sclk
+      - const: lrclk
+      - const: mclk
+
+required:
+  - compatible
+  - "#sound-dai-cells"
+  - clocks
+  - clock-names
+
+unevaluatedProperties: false
+
+examples:
+  - |
+    #include <dt-bindings/clock/axg-audio-clkc.h>
+
+    audio-controller {
+        compatible = "amlogic,axg-tdm-iface";
+        #sound-dai-cells = <0>;
+        clocks = <&clkc_audio AUD_CLKID_MST_A_SCLK>,
+                 <&clkc_audio AUD_CLKID_MST_A_LRCLK>,
+                 <&clkc_audio AUD_CLKID_MST_A_MCLK>;
+        clock-names = "sclk", "lrclk", "mclk";
+    };

Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml

@@ -62,7 +62,7 @@ patternProperties:
         description: phandle of the CPU DAI
 
     patternProperties:
-      "^codec-[0-9]+$":
+      "^codec(-[0-9]+)?$":
         type: object
         additionalProperties: false
         description: |-

Documentation/devicetree/bindings/sound/audio-graph-port.yaml

@@ -11,32 +11,24 @@ maintainers:
 
 select: false
 
-allOf:
-  - $ref: /schemas/graph.yaml#/$defs/port-base
-
-properties:
-  prefix:
-    description: "device name prefix"
-    $ref: /schemas/types.yaml#/definitions/string
-  convert-rate:
-    $ref: "/schemas/sound/dai-params.yaml#/$defs/dai-sample-rate"
-  convert-channels:
-    $ref: "/schemas/sound/dai-params.yaml#/$defs/dai-channels"
-  convert-sample-format:
-    $ref: "/schemas/sound/dai-params.yaml#/$defs/dai-sample-format"
+definitions:
+  port-base:
+    $ref: /schemas/graph.yaml#/$defs/port-base
+    properties:
+      convert-rate:
+        $ref: "/schemas/sound/dai-params.yaml#/$defs/dai-sample-rate"
+      convert-channels:
+        $ref: "/schemas/sound/dai-params.yaml#/$defs/dai-channels"
+      convert-sample-format:
+        $ref: "/schemas/sound/dai-params.yaml#/$defs/dai-sample-format"
+      mclk-fs:
+        $ref: "simple-card.yaml#/definitions/mclk-fs"
 
-patternProperties:
-  "^endpoint(@[0-9a-f]+)?":
+  endpoint-base:
     $ref: /schemas/graph.yaml#/$defs/endpoint-base
-    unevaluatedProperties: false
-
     properties:
       mclk-fs:
-        description: |
-          Multiplication factor between stream rate and codec mclk.
-          When defined, mclk-fs property defined in dai-link sub nodes are
-          ignored.
-        $ref: /schemas/types.yaml#/definitions/uint32
+        $ref: "simple-card.yaml#/definitions/mclk-fs"
       frame-inversion:
         description: dai-link uses frame clock inversion
         $ref: /schemas/types.yaml#/definitions/flag
@@ -53,6 +45,15 @@ patternProperties:
         oneOf:
           - $ref: /schemas/types.yaml#/definitions/flag
           - $ref: /schemas/types.yaml#/definitions/phandle
+      clocks:
+        description: Indicates system clock
+        $ref: /schemas/types.yaml#/definitions/phandle
+      system-clock-frequency:
+        $ref: "simple-card.yaml#/definitions/system-clock-frequency"
+      system-clock-direction-out:
+        $ref: "simple-card.yaml#/definitions/system-clock-direction-out"
+      system-clock-fixed:
+        $ref: "simple-card.yaml#/definitions/system-clock-fixed"
 
       dai-format:
         description: audio format.
@@ -100,4 +101,24 @@ patternProperties:
               minimum: 1
               maximum: 64
 
+  ports:
+    $ref: "#/definitions/port-base"
+    unevaluatedProperties: false
+    patternProperties:
+      "^port(@[0-9a-f]+)?$":
+        $ref: "#/definitions/port-base"
+        unevaluatedProperties: false
+        patternProperties:
+          "^endpoint(@[0-9a-f]+)?":
+            $ref: "#/definitions/endpoint-base"
+            unevaluatedProperties: false
+
+allOf:
+  - $ref: "#/definitions/port-base"
+
+patternProperties:
+  "^endpoint(@[0-9a-f]+)?":
+    $ref: "#/definitions/endpoint-base"
+    unevaluatedProperties: false
+
 additionalProperties: true

Documentation/devicetree/bindings/sound/awinic,aw88395.yaml

+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/awinic,aw88395.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Awinic AW88395 Smart Audio Amplifier
+
+maintainers:
+  - Weidong Wang <wangweidong.a@awinic.com>
+
+description:
+  The Awinic AW88395 is an I2S/TDM input, high efficiency
+  digital Smart K audio amplifier with an integrated 10.25V
+  smart boost convert.
+
+allOf:
+  - $ref: dai-common.yaml#
+
+properties:
+  compatible:
+    const: awinic,aw88395
+
+  reg:
+    maxItems: 1
+
+  '#sound-dai-cells':
+    const: 0
+
+  reset-gpios:
+    maxItems: 1
+
+required:
+  - compatible
+  - reg
+  - '#sound-dai-cells'
+  - reset-gpios
+
+unevaluatedProperties: false
+
+examples:
+  - |
+    #include <dt-bindings/gpio/gpio.h>
+    i2c {
+        #address-cells = <1>;
+        #size-cells = <0>;
+        audio-codec@34 {
+            compatible = "awinic,aw88395";
+            reg = <0x34>;
+            #sound-dai-cells = <0>;
+            reset-gpios = <&gpio 10 GPIO_ACTIVE_LOW>;
+        };
+    };

Documentation/devicetree/bindings/sound/cirrus,cs35l41.yaml

@@ -22,6 +22,9 @@ properties:
   reg:
     maxItems: 1
 
+  interrupts:
+    maxItems: 1
+
   '#sound-dai-cells':
     description:
       The first cell indicating the audio interface.
@@ -42,7 +45,7 @@ properties:
       Configures the peak current by monitoring the current through the boost FET.
       Range starts at 1600 mA and goes to a maximum of 4500 mA with increments
       of 50 mA. See section 4.3.6 of the datasheet for details.
-    $ref: "/schemas/types.yaml#/definitions/uint32"
+    $ref: /schemas/types.yaml#/definitions/uint32
     minimum: 1600
     maximum: 4500
     default: 4500
@@ -51,7 +54,7 @@ properties:
     description:
       Boost inductor value, expressed in nH. Valid
       values include 1000, 1200, 1500 and 2200.
-    $ref: "/schemas/types.yaml#/definitions/uint32"
+    $ref: /schemas/types.yaml#/definitions/uint32
     minimum: 1000
     maximum: 2200
 
@@ -60,7 +63,7 @@ properties:
       Total equivalent boost capacitance on the VBST
       and VAMP pins, derated at 11 volts DC. The value must be rounded to the
       nearest integer and expressed in uF.
-    $ref: "/schemas/types.yaml#/definitions/uint32"
+    $ref: /schemas/types.yaml#/definitions/uint32
 
   cirrus,asp-sdout-hiz:
     description:
@@ -70,7 +73,7 @@ properties:
       1 = Hi-Z during unused slots but logic 0 while all transmit channels disabled
       2 = (Default) Logic 0 during unused slots, but Hi-Z while all transmit channels disabled
       3 = Hi-Z during unused slots and while all transmit channels disabled
-    $ref: "/schemas/types.yaml#/definitions/uint32"
+    $ref: /schemas/types.yaml#/definitions/uint32
     minimum: 0
     maximum: 3
     default: 2
@@ -84,7 +87,7 @@ properties:
       enable boost voltage.
       0 = Internal Boost
       1 = External Boost
-    $ref: "/schemas/types.yaml#/definitions/uint32"
+    $ref: /schemas/types.yaml#/definitions/uint32
     minimum: 0
     maximum: 1
 
@@ -109,7 +112,7 @@ properties:
       1 = GPIO
       2 = Sync
       3 = MCLK input
-    $ref: "/schemas/types.yaml#/definitions/uint32"
+    $ref: /schemas/types.yaml#/definitions/uint32
     minimum: 0
     maximum: 3
 
@@ -136,7 +139,7 @@ properties:
       3 = MCLK input
       4 = Push-pull INTB (active low)
       5 = Push-pull INT (active high)
-    $ref: "/schemas/types.yaml#/definitions/uint32"
+    $ref: /schemas/types.yaml#/definitions/uint32
     minimum: 0
     maximum: 5
 
@@ -176,21 +179,23 @@ unevaluatedProperties: false
 
 examples:
   - |
+    #include <dt-bindings/gpio/gpio.h>
+
     spi {
         #address-cells = <1>;
         #size-cells = <0>;
 
-        cs35l41: cs35l41@2 {
-          #sound-dai-cells = <1>;
-          compatible = "cirrus,cs35l41";
-          reg = <2>;
-          VA-supply = <&dummy_vreg>;
-          VP-supply = <&dummy_vreg>;
-          reset-gpios = <&gpio 110 0>;
-
-          cirrus,boost-type = <0>;
-          cirrus,boost-peak-milliamp = <4500>;
-          cirrus,boost-ind-nanohenry = <1000>;
-          cirrus,boost-cap-microfarad = <15>;
+        cs35l41: speaker-amp@2 {
+            #sound-dai-cells = <1>;
+            compatible = "cirrus,cs35l41";
+            reg = <2>;
+            VA-supply = <&dummy_vreg>;
+            VP-supply = <&dummy_vreg>;
+            reset-gpios = <&gpio 110 GPIO_ACTIVE_HIGH>;
+
+            cirrus,boost-type = <0>;
+            cirrus,boost-peak-milliamp = <4500>;
+            cirrus,boost-ind-nanohenry = <1000>;
+            cirrus,boost-cap-microfarad = <15>;
         };
     };

Documentation/devicetree/bindings/sound/component-common.yaml

+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/component-common.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Audio Component Common Properties
+
+maintainers:
+  - Jerome Brunet <jbrunet@baylibre.com>
+
+properties:
+  sound-name-prefix:
+    $ref: /schemas/types.yaml#/definitions/string
+    description: |
+      Card implementing the routing property define the connection between
+      audio components as list of string pair. Component using the same
+      sink/source names may use this property to prepend the name of their
+      sinks/sources with the provided string.
+
+additionalProperties: true

Documentation/devicetree/bindings/sound/dai-common.yaml

@@ -9,15 +9,10 @@ title: Digital Audio Interface Common Properties
 maintainers:
   - Jerome Brunet <jbrunet@baylibre.com>
 
-properties:
-  sound-name-prefix:
-    $ref: /schemas/types.yaml#/definitions/string
-    description: |
-      Card implementing the routing property define the connection between
-      audio components as list of string pair. Component using the same
-      sink/source names may use this property to prepend the name of their
-      sinks/sources with the provided string.
+allOf:
+  - $ref: component-common.yaml#
 
+properties:
   '#sound-dai-cells': true
 
 additionalProperties: true

Documentation/devicetree/bindings/sound/fsl,sai.yaml

@@ -76,10 +76,14 @@ properties:
         minItems: 4
 
   dmas:
-    maxItems: 2
+    items:
+      - description: DMA controller phandle and request line for RX
+      - description: DMA controller phandle and request line for TX
 
   dma-names:
-    maxItems: 2
+    items:
+      - const: rx
+      - const: tx
 
   interrupts:
     items:
@@ -142,31 +146,6 @@ properties:
 
 allOf:
   - $ref: dai-common.yaml#
-  - if:
-      properties:
-        compatible:
-          contains:
-            const: fsl,vf610-sai
-    then:
-      properties:
-        dmas:
-          items:
-            - description: DMA controller phandle and request line for TX
-            - description: DMA controller phandle and request line for RX
-        dma-names:
-          items:
-            - const: tx
-            - const: rx
-    else:
-      properties:
-        dmas:
-          items:
-            - description: DMA controller phandle and request line for RX
-            - description: DMA controller phandle and request line for TX
-        dma-names:
-          items:
-            - const: rx
-            - const: tx
   - if:
       required:
         - fsl,sai-asynchronous
@@ -199,9 +178,8 @@ examples:
                  <&clks VF610_CLK_SAI2>,
                  <&clks 0>, <&clks 0>;
         clock-names = "bus", "mclk1", "mclk2", "mclk3";
-        dma-names = "tx", "rx";
-        dmas = <&edma0 0 21>,
-               <&edma0 0 20>;
+        dma-names = "rx", "tx";
+        dmas = <&edma0 0 20>, <&edma0 0 21>;
         big-endian;
         lsb-first;
     };

Documentation/devicetree/bindings/sound/fsl,xcvr.yaml

@@ -21,6 +21,7 @@ properties:
   compatible:
     enum:
       - fsl,imx8mp-xcvr
+      - fsl,imx93-xcvr
 
   reg:
     items:

Documentation/devicetree/bindings/sound/google,sc7280-herobrine.yaml

@@ -75,6 +75,18 @@ patternProperties:
 
         additionalProperties: false
 
+      platform:
+        description: Holds subnode which includes the phandle of q6apm platform device.
+        type: object
+        properties:
+          sound-dai:
+            maxItems: 1
+
+        required:
+          - sound-dai
+
+        additionalProperties: false
+
     required:
       - link-name
       - cpu

Documentation/devicetree/bindings/sound/infineon,peb2466.yaml

+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/infineon,peb2466.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Infineon PEB2466 codec
+
+maintainers:
+  - Herve Codina <herve.codina@bootlin.com>
+
+description: |
+  The Infineon PEB2466 codec is a programmable DSP-based four channels codec
+  with filters capabilities.
+
+  The time-slots used by the codec must be set and so, the properties
+  'dai-tdm-slot-num', 'dai-tdm-slot-width', 'dai-tdm-slot-tx-mask' and
+  'dai-tdm-slot-rx-mask' must be present in the sound card node for sub-nodes
+  that involve the codec. The codec uses one 8bit time-slot per channel.
+  'dai-tdm-tdm-slot-with' must be set to 8.
+
+  The PEB2466 codec also supports 28 gpios (signaling pins).
+
+allOf:
+  - $ref: /schemas/spi/spi-peripheral-props.yaml
+  - $ref: dai-common.yaml#
+
+properties:
+  compatible:
+    const: infineon,peb2466
+
+  reg:
+    description:
+      SPI device address.
+    maxItems: 1
+
+  clocks:
+    items:
+      - description: Master clock
+
+  clock-names:
+    items:
+      - const: mclk
+
+  spi-max-frequency:
+    maximum: 8192000
+
+  reset-gpios:
+    description:
+      GPIO used to reset the device.
+    maxItems: 1
+
+  firmware-name:
+    $ref: /schemas/types.yaml#/definitions/string
+    description:
+      Filters coefficients file to load. If this property is omitted, internal
+      filters are disabled.
+
+  '#sound-dai-cells':
+    const: 0
+
+  '#gpio-cells':
+    const: 2
+
+  gpio-controller: true
+
+required:
+  - compatible
+  - reg
+  - '#sound-dai-cells'
+  - gpio-controller
+  - '#gpio-cells'
+
+unevaluatedProperties: false
+
+examples:
+  - |
+    #include <dt-bindings/gpio/gpio.h>
+    spi {
+        #address-cells = <1>;
+        #size-cells = <0>;
+        audio-codec@0 {
+            compatible = "infineon,peb2466";
+            reg = <0>;
+            spi-max-frequency = <8192000>;
+            reset-gpios = <&gpio 10 GPIO_ACTIVE_LOW>;
+            #sound-dai-cells = <0>;
+            gpio-controller;
+            #gpio-cells = <2>;
+        };
+    };

Documentation/devicetree/bindings/sound/irondevice,sma1303.yaml

+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/irondevice,sma1303.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Iron Device SMA1303 Audio Amplifier
+
+maintainers:
+  - Kiseok Jo <kiseok.jo@irondevice.com>
+
+description:
+  SMA1303 digital class-D audio amplifier
+  with an integrated boost converter.
+
+allOf:
+  - $ref: dai-common.yaml#
+
+properties:
+  compatible:
+    enum:
+      - irondevice,sma1303
+
+  reg:
+    maxItems: 1
+
+  '#sound-dai-cells':
+    const: 1
+
+required:
+  - compatible
+  - reg
+  - '#sound-dai-cells'
+
+additionalProperties: false
+
+examples:
+  - |
+    i2c {
+        #address-cells = <1>;
+        #size-cells = <0>;
+
+        amplifier@1e {
+            compatible = "irondevice,sma1303";
+            reg = <0x1e>;
+            #sound-dai-cells = <1>;
+        };
+    };

Documentation/devicetree/bindings/sound/max98090.txt

-MAX98090 audio CODEC
-
-This device supports I2C only.
-
-Required properties:
-
-- compatible : "maxim,max98090" or "maxim,max98091".
-
-- reg : The I2C address of the device.
-
-- interrupts : The CODEC's interrupt output.
-
-Optional properties:
-
-- clocks: The phandle of the master clock to the CODEC
-
-- clock-names: Should be "mclk"
-
-- #sound-dai-cells : should be 0.
-
-- maxim,dmic-freq: Frequency at which to clock DMIC
-
-- maxim,micbias: Micbias voltage applies to the analog mic, valid voltages value are:
-	0 - 2.2v
-	1 - 2.55v
-	2 - 2.4v
-	3 - 2.8v
-
-Pins on the device (for linking into audio routes):
-
-  * MIC1
-  * MIC2
-  * DMICL
-  * DMICR
-  * IN1
-  * IN2
-  * IN3
-  * IN4
-  * IN5
-  * IN6
-  * IN12
-  * IN34
-  * IN56
-  * HPL
-  * HPR
-  * SPKL
-  * SPKR
-  * RCVL
-  * RCVR
-  * MICBIAS
-
-Example:
-
-audio-codec@10 {
-	compatible = "maxim,max98090";
-	reg = <0x10>;
-	interrupt-parent = <&gpio>;
-	interrupts = <TEGRA_GPIO(H, 4) IRQ_TYPE_LEVEL_HIGH>;
-};

Documentation/devicetree/bindings/sound/max98095.txt

-MAX98095 audio CODEC
-
-This device supports I2C only.
-
-Required properties:
-
-- compatible : "maxim,max98095".
-
-- reg : The I2C address of the device.
-
-Optional properties:
-
-- clocks: The phandle of the master clock to the CODEC
-
-- clock-names: Should be "mclk"
-
-Example:
-
-max98095: codec@11 {
-	compatible = "maxim,max98095";
-	reg = <0x11>;
-};

Documentation/devicetree/bindings/sound/maxim,max98090.yaml

+# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/maxim,max98090.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Maxim Integrated MAX98090/MAX98091 audio codecs
+
+maintainers:
+  - Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
+
+description: |
+  Pins on the device (for linking into audio routes):
+  MIC1, MIC2, DMICL, DMICR, IN1, IN2, IN3, IN4, IN5, IN6, IN12, IN34, IN56,
+  HPL, HPR, SPKL, SPKR, RCVL, RCVR, MICBIAS
+
+allOf:
+  - $ref: dai-common.yaml#
+
+properties:
+  compatible:
+    enum:
+      - maxim,max98090
+      - maxim,max98091
+
+  reg:
+    maxItems: 1
+
+  clocks:
+    items:
+      - description: master clock
+
+  clock-names:
+    items:
+      - const: mclk
+
+  interrupts:
+    maxItems: 1
+
+  maxim,dmic-freq:
+    $ref: /schemas/types.yaml#/definitions/uint32
+    default: 2500000
+    description:
+      DMIC clock frequency
+
+  maxim,micbias:
+    $ref: /schemas/types.yaml#/definitions/uint32
+    enum: [ 0, 1, 2, 3 ]
+    default: 3
+    description: |
+      Micbias voltage applied to the analog mic, valid voltages value are:
+        0 - 2.2v
+        1 - 2.55v
+        2 - 2.4v
+        3 - 2.8v
+
+  '#sound-dai-cells':
+    const: 0
+
+required:
+  - compatible
+  - reg
+  - interrupts
+
+unevaluatedProperties: false
+
+examples:
+  - |
+    #include <dt-bindings/interrupt-controller/irq.h>
+
+    i2c {
+        #address-cells = <1>;
+        #size-cells = <0>;
+
+        audio-codec@10 {
+            compatible = "maxim,max98090";
+            reg = <0x10>;
+            interrupt-parent = <&gpx3>;
+            interrupts = <2 IRQ_TYPE_EDGE_FALLING>;
+            clocks = <&i2s0 0>;
+            clock-names = "mclk";
+            #sound-dai-cells = <0>;
+        };
+    };

Documentation/devicetree/bindings/sound/maxim,max98095.yaml

+# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/maxim,max98095.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Maxim Integrated MAX98095 audio codec
+
+maintainers:
+  - Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
+
+allOf:
+  - $ref: dai-common.yaml#
+
+properties:
+  compatible:
+    enum:
+      - maxim,max98095
+
+  reg:
+    maxItems: 1
+
+  clocks:
+    items:
+      - description: master clock
+
+  clock-names:
+    items:
+      - const: mclk
+
+  '#sound-dai-cells':
+    const: 1
+
+required:
+  - compatible
+  - reg
+
+unevaluatedProperties: false
+
+examples:
+  - |
+    #include <dt-bindings/interrupt-controller/irq.h>
+
+    i2c {
+        #address-cells = <1>;
+        #size-cells = <0>;
+
+        audio-codec@11 {
+            compatible = "maxim,max98095";
+            reg = <0x11>;
+            clocks = <&i2s0 0>;
+            clock-names = "mclk";
+        };
+    };

Documentation/devicetree/bindings/sound/mediatek,mt8188-afe.yaml

+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/mediatek,mt8188-afe.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: MediaTek AFE PCM controller for mt8188
+
+maintainers:
+  - Trevor Wu <trevor.wu@mediatek.com>
+
+properties:
+  compatible:
+    const: mediatek,mt8188-afe
+
+  reg:
+    maxItems: 1
+
+  interrupts:
+    maxItems: 1
+
+  resets:
+    maxItems: 1
+
+  reset-names:
+    const: audiosys
+
+  mediatek,topckgen:
+    $ref: /schemas/types.yaml#/definitions/phandle
+    description: The phandle of the mediatek topckgen controller
+
+  power-domains:
+    maxItems: 1
+
+  clocks:
+    items:
+      - description: 26M clock
+      - description: audio pll1 clock
+      - description: audio pll2 clock
+      - description: clock divider for i2si1_mck
+      - description: clock divider for i2si2_mck
+      - description: clock divider for i2so1_mck
+      - description: clock divider for i2so2_mck
+      - description: clock divider for dptx_mck
+      - description: a1sys hoping clock
+      - description: audio intbus clock
+      - description: audio hires clock
+      - description: audio local bus clock
+      - description: mux for dptx_mck
+      - description: mux for i2so1_mck
+      - description: mux for i2so2_mck
+      - description: mux for i2si1_mck
+      - description: mux for i2si2_mck
+      - description: audio 26m clock
+
+  clock-names:
+    items:
+      - const: clk26m
+      - const: apll1
+      - const: apll2
+      - const: apll12_div0
+      - const: apll12_div1
+      - const: apll12_div2
+      - const: apll12_div3
+      - const: apll12_div9
+      - const: a1sys_hp_sel
+      - const: aud_intbus_sel
+      - const: audio_h_sel
+      - const: audio_local_bus_sel
+      - const: dptx_m_sel
+      - const: i2so1_m_sel
+      - const: i2so2_m_sel
+      - const: i2si1_m_sel
+      - const: i2si2_m_sel
+      - const: adsp_audio_26m
+
+  mediatek,etdm-in1-cowork-source:
+    $ref: /schemas/types.yaml#/definitions/uint32
+    description:
+      etdm modules can share the same external clock pin. Specify
+      which etdm clock source is required by this etdm in module.
+    enum:
+      - 1 # etdm2_in
+      - 2 # etdm1_out
+      - 3 # etdm2_out
+
+  mediatek,etdm-in2-cowork-source:
+    $ref: /schemas/types.yaml#/definitions/uint32
+    description:
+      etdm modules can share the same external clock pin. Specify
+      which etdm clock source is required by this etdm in module.
+    enum:
+      - 0 # etdm1_in
+      - 2 # etdm1_out
+      - 3 # etdm2_out
+
+  mediatek,etdm-out1-cowork-source:
+    $ref: /schemas/types.yaml#/definitions/uint32
+    description:
+      etdm modules can share the same external clock pin. Specify
+      which etdm clock source is required by this etdm out module.
+    enum:
+      - 0 # etdm1_in
+      - 1 # etdm2_in
+      - 3 # etdm2_out
+
+  mediatek,etdm-out2-cowork-source:
+    $ref: /schemas/types.yaml#/definitions/uint32
+    description:
+      etdm modules can share the same external clock pin. Specify
+      which etdm clock source is required by this etdm out module.
+    enum:
+      - 0 # etdm1_in
+      - 1 # etdm2_in
+      - 2 # etdm1_out
+
+patternProperties:
+  "^mediatek,etdm-in[1-2]-chn-disabled$":
+    $ref: /schemas/types.yaml#/definitions/uint8-array
+    minItems: 1
+    maxItems: 16
+    description:
+      This is a list of channel IDs which should be disabled.
+      By default, all data received from ETDM pins will be outputed to
+      memory. etdm in supports disable_out in direct mode(w/o interconn),
+      so user can disable the specified channels by the property.
+    uniqueItems: true
+    items:
+      minimum: 0
+      maximum: 15
+
+  "^mediatek,etdm-in[1-2]-multi-pin-mode$":
+    type: boolean
+    description: if present, the etdm data mode is I2S.
+
+  "^mediatek,etdm-out[1-3]-multi-pin-mode$":
+    type: boolean
+    description: if present, the etdm data mode is I2S.
+
+required:
+  - compatible
+  - reg
+  - interrupts
+  - resets
+  - reset-names
+  - mediatek,topckgen
+  - power-domains
+  - clocks
+  - clock-names
+
+additionalProperties: false
+
+examples:
+  - |
+    #include <dt-bindings/interrupt-controller/arm-gic.h>
+    #include <dt-bindings/interrupt-controller/irq.h>
+
+    afe@10b10000 {
+        compatible = "mediatek,mt8188-afe";
+        reg = <0x10b10000 0x10000>;
+        interrupts = <GIC_SPI 822 IRQ_TYPE_LEVEL_HIGH 0>;
+        resets = <&watchdog 14>;
+        reset-names = "audiosys";
+        mediatek,topckgen = <&topckgen>;
+        power-domains = <&spm 13>; //MT8188_POWER_DOMAIN_AUDIO
+        mediatek,etdm-in2-cowork-source = <2>;
+        mediatek,etdm-out2-cowork-source = <0>;
+        mediatek,etdm-in1-multi-pin-mode;
+        mediatek,etdm-in1-chn-disabled = /bits/ 8 <0x0 0x2>;
+        clocks = <&clk26m>,
+                 <&apmixedsys 9>, //CLK_APMIXED_APLL1
+                 <&apmixedsys 10>, //CLK_APMIXED_APLL2
+                 <&topckgen 186>, //CLK_TOP_APLL12_CK_DIV0
+                 <&topckgen 187>, //CLK_TOP_APLL12_CK_DIV1
+                 <&topckgen 188>, //CLK_TOP_APLL12_CK_DIV2
+                 <&topckgen 189>, //CLK_TOP_APLL12_CK_DIV3
+                 <&topckgen 191>, //CLK_TOP_APLL12_CK_DIV9
+                 <&topckgen 83>, //CLK_TOP_A1SYS_HP
+                 <&topckgen 31>, //CLK_TOP_AUD_INTBUS
+                 <&topckgen 32>, //CLK_TOP_AUDIO_H
+                 <&topckgen 69>, //CLK_TOP_AUDIO_LOCAL_BUS
+                 <&topckgen 81>, //CLK_TOP_DPTX
+                 <&topckgen 77>, //CLK_TOP_I2SO1
+                 <&topckgen 78>, //CLK_TOP_I2SO2
+                 <&topckgen 79>, //CLK_TOP_I2SI1
+                 <&topckgen 80>, //CLK_TOP_I2SI2
+                 <&adsp_audio26m 0>; //CLK_AUDIODSP_AUDIO26M
+        clock-names = "clk26m",
+                      "apll1",
+                      "apll2",
+                      "apll12_div0",
+                      "apll12_div1",
+                      "apll12_div2",
+                      "apll12_div3",
+                      "apll12_div9",
+                      "a1sys_hp_sel",
+                      "aud_intbus_sel",
+                      "audio_h_sel",
+                      "audio_local_bus_sel",
+                      "dptx_m_sel",
+                      "i2so1_m_sel",
+                      "i2so2_m_sel",
+                      "i2si1_m_sel",
+                      "i2si2_m_sel",
+                      "adsp_audio_26m";
+    };
+
+...

Documentation/devicetree/bindings/sound/mediatek,mt8188-mt6359.yaml

+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/mediatek,mt8188-mt6359.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: MediaTek MT8188 ASoC sound card
+
+maintainers:
+  - Trevor Wu <trevor.wu@mediatek.com>
+
+properties:
+  compatible:
+    const: mediatek,mt8188-mt6359-evb
+
+  model:
+    $ref: /schemas/types.yaml#/definitions/string
+    description: User specified audio sound card name
+
+  audio-routing:
+    $ref: /schemas/types.yaml#/definitions/non-unique-string-array
+    description:
+      A list of the connections between audio components. Each entry is a
+      sink/source pair of strings. Valid names could be the input or output
+      widgets of audio components, power supplies, MicBias of codec and the
+      software switch.
+
+  mediatek,platform:
+    $ref: /schemas/types.yaml#/definitions/phandle
+    description: The phandle of MT8188 ASoC platform.
+
+patternProperties:
+  "^dai-link-[0-9]+$":
+    type: object
+    description:
+      Container for dai-link level properties and CODEC sub-nodes.
+
+    properties:
+      link-name:
+        description:
+          This property corresponds to the name of the BE dai-link to which
+          we are going to update parameters in this node.
+        items:
+          enum:
+            - ADDA_BE
+            - DPTX_BE
+            - ETDM1_IN_BE
+            - ETDM2_IN_BE
+            - ETDM1_OUT_BE
+            - ETDM2_OUT_BE
+            - ETDM3_OUT_BE
+            - PCM1_BE
+
+      codec:
+        description: Holds subnode which indicates codec dai.
+        type: object
+        additionalProperties: false
+        properties:
+          sound-dai:
+            minItems: 1
+            maxItems: 2
+        required:
+          - sound-dai
+
+    additionalProperties: false
+
+    required:
+      - link-name
+      - codec
+
+additionalProperties: false
+
+required:
+  - compatible
+  - mediatek,platform
+
+examples:
+  - |
+    sound {
+        compatible = "mediatek,mt8188-mt6359-evb";
+        mediatek,platform = <&afe>;
+        pinctrl-names = "default";
+        pinctrl-0 = <&aud_pins_default>;
+        audio-routing =
+            "Headphone", "Headphone L",
+            "Headphone", "Headphone R",
+            "AIN1", "Headset Mic";
+        dai-link-0 {
+            link-name = "ETDM3_OUT_BE";
+
+            codec {
+                sound-dai = <&hdmi0>;
+            };
+        };
+    };
+
+...

Documentation/devicetree/bindings/sound/mchp,i2s-mcc.yaml → Documentation/devicetree/bindings/sound/microchip,sama7g5-i2smcc.yaml

 # SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause)
 %YAML 1.2
 ---
-$id: http://devicetree.org/schemas/sound/mchp,i2s-mcc.yaml#
+$id: http://devicetree.org/schemas/sound/microchip,sama7g5-i2smcc.yaml#
 $schema: http://devicetree.org/meta-schemas/core.yaml#
 
 title: Microchip I2S Multi-Channel Controller

Documentation/devicetree/bindings/sound/microchip,pdmc.yaml → Documentation/devicetree/bindings/sound/microchip,sama7g5-pdmc.yaml

 # SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause)
 %YAML 1.2
 ---
-$id: http://devicetree.org/schemas/sound/microchip,pdmc.yaml#
+$id: http://devicetree.org/schemas/sound/microchip,sama7g5-pdmc.yaml#
 $schema: http://devicetree.org/meta-schemas/core.yaml#
 
 title: Microchip Pulse Density Microphone Controller

Documentation/devicetree/bindings/sound/mchp,spdifrx.yaml → Documentation/devicetree/bindings/sound/microchip,sama7g5-spdifrx.yaml

 # SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause)
 %YAML 1.2
 ---
-$id: http://devicetree.org/schemas/sound/mchp,spdifrx.yaml#
+$id: http://devicetree.org/schemas/sound/microchip,sama7g5-spdifrx.yaml#
 $schema: http://devicetree.org/meta-schemas/core.yaml#
 
 title: Microchip S/PDIF Rx Controller

Documentation/devicetree/bindings/sound/mchp,spdiftx.yaml → Documentation/devicetree/bindings/sound/microchip,sama7g5-spdiftx.yaml

 # SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause)
 %YAML 1.2
 ---
-$id: http://devicetree.org/schemas/sound/mchp,spdiftx.yaml#
+$id: http://devicetree.org/schemas/sound/microchip,sama7g5-spdiftx.yaml#
 $schema: http://devicetree.org/meta-schemas/core.yaml#
 
 title: Microchip S/PDIF Tx Controller

Documentation/devicetree/bindings/sound/nau8822.txt

-NAU8822 audio CODEC
-
-This device supports I2C only.
-
-Required properties:
-
-  - compatible : "nuvoton,nau8822"
-
-  - reg : the I2C address of the device.
-
-Example:
-
-codec: nau8822@1a {
-	compatible = "nuvoton,nau8822";
-	reg = <0x1a>;
-};

Documentation/devicetree/bindings/sound/nuvoton,nau8822.yaml

+# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/nuvoton,nau8822.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: NAU8822 audio CODEC
+
+description: |
+  24 bit stereo audio codec with speaker driver.
+  This device supports I2C/SPI.
+
+maintainers:
+  - David Lin <CTLIN0@nuvoton.com>
+
+properties:
+  compatible:
+    enum:
+      - nuvoton,nau8822
+
+  reg:
+    maxItems: 1
+
+  nuvoton,spk-btl:
+    description:
+      If set, configure the two loudspeaker outputs as a Bridge Tied Load output
+      to drive a high power external loudspeaker.
+    $ref: /schemas/types.yaml#/definitions/flag
+
+required:
+  - compatible
+  - reg
+
+additionalProperties: false
+
+examples:
+  - |
+    i2c {
+        #address-cells = <1>;
+        #size-cells = <0>;
+
+        codec@1a {
+            compatible = "nuvoton,nau8822";
+            reg = <0x1a>;
+        };
+    };

Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml

@@ -35,7 +35,7 @@ properties:
 
   clocks:
     minItems: 3
-    maxItems: 7
+    maxItems: 10
 
   clock-names:
     minItems: 1
@@ -65,6 +65,9 @@ properties:
   power-domain-names:
     maxItems: 1
 
+  required-opps:
+    maxItems: 1
+
   '#sound-dai-cells':
     const: 1
 
@@ -75,7 +78,7 @@ properties:
     const: 0
 
 patternProperties:
-  "^dai-link@[0-9a-f]$":
+  "^dai-link@[0-9a-f]+$":
     type: object
     description: |
       LPASS CPU dai node for each I2S device or Soundwire device. Bindings of each node
@@ -121,6 +124,8 @@ allOf:
 
     then:
       properties:
+        clocks:
+          maxItems: 3
         clock-names:
           items:
             - const: ahbix-clk
@@ -135,6 +140,9 @@ allOf:
 
     then:
       properties:
+        clocks:
+          minItems: 7
+          maxItems: 7
         clock-names:
           items:
             - const: ahbix-clk
@@ -153,33 +161,31 @@ allOf:
 
     then:
       properties:
+        clocks:
+          minItems: 6
+          maxItems: 6
         clock-names:
-          oneOf:
-            - items:   #for I2S
-                - const: pcnoc-sway-clk
-                - const: audio-core
-                - const: mclk0
-                - const: pcnoc-mport-clk
-                - const: mi2s-bit-clk0
-                - const: mi2s-bit-clk1
-            - items:   #for HDMI
-                - const: pcnoc-sway-clk
-                - const: audio-core
-                - const: pcnoc-mport-clk
+          items:
+            - const: pcnoc-sway-clk
+            - const: audio-core
+            - const: mclk0
+            - const: pcnoc-mport-clk
+            - const: mi2s-bit-clk0
+            - const: mi2s-bit-clk1
+        reg:
+          minItems: 2
+          maxItems: 2
         reg-names:
-          anyOf:
-            - items:   #for I2S
-                - const: lpass-lpaif
-            - items:   #for I2S and HDMI
-                - const: lpass-hdmiif
-                - const: lpass-lpaif
+          items:
+            - const: lpass-hdmiif
+            - const: lpass-lpaif
+        interrupts:
+          minItems: 2
+          maxItems: 2
         interrupt-names:
-          anyOf:
-            - items:   #for I2S
-                - const: lpass-irq-lpaif
-            - items:   #for I2S and HDMI
-                - const: lpass-irq-lpaif
-                - const: lpass-irq-hdmi
+          items:
+            - const: lpass-irq-lpaif
+            - const: lpass-irq-hdmi
       required:
         - iommus
         - power-domains
@@ -192,54 +198,44 @@ allOf:
 
     then:
       properties:
+        clocks:
+          minItems: 10
+          maxItems: 10
         clock-names:
-          oneOf:
-            - items:   #for I2S
-                - const: aon_cc_audio_hm_h
-                - const: audio_cc_ext_mclk0
-                - const: core_cc_sysnoc_mport_core
-                - const: core_cc_ext_if0_ibit
-                - const: core_cc_ext_if1_ibit
-            - items:   #for Soundwire
-                - const: aon_cc_audio_hm_h
-                - const: audio_cc_codec_mem
-                - const: audio_cc_codec_mem0
-                - const: audio_cc_codec_mem1
-                - const: audio_cc_codec_mem2
-                - const: aon_cc_va_mem0
-            - items:   #for HDMI
-                - const: core_cc_sysnoc_mport_core
-
+          items:
+            - const: aon_cc_audio_hm_h
+            - const: audio_cc_ext_mclk0
+            - const: core_cc_sysnoc_mport_core
+            - const: core_cc_ext_if0_ibit
+            - const: core_cc_ext_if1_ibit
+            - const: audio_cc_codec_mem
+            - const: audio_cc_codec_mem0
+            - const: audio_cc_codec_mem1
+            - const: audio_cc_codec_mem2
+            - const: aon_cc_va_mem0
+        reg:
+          minItems: 6
+          maxItems: 6
         reg-names:
-          anyOf:
-            - items:   #for I2S
-                - const: lpass-lpaif
-            - items:   #for I2S and HDMI
-                - const: lpass-hdmiif
-                - const: lpass-lpaif
-            - items:   #for I2S, soundwire and HDMI
-                - const: lpass-hdmiif
-                - const: lpass-lpaif
-                - const: lpass-rxtx-cdc-dma-lpm
-                - const: lpass-rxtx-lpaif
-                - const: lpass-va-lpaif
-                - const: lpass-va-cdc-dma-lpm
+          items:
+            - const: lpass-hdmiif
+            - const: lpass-lpaif
+            - const: lpass-rxtx-cdc-dma-lpm
+            - const: lpass-rxtx-lpaif
+            - const: lpass-va-lpaif
+            - const: lpass-va-cdc-dma-lpm
+        interrupts:
+          minItems: 4
+          maxItems: 4
         interrupt-names:
-          anyOf:
-            - items:   #for I2S
-                - const: lpass-irq-lpaif
-            - items:   #for I2S and HDMI
-                - const: lpass-irq-lpaif
-                - const: lpass-irq-hdmi
-            - items:   #for I2S, soundwire and HDMI
-                - const: lpass-irq-lpaif
-                - const: lpass-irq-hdmi
-                - const: lpass-irq-vaif
-                - const: lpass-irq-rxtxif
+          items:
+            - const: lpass-irq-lpaif
+            - const: lpass-irq-hdmi
+            - const: lpass-irq-vaif
+            - const: lpass-irq-rxtxif
         power-domain-names:
-          allOf:
-            - items:
-                - const: lcx
+          items:
+            - const: lcx
 
       required:
         - iommus

Documentation/devicetree/bindings/sound/qcom,lpass-va-macro.yaml

@@ -36,7 +36,7 @@ properties:
     oneOf:
       - items:   #for ADSP based platforms
           - const: mclk
-          - const: core
+          - const: macro
           - const: dcodec
       - items:   #for ADSP bypass based platforms
           - const: mclk
@@ -77,7 +77,7 @@ examples:
       clocks = <&aoncc 0>,
                <&q6afecc LPASS_HW_MACRO_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>,
                <&q6afecc LPASS_HW_DCODEC_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>;
-      clock-names = "mclk", "core", "dcodec";
+      clock-names = "mclk", "macro", "dcodec";
       clock-output-names = "fsgen";
       qcom,dmic-sample-rate = <600000>;
       vdd-micb-supply = <&vreg_s4a_1p8>;

Documentation/devicetree/bindings/sound/qcom,q6apm-dai.yaml

@@ -17,7 +17,8 @@ properties:
     const: qcom,q6apm-dais
 
   iommus:
-    maxItems: 1
+    minItems: 1
+    maxItems: 2
 
 required:
   - compatible

Documentation/devicetree/bindings/sound/qcom,sm8250.yaml

@@ -15,16 +15,20 @@ description:
 
 properties:
   compatible:
-    enum:
-      - lenovo,yoga-c630-sndcard
-      - qcom,apq8016-sbc-sndcard
-      - qcom,db845c-sndcard
-      - qcom,msm8916-qdsp6-sndcard
-      - qcom,qrb5165-rb5-sndcard
-      - qcom,sc8280xp-sndcard
-      - qcom,sdm845-sndcard
-      - qcom,sm8250-sndcard
-      - qcom,sm8450-sndcard
+    oneOf:
+      - items:
+          - enum:
+              - lenovo,yoga-c630-sndcard
+              - qcom,db845c-sndcard
+          - const: qcom,sdm845-sndcard
+      - enum:
+          - qcom,apq8016-sbc-sndcard
+          - qcom,msm8916-qdsp6-sndcard
+          - qcom,qrb5165-rb5-sndcard
+          - qcom,sc8280xp-sndcard
+          - qcom,sdm845-sndcard
+          - qcom,sm8250-sndcard
+          - qcom,sm8450-sndcard
 
   audio-routing:
     $ref: /schemas/types.yaml#/definitions/non-unique-string-array

Documentation/devicetree/bindings/sound/qcom,wcd934x.yaml

@@ -28,7 +28,9 @@ properties:
     description: GPIO spec for reset line to use
     maxItems: 1
 
-  slim-ifc-dev: true
+  slim-ifc-dev:
+    description: IFC device interface
+    $ref: /schemas/types.yaml#/definitions/phandle
 
   clocks:
     maxItems: 1
@@ -147,21 +149,49 @@ patternProperties:
 required:
   - compatible
   - reg
-  - reset-gpios
-  - slim-ifc-dev
-  - interrupts
-  - interrupt-controller
-  - clock-frequency
-  - clock-output-names
-  - qcom,micbias1-microvolt
-  - qcom,micbias2-microvolt
-  - qcom,micbias3-microvolt
-  - qcom,micbias4-microvolt
-  - "#interrupt-cells"
-  - "#clock-cells"
-  - "#sound-dai-cells"
-  - "#address-cells"
-  - "#size-cells"
+
+allOf:
+  - if:
+      required:
+        - slim-ifc-dev
+    then:
+      required:
+        - reset-gpios
+        - slim-ifc-dev
+        - interrupt-controller
+        - clock-frequency
+        - clock-output-names
+        - qcom,micbias1-microvolt
+        - qcom,micbias2-microvolt
+        - qcom,micbias3-microvolt
+        - qcom,micbias4-microvolt
+        - "#interrupt-cells"
+        - "#clock-cells"
+        - "#sound-dai-cells"
+        - "#address-cells"
+        - "#size-cells"
+      oneOf:
+        - required:
+            - interrupts-extended
+        - required:
+            - interrupts
+    else:
+      properties:
+        reset-gpios: false
+        slim-ifc-dev: false
+        interrupts: false
+        interrupt-controller: false
+        clock-frequency: false
+        clock-output-names: false
+        qcom,micbias1-microvolt: false
+        qcom,micbias2-microvolt: false
+        qcom,micbias3-microvolt: false
+        qcom,micbias4-microvolt: false
+        "#interrupt-cells": false
+        "#clock-cells": false
+        "#sound-dai-cells": false
+        "#address-cells": false
+        "#size-cells": false
 
 additionalProperties: false
 

Documentation/devicetree/bindings/sound/qcom,wsa881x.yaml

@@ -15,6 +15,9 @@ description: |
   Their primary operating mode uses a SoundWire digital audio
   interface. This binding is for SoundWire interface.
 
+allOf:
+  - $ref: dai-common.yaml#
+
 properties:
   compatible:
     const: sdw10217201000
@@ -39,7 +42,7 @@ required:
   - "#thermal-sensor-cells"
   - "#sound-dai-cells"
 
-additionalProperties: false
+unevaluatedProperties: false
 
 examples:
   - |

Documentation/devicetree/bindings/sound/renesas,idt821034.yaml

+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/renesas,idt821034.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Renesas IDT821034 codec device
+
+maintainers:
+  - Herve Codina <herve.codina@bootlin.com>
+
+description: |
+  The IDT821034 codec is a four channel PCM codec with onchip filters and
+  programmable gain setting.
+
+  The time-slots used by the codec must be set and so, the properties
+  'dai-tdm-slot-num', 'dai-tdm-slot-width', 'dai-tdm-slot-tx-mask' and
+  'dai-tdm-slot-rx-mask' must be present in the ALSA sound card node for
+  sub-nodes that involve the codec. The codec uses one 8bit time-slot per
+  channel.
+  'dai-tdm-tdm-slot-with' must be set to 8.
+
+  The IDT821034 codec also supports 5 gpios (SLIC signals) per channel.
+
+allOf:
+  - $ref: /schemas/spi/spi-peripheral-props.yaml#
+  - $ref: dai-common.yaml#
+
+properties:
+  compatible:
+    const: renesas,idt821034
+
+  reg:
+    description:
+      SPI device address.
+    maxItems: 1
+
+  spi-max-frequency:
+    maximum: 8192000
+
+  spi-cpha: true
+
+  '#sound-dai-cells':
+    const: 0
+
+  '#gpio-cells':
+    const: 2
+
+  gpio-controller: true
+
+required:
+  - compatible
+  - reg
+  - spi-cpha
+  - '#sound-dai-cells'
+  - gpio-controller
+  - '#gpio-cells'
+
+unevaluatedProperties: false
+
+examples:
+  - |
+    spi {
+        #address-cells = <1>;
+        #size-cells = <0>;
+        audio-codec@0 {
+            compatible = "renesas,idt821034";
+            reg = <0>;
+            spi-max-frequency = <8192000>;
+            spi-cpha;
+            #sound-dai-cells = <0>;
+            gpio-controller;
+            #gpio-cells = <2>;
+        };
+    };

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

@@ -18,8 +18,7 @@ properties:
           - enum:
               - renesas,rcar_sound-r8a7778   # R-Car M1A
               - renesas,rcar_sound-r8a7779   # R-Car H1
-          - enum:
-              - renesas,rcar_sound-gen1
+          - const: renesas,rcar_sound-gen1
       # for Gen2 SoC
       - items:
           - enum:
@@ -32,8 +31,7 @@ properties:
               - renesas,rcar_sound-r8a7791   # R-Car M2-W
               - renesas,rcar_sound-r8a7793   # R-Car M2-N
               - renesas,rcar_sound-r8a7794   # R-Car E2
-          - enum:
-              - renesas,rcar_sound-gen2
+          - const: renesas,rcar_sound-gen2
       # for Gen3 SoC
       - items:
           - enum:
@@ -47,14 +45,16 @@ properties:
               - renesas,rcar_sound-r8a77965  # R-Car M3-N
               - renesas,rcar_sound-r8a77990  # R-Car E3
               - renesas,rcar_sound-r8a77995  # R-Car D3
-          - enum:
-              - renesas,rcar_sound-gen3
-      # for Generic
+          - const: renesas,rcar_sound-gen3
+      # for Gen4 SoC
       - items:
-          - enum:
-              - renesas,rcar_sound-gen1
-              - renesas,rcar_sound-gen2
-              - renesas,rcar_sound-gen3
+          - const: renesas,rcar_sound-r8a779g0  # R-Car V4H
+          - const: renesas,rcar_sound-gen4
+      # for Generic
+      - enum:
+          - renesas,rcar_sound-gen1
+          - renesas,rcar_sound-gen2
+          - renesas,rcar_sound-gen3
 
   reg:
     minItems: 1
@@ -68,6 +68,7 @@ properties:
     description: |
       it must be 0 if your system is using single DAI
       it must be 1 if your system is using multi  DAIs
+      This is used on simple-audio-card
     enum: [0, 1]
 
   "#clock-cells":
@@ -113,15 +114,34 @@ properties:
         - pattern: '^clk_(a|b|c|i)$'
 
   ports:
-    $ref: /schemas/graph.yaml#/properties/ports
+    $ref: audio-graph-port.yaml#/definitions/port-base
+    unevaluatedProperties: false
     patternProperties:
       '^port(@[0-9a-f]+)?$':
-        $ref: audio-graph-port.yaml#
+        $ref: audio-graph-port.yaml#/definitions/port-base
         unevaluatedProperties: false
+        patternProperties:
+          "^endpoint(@[0-9a-f]+)?":
+            $ref: audio-graph-port.yaml#/definitions/endpoint-base
+            properties:
+              playback:
+                $ref: /schemas/types.yaml#/definitions/phandle-array
+              capture:
+                $ref: /schemas/types.yaml#/definitions/phandle-array
+            unevaluatedProperties: false
 
   port:
-    $ref: audio-graph-port.yaml#
+    $ref: audio-graph-port.yaml#/definitions/port-base
     unevaluatedProperties: false
+    patternProperties:
+      "^endpoint(@[0-9a-f]+)?":
+        $ref: audio-graph-port.yaml#/definitions/endpoint-base
+        properties:
+          playback:
+            $ref: /schemas/types.yaml#/definitions/phandle-array
+          capture:
+            $ref: /schemas/types.yaml#/definitions/phandle-array
+        unevaluatedProperties: false
 
   rcar_sound,dvc:
     description: DVC subnode.
@@ -178,10 +198,6 @@ properties:
                   enum:
                     - tx
                     - rx
-        required:
-          - interrupts
-          - dmas
-          - dma-names
     additionalProperties: false
 
   rcar_sound,ssiu:
@@ -240,8 +256,6 @@ properties:
             $ref: /schemas/types.yaml#/definitions/flag
         required:
           - interrupts
-          - dmas
-          - dma-names
     additionalProperties: false
 
   # For DAI base
@@ -271,7 +285,6 @@ required:
   - reg-names
   - clocks
   - clock-names
-  - "#sound-dai-cells"
 
 allOf:
   - $ref: dai-common.yaml#
@@ -285,7 +298,6 @@ allOf:
         reg:
           maxItems: 3
         reg-names:
-          maxItems: 3
           items:
             enum:
               - scu
@@ -294,9 +306,8 @@ allOf:
     else:
       properties:
         reg:
-          maxItems: 5
+          minItems: 5
         reg-names:
-          maxItems: 5
           items:
             enum:
               - scu

Documentation/devicetree/bindings/sound/rt5640.txt

@@ -20,6 +20,9 @@ Optional properties:
 - realtek,in3-differential
   Boolean. Indicate MIC1/2/3 input are differential, rather than single-ended.
 
+- realtek,lout-differential
+  Boolean. Indicate LOUT output is differential, rather than stereo.
+
 - realtek,ldo1-en-gpios : The GPIO that controls the CODEC's LDO1_EN pin.
 
 - realtek,dmic1-data-pin

Documentation/devicetree/bindings/sound/samsung,odroid.yaml

@@ -43,9 +43,10 @@ properties:
     type: object
     properties:
       sound-dai:
+        minItems: 1
         items:
-          - description: phandle of the MAX98090 CODEC
           - description: phandle of the HDMI IP block node
+          - description: phandle of the MAX98090 CODEC
 
   samsung,audio-routing:
     $ref: /schemas/types.yaml#/definitions/non-unique-string-array

Documentation/devicetree/bindings/sound/samsung-i2s.yaml

@@ -37,12 +37,20 @@ properties:
       samsung,exynos7-i2s1: I2S1 on previous samsung platforms supports
       stereo channels. Exynos7 I2S1 upgraded to 5.1 multichannel with
       slightly modified bit offsets.
+
+      tesla,fsd-i2s: for 8/16/24bit stereo channel I2S for playback and
+      capture, secondary FIFO using external DMA, s/w reset control,
+      internal mux for root clock source with all root clock sampling
+      frequencies supported by Exynos7 I2S and 7.1 channel TDM support
+      for playback and capture TDM (Time division multiplexing) to allow
+      transfer of multiple channel audio data on single data line.
     enum:
       - samsung,s3c6410-i2s
       - samsung,s5pv210-i2s
       - samsung,exynos5420-i2s
       - samsung,exynos7-i2s
       - samsung,exynos7-i2s1
+      - tesla,fsd-i2s
 
   '#address-cells':
     const: 1
@@ -67,9 +75,6 @@ properties:
           - const: rx
           - const: tx-sec
 
-  assigned-clock-parents: true
-  assigned-clocks: true
-
   clocks:
     minItems: 1
     maxItems: 3

Documentation/devicetree/bindings/sound/simple-card.yaml

@@ -205,6 +205,8 @@ patternProperties:
     $ref: "#/definitions/dai"
   "^simple-audio-card,codec(@[0-9a-f]+)?$":
     $ref: "#/definitions/dai"
+  "^simple-audio-card,plat(@[0-9a-f]+)?$":
+    $ref: "#/definitions/dai"
 
   "^simple-audio-card,dai-link(@[0-9a-f]+)?$":
     description: |
@@ -215,6 +217,10 @@ patternProperties:
       reg:
         maxItems: 1
 
+      "#address-cells":
+        const: 1
+      "#size-cells":
+        const: 0
       # common properties
       frame-master:
         $ref: "#/definitions/frame-master"
@@ -244,9 +250,9 @@ patternProperties:
         maxItems: 1
 
     patternProperties:
-      "^cpu(@[0-9a-f]+)?":
+      "^cpu(-[0-9]+)?$":
         $ref: "#/definitions/dai"
-      "^codec(@[0-9a-f]+)?":
+      "^codec(-[0-9]+)?$":
         $ref: "#/definitions/dai"
     additionalProperties: false
 
@@ -462,16 +468,16 @@ examples:
 
             convert-channels = <8>; /* TDM Split */
 
-            sndcpu1: cpu0 {
+            sndcpu1: cpu-0 {
                 sound-dai = <&rcar_sound 1>;
             };
-            cpu1 {
+            cpu-1 {
                 sound-dai = <&rcar_sound 2>;
             };
-            cpu2 {
+            cpu-2 {
                 sound-dai = <&rcar_sound 3>;
             };
-            cpu3 {
+            cpu-3 {
                 sound-dai = <&rcar_sound 4>;
             };
             codec {

Documentation/devicetree/bindings/sound/tas5720.txt

@@ -6,11 +6,13 @@ audio playback. For more product information please see the links below:
 
 https://www.ti.com/product/TAS5720L
 https://www.ti.com/product/TAS5720M
+https://www.ti.com/product/TAS5720A-Q1
 https://www.ti.com/product/TAS5722L
 
 Required properties:
 
 - compatible : "ti,tas5720",
+               "ti,tas5720a-q1",
                "ti,tas5722"
 - reg : I2C slave address
 - dvdd-supply : phandle to a 3.3-V supply for the digital circuitry

Documentation/devicetree/bindings/sound/ti,pcm3168a.txt

-Texas Instruments pcm3168a DT bindings
-
-This driver supports both SPI and I2C bus access for this codec
-
-Required properties:
-
-  - compatible: "ti,pcm3168a"
-
-  - clocks : Contains an entry for each entry in clock-names
-
-  - clock-names : Includes the following entries:
-	"scki"	The system clock
-
-  - VDD1-supply : Digital power supply regulator 1 (+3.3V)
-
-  - VDD2-supply : Digital power supply regulator 2 (+3.3V)
-
-  - VCCAD1-supply : ADC power supply regulator 1 (+5V)
-
-  - VCCAD2-supply : ADC power supply regulator 2 (+5V)
-
-  - VCCDA1-supply : DAC power supply regulator 1 (+5V)
-
-  - VCCDA2-supply : DAC power supply regulator 2 (+5V)
-
-For required properties on SPI/I2C, consult SPI/I2C device tree documentation
-
-Optional properties:
-
-  - reset-gpios : Optional reset gpio line connected to RST pin of the codec.
-		  The RST line is low active:
-		  RST = low: device power-down
-		  RST = high: device is enabled
-
-Examples:
-
-i2c0: i2c0@0 {
-
-	...
-
-	pcm3168a: audio-codec@44 {
-		compatible = "ti,pcm3168a";
-		reg = <0x44>;
-		reset-gpios = <&gpio0 4 GPIO_ACTIVE_LOW>;
-		clocks = <&clk_core CLK_AUDIO>;
-		clock-names = "scki";
-		VDD1-supply = <&supply3v3>;
-		VDD2-supply = <&supply3v3>;
-		VCCAD1-supply = <&supply5v0>;
-		VCCAD2-supply = <&supply5v0>;
-		VCCDA1-supply = <&supply5v0>;
-		VCCDA2-supply = <&supply5v0>;
-		pinctrl-names = "default";
-		pinctrl-0 = <&dac_clk_pin>;
-	};
-};

Documentation/devicetree/bindings/sound/ti,pcm3168a.yaml

+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/ti,pcm3168a.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Texas Instruments PCM3168A Audio Codec
+
+maintainers:
+  - Damien Horsley <Damien.Horsley@imgtec.com>
+  - Geert Uytterhoeven <geert+renesas@glider.be>
+  - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+
+description:
+  The Texas Instruments PCM3168A is a 24-bit Multi-channel Audio CODEC with
+  96/192kHz sampling rate, supporting both SPI and I2C bus access.
+
+properties:
+  compatible:
+    const: ti,pcm3168a
+
+  reg:
+    maxItems: 1
+
+  clocks:
+    items:
+      - description: System clock input
+
+  clock-names:
+    items:
+      - const: scki
+
+  reset-gpios:
+    items:
+      - description: |
+          GPIO line connected to the active-low RST pin of the codec.
+            RST = low: device power-down
+            RST = high: device is enabled
+
+  "#sound-dai-cells":
+    enum: [0, 1]
+
+  VDD1-supply:
+    description: Digital power supply regulator 1 (+3.3V)
+
+  VDD2-supply:
+    description: Digital power supply regulator 2 (+3.3V)
+
+  VCCAD1-supply:
+    description: ADC power supply regulator 1 (+5V)
+
+  VCCAD2-supply:
+    description: ADC power supply regulator 2 (+5V)
+
+  VCCDA1-supply:
+    description: DAC power supply regulator 1 (+5V)
+
+  VCCDA2-supply:
+    description: DAC power supply regulator 2 (+5V)
+
+  ports:
+    $ref: audio-graph-port.yaml#/definitions/port-base
+    properties:
+      port@0:
+        $ref: audio-graph-port.yaml#
+        description: Audio input port.
+
+      port@1:
+        $ref: audio-graph-port.yaml#
+        description: Audio output port.
+
+required:
+  - compatible
+  - reg
+  - clocks
+  - clock-names
+  - VDD1-supply
+  - VDD2-supply
+  - VCCAD1-supply
+  - VCCAD2-supply
+  - VCCDA1-supply
+  - VCCDA2-supply
+
+additionalProperties: false
+
+examples:
+  - |
+    #include <dt-bindings/gpio/gpio.h>
+
+    i2c {
+        #address-cells = <1>;
+        #size-cells = <0>;
+
+        pcm3168a: audio-codec@44 {
+            compatible = "ti,pcm3168a";
+            reg = <0x44>;
+            reset-gpios = <&gpio0 4 GPIO_ACTIVE_LOW>;
+            clocks = <&clk_core 42>;
+            clock-names = "scki";
+            VDD1-supply = <&supply3v3>;
+            VDD2-supply = <&supply3v3>;
+            VCCAD1-supply = <&supply5v0>;
+            VCCAD2-supply = <&supply5v0>;
+            VCCDA1-supply = <&supply5v0>;
+            VCCDA2-supply = <&supply5v0>;
+        };
+    };

Documentation/devicetree/bindings/sound/ti,tlv320aic3x.yaml

+# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause
+# Copyright (C) 2022 Texas Instruments Incorporated
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/ti,tlv320aic3x.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Texas Instruments TLV320AIC3x Codec
+
+description: |
+  TLV320AIC3x are a series of low-power stereo audio codecs with stereo
+  headphone amplifier, as well as multiple inputs and outputs programmable in
+  single-ended or fully differential configurations.
+
+  The serial control bus supports SPI or I2C protocols, while the serial audio
+  data bus is programmable for I2S, left/right-justified, DSP, or TDM modes.
+
+  The following pins can be referred in the sound node's audio routing property:
+
+  CODEC output pins:
+     LLOUT
+     RLOUT
+     MONO_LOUT
+     HPLOUT
+     HPROUT
+     HPLCOM
+     HPRCOM
+
+  CODEC input pins for TLV320AIC3104:
+     MIC2L
+     MIC2R
+     LINE1L
+     LINE1R
+
+  CODEC input pins for other compatible codecs:
+     MIC3L
+     MIC3R
+     LINE1L
+     LINE2L
+     LINE1R
+     LINE2R
+
+maintainers:
+  - Jai Luthra <j-luthra@ti.com>
+
+properties:
+  compatible:
+    enum:
+      - ti,tlv320aic3x
+      - ti,tlv320aic33
+      - ti,tlv320aic3007
+      - ti,tlv320aic3106
+      - ti,tlv320aic3104
+
+  reg:
+    maxItems: 1
+
+  reset-gpios:
+    maxItems: 1
+    description:
+      GPIO specification for the active low RESET input.
+
+  gpio-reset:
+    maxItems: 1
+    description:
+      Deprecated, please use reset-gpios instead.
+    deprecated: true
+
+  ai3x-gpio-func:
+    description: AIC3X_GPIO1 & AIC3X_GPIO2 Functionality
+    $ref: /schemas/types.yaml#/definitions/uint32-array
+    maxItems: 2
+
+  ai3x-micbias-vg:
+    description: MicBias required voltage. If node is omitted then MicBias is powered down.
+    $ref: /schemas/types.yaml#/definitions/uint32
+    oneOf:
+      - const: 1
+        description: MICBIAS output is powered to 2.0V.
+      - const: 2
+        description: MICBIAS output is powered to 2.5V.
+      - const: 3
+        description: MICBIAS output is connected to AVDD.
+
+  ai3x-ocmv:
+    description: Output Common-Mode Voltage selection.
+    $ref: /schemas/types.yaml#/definitions/uint32
+    oneOf:
+      - const: 0
+        description: 1.35V
+      - const: 1
+        description: 1.5V
+      - const: 2
+        description: 1.65V
+      - const: 3
+        description: 1.8V
+
+  AVDD-supply:
+    description: Analog DAC voltage.
+
+  IOVDD-supply:
+    description: I/O voltage.
+
+  DRVDD-supply:
+    description: ADC analog and output driver voltage.
+
+  DVDD-supply:
+    description: Digital core voltage.
+
+  '#sound-dai-cells':
+    const: 0
+
+  clocks:
+    maxItems: 1
+
+  port:
+    $ref: audio-graph-port.yaml#
+    unevaluatedProperties: false
+
+required:
+  - compatible
+  - reg
+
+additionalProperties: false
+
+examples:
+  - |
+    #include <dt-bindings/gpio/gpio.h>
+    i2c {
+      #address-cells = <1>;
+      #size-cells = <0>;
+
+      tlv320aic3x_i2c: audio-codec@1b {
+        compatible = "ti,tlv320aic3x";
+        reg = <0x1b>;
+
+        reset-gpios = <&gpio1 17 GPIO_ACTIVE_LOW>;
+
+        AVDD-supply = <&regulator>;
+        IOVDD-supply = <&regulator>;
+        DRVDD-supply = <&regulator>;
+        DVDD-supply = <&regulator>;
+      };
+    };
+
+  - |
+    #include <dt-bindings/gpio/gpio.h>
+    spi {
+      #address-cells = <1>;
+      #size-cells = <0>;
+
+      tlv320aic3x_spi: audio-codec@0 {
+        compatible = "ti,tlv320aic3x";
+        reg = <0>; /* CS number */
+        #sound-dai-cells = <0>;
+
+        AVDD-supply = <&regulator>;
+        IOVDD-supply = <&regulator>;
+        DRVDD-supply = <&regulator>;
+        DVDD-supply = <&regulator>;
+        ai3x-ocmv = <0>;
+      };
+    };
+
+...

Documentation/devicetree/bindings/sound/tlv320aic3x.txt

-Texas Instruments - tlv320aic3x Codec module
-
-The tlv320aic3x serial control bus communicates through both I2C and SPI bus protocols
-
-Required properties:
-
-- compatible - "string" - One of:
-    "ti,tlv320aic3x" - Generic TLV320AIC3x device
-    "ti,tlv320aic33" - TLV320AIC33
-    "ti,tlv320aic3007" - TLV320AIC3007
-    "ti,tlv320aic3106" - TLV320AIC3106
-    "ti,tlv320aic3104" - TLV320AIC3104
-
-
-- reg - <int> -  I2C slave address
-
-
-Optional properties:
-
-- reset-gpios - GPIO specification for the active low RESET input.
-- ai3x-gpio-func - <array of 2 int> - AIC3X_GPIO1 & AIC3X_GPIO2 Functionality
-				    - Not supported on tlv320aic3104
-- ai3x-micbias-vg - MicBias Voltage required.
-	1 - MICBIAS output is powered to 2.0V,
-	2 - MICBIAS output is powered to 2.5V,
-	3 - MICBIAS output is connected to AVDD,
-	If this node is not mentioned or if the value is incorrect, then MicBias
-	is powered down.
-- ai3x-ocmv - Output Common-Mode Voltage selection:
-	0 - 1.35V,
-	1 - 1.5V,
-	2 - 1.65V,
-	3 - 1.8V
-- AVDD-supply, IOVDD-supply, DRVDD-supply, DVDD-supply : power supplies for the
-  device as covered in Documentation/devicetree/bindings/regulator/regulator.txt
-
-Deprecated properties:
-
-- gpio-reset - gpio pin number used for codec reset
-
-CODEC output pins:
-  * LLOUT
-  * RLOUT
-  * MONO_LOUT
-  * HPLOUT
-  * HPROUT
-  * HPLCOM
-  * HPRCOM
-
-CODEC input pins for TLV320AIC3104:
-  * MIC2L
-  * MIC2R
-  * LINE1L
-  * LINE1R
-
-CODEC input pins for other compatible codecs:
-  * MIC3L
-  * MIC3R
-  * LINE1L
-  * LINE2L
-  * LINE1R
-  * LINE2R
-
-The pins can be used in referring sound node's audio-routing property.
-
-I2C example:
-
-#include <dt-bindings/gpio/gpio.h>
-
-tlv320aic3x: tlv320aic3x@1b {
-	compatible = "ti,tlv320aic3x";
-	reg = <0x1b>;
-
-	reset-gpios = <&gpio1 17 GPIO_ACTIVE_LOW>;
-
-	AVDD-supply = <&regulator>;
-	IOVDD-supply = <&regulator>;
-	DRVDD-supply = <&regulator>;
-	DVDD-supply = <&regulator>;
-};
-
-SPI example:
-
-spi0: spi@f0000000 {
-	tlv320aic3x: codec@0 {
-		compatible = "ti,tlv320aic3x";
-		reg = <0>; /* CS number */
-		#sound-dai-cells = <0>;
-		spi-max-frequency = <1000000>;
-
-		AVDD-supply = <&regulator>;
-		IOVDD-supply = <&regulator>;
-		DRVDD-supply = <&regulator>;
-		DVDD-supply = <&regulator>;
-		ai3x-ocmv = <0>;
-	};
-};

Documentation/devicetree/bindings/vendor-prefixes.yaml

@@ -635,6 +635,8 @@ patternProperties:
     description: Inverse Path
   "^iom,.*":
     description: Iomega Corporation
+  "^irondevice,.*":
+    description: Iron Device Corporation
   "^isee,.*":
     description: ISEE 2007 S.L.
   "^isil,.*":

MAINTAINERS

@@ -10166,6 +10166,13 @@ L:	linux-iio@vger.kernel.org
 S:	Maintained
 F:	drivers/iio/pressure/dps310.c
 
+INFINEON PEB2466 ASoC CODEC
+M:	Herve Codina <herve.codina@bootlin.com>
+L:	alsa-devel@alsa-project.org (moderated for non-subscribers)
+S:	Maintained
+F:	Documentation/devicetree/bindings/sound/infineon,peb2466.yaml
+F:	sound/soc/codecs/peb2466.c
+
 INFINIBAND SUBSYSTEM
 M:	Jason Gunthorpe <jgg@nvidia.com>
 M:	Leon Romanovsky <leonro@nvidia.com>
@@ -10927,6 +10934,13 @@ M:	David Sterba <dsterba@suse.com>
 S:	Odd Fixes
 F:	drivers/tty/ipwireless/
 
+IRON DEVICE AUDIO CODEC DRIVERS
+M:	Kiseok Jo <kiseok.jo@irondevice.com>
+L:	alsa-devel@alsa-project.org (moderated for non-subscribers)
+S:	Maintained
+F:	Documentation/devicetree/bindings/sound/irondevice,*
+F:	sound/soc/codecs/sma*
+
 IRQ DOMAINS (IRQ NUMBER MAPPING LIBRARY)
 M:	Marc Zyngier <maz@kernel.org>
 S:	Maintained
@@ -17803,6 +17817,13 @@ F:	Documentation/devicetree/bindings/net/renesas,*.yaml
 F:	drivers/net/ethernet/renesas/
 F:	include/linux/sh_eth.h
 
+RENESAS IDT821034 ASoC CODEC
+M:	Herve Codina <herve.codina@bootlin.com>
+L:	alsa-devel@alsa-project.org (moderated for non-subscribers)
+S:	Maintained
+F:	Documentation/devicetree/bindings/sound/renesas,idt821034.yaml
+F:	sound/soc/codecs/idt821034.c
+
 RENESAS R-CAR GYROADC DRIVER
 M:	Marek Vasut <marek.vasut@gmail.com>
 L:	linux-iio@vger.kernel.org

drivers/soundwire/stream.c

@@ -469,7 +469,7 @@ static int sdw_prep_deprep_slave_ports(struct sdw_bus *bus,
 	}
 
 	/* Inform slave about the impending port prepare */
-	sdw_do_port_prep(s_rt, prep_ch, SDW_OPS_PORT_PRE_PREP);
+	sdw_do_port_prep(s_rt, prep_ch, prep ? SDW_OPS_PORT_PRE_PREP : SDW_OPS_PORT_PRE_DEPREP);
 
 	/* Prepare Slave port implementing CP_SM */
 	if (!dpn_prop->simple_ch_prep_sm) {
@@ -501,7 +501,7 @@ static int sdw_prep_deprep_slave_ports(struct sdw_bus *bus,
 	}
 
 	/* Inform slaves about ports prepared */
-	sdw_do_port_prep(s_rt, prep_ch, SDW_OPS_PORT_POST_PREP);
+	sdw_do_port_prep(s_rt, prep_ch, prep ? SDW_OPS_PORT_POST_PREP : SDW_OPS_PORT_POST_DEPREP);
 
 	/* Disable interrupt after Port de-prepare */
 	if (!prep && intr)

include/linux/platform_data/asoc-ux500-msp.h

-/* SPDX-License-Identifier: GPL-2.0-only */
-/*
- * Copyright (C) ST-Ericsson SA 2010
- *
- * Author: Rabin Vincent <rabin.vincent@stericsson.com> for ST-Ericsson
- */
-
-#ifndef __MSP_H
-#define __MSP_H
-
-#include <linux/platform_data/dma-ste-dma40.h>
-
-/* Platform data structure for a MSP I2S-device */
-struct msp_i2s_platform_data {
-	int id;
-	struct stedma40_chan_cfg *msp_i2s_dma_rx;
-	struct stedma40_chan_cfg *msp_i2s_dma_tx;
-};
-
-#endif

include/linux/soundwire/sdw.h

@@ -566,13 +566,15 @@ struct sdw_prepare_ch {
  * enum sdw_port_prep_ops: Prepare operations for Data Port
  *
  * @SDW_OPS_PORT_PRE_PREP: Pre prepare operation for the Port
- * @SDW_OPS_PORT_PREP: Prepare operation for the Port
+ * @SDW_OPS_PORT_PRE_DEPREP: Pre deprepare operation for the Port
  * @SDW_OPS_PORT_POST_PREP: Post prepare operation for the Port
+ * @SDW_OPS_PORT_POST_DEPREP: Post deprepare operation for the Port
  */
 enum sdw_port_prep_ops {
 	SDW_OPS_PORT_PRE_PREP = 0,
-	SDW_OPS_PORT_PREP = 1,
-	SDW_OPS_PORT_POST_PREP = 2,
+	SDW_OPS_PORT_PRE_DEPREP,
+	SDW_OPS_PORT_POST_PREP,
+	SDW_OPS_PORT_POST_DEPREP,
 };
 
 /**

include/sound/cs42l42.h

@@ -34,6 +34,7 @@
 #define CS42L42_PAGE_24		0x2400
 #define CS42L42_PAGE_25		0x2500
 #define CS42L42_PAGE_26		0x2600
+#define CS42L42_PAGE_27		0x2700
 #define CS42L42_PAGE_28		0x2800
 #define CS42L42_PAGE_29		0x2900
 #define CS42L42_PAGE_2A		0x2A00
@@ -720,6 +721,10 @@
 
 #define CS42L42_SRC_SDOUT_FS		(CS42L42_PAGE_26 + 0x09)
 
+/* Page 0x27 DMA */
+#define CS42L42_SOFT_RESET_REBOOT	(CS42L42_PAGE_27 + 0x01)
+#define CS42L42_SFT_RST_REBOOT_MASK	BIT(1)
+
 /* Page 0x28 S/PDIF Registers */
 #define CS42L42_SPDIF_CTL1		(CS42L42_PAGE_28 + 0x01)
 #define CS42L42_SPDIF_CTL2		(CS42L42_PAGE_28 + 0x02)

include/sound/simple_card_utils.h

@@ -69,6 +69,7 @@ struct asoc_simple_priv {
 	} *dai_props;
 	struct asoc_simple_jack hp_jack;
 	struct asoc_simple_jack mic_jack;
+	struct snd_soc_jack *aux_jacks;
 	struct snd_soc_dai_link *dai_link;
 	struct asoc_simple_dai *dais;
 	struct snd_soc_dai_link_component *dlcs;
@@ -187,6 +188,8 @@ 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, char *pin);
+int asoc_simple_init_aux_jacks(struct asoc_simple_priv *priv,
+				char *prefix);
 int asoc_simple_init_priv(struct asoc_simple_priv *priv,
 			       struct link_info *li);
 int asoc_simple_remove(struct platform_device *pdev);

include/sound/soc-component.h

@@ -98,6 +98,7 @@ struct snd_soc_component_driver {
 		       int source, unsigned int freq_in, unsigned int freq_out);
 	int (*set_jack)(struct snd_soc_component *component,
 			struct snd_soc_jack *jack,  void *data);
+	int (*get_jack_type)(struct snd_soc_component *component);
 
 	/* DT */
 	int (*of_xlate_dai_name)(struct snd_soc_component *component,
@@ -384,6 +385,7 @@ int snd_soc_component_set_pll(struct snd_soc_component *component, int pll_id,
 			      unsigned int freq_out);
 int snd_soc_component_set_jack(struct snd_soc_component *component,
 			       struct snd_soc_jack *jack, void *data);
+int snd_soc_component_get_jack_type(struct snd_soc_component *component);
 
 void snd_soc_component_seq_notifier(struct snd_soc_component *component,
 				    enum snd_soc_dapm_type type, int subseq);

include/sound/soc-dai.h

@@ -423,6 +423,16 @@ struct snd_soc_dai_driver {
 	int remove_order;
 };
 
+/* for Playback/Capture */
+struct snd_soc_dai_stream {
+	struct snd_soc_dapm_widget *widget;
+
+	unsigned int active;	/* usage count */
+	unsigned int tdm_mask;	/* CODEC TDM slot masks and params (for fixup) */
+
+	void *dma_data;		/* DAI DMA data */
+};
+
 /*
  * Digital Audio Interface runtime data.
  *
@@ -437,14 +447,7 @@ struct snd_soc_dai {
 	struct snd_soc_dai_driver *driver;
 
 	/* DAI runtime info */
-	unsigned int stream_active[SNDRV_PCM_STREAM_LAST + 1]; /* usage count */
-
-	struct snd_soc_dapm_widget *playback_widget;
-	struct snd_soc_dapm_widget *capture_widget;
-
-	/* DAI DMA data */
-	void *playback_dma_data;
-	void *capture_dma_data;
+	struct snd_soc_dai_stream stream[SNDRV_PCM_STREAM_LAST + 1];
 
 	/* Symmetry data - only valid if symmetry is being enforced */
 	unsigned int rate;
@@ -454,10 +457,6 @@ struct snd_soc_dai {
 	/* parent platform/codec */
 	struct snd_soc_component *component;
 
-	/* CODEC TDM slot masks and params (for fixup) */
-	unsigned int tx_mask;
-	unsigned int rx_mask;
-
 	struct list_head list;
 
 	/* function mark */
@@ -477,36 +476,59 @@ snd_soc_dai_get_pcm_stream(const struct snd_soc_dai *dai, int stream)
 		&dai->driver->playback : &dai->driver->capture;
 }
 
+#define snd_soc_dai_get_widget_playback(dai)	snd_soc_dai_get_widget(dai, SNDRV_PCM_STREAM_PLAYBACK)
+#define snd_soc_dai_get_widget_capture(dai)	snd_soc_dai_get_widget(dai, SNDRV_PCM_STREAM_CAPTURE)
 static inline
-struct snd_soc_dapm_widget *snd_soc_dai_get_widget(
-	struct snd_soc_dai *dai, int stream)
+struct snd_soc_dapm_widget *snd_soc_dai_get_widget(struct snd_soc_dai *dai, int stream)
 {
-	return (stream == SNDRV_PCM_STREAM_PLAYBACK) ?
-		dai->playback_widget : dai->capture_widget;
+	return dai->stream[stream].widget;
 }
 
-static inline void *snd_soc_dai_get_dma_data(const struct snd_soc_dai *dai,
-					     const struct snd_pcm_substream *ss)
+#define snd_soc_dai_set_widget_playback(dai, widget)	snd_soc_dai_set_widget(dai, SNDRV_PCM_STREAM_PLAYBACK, widget)
+#define snd_soc_dai_set_widget_capture(dai,  widget)	snd_soc_dai_set_widget(dai, SNDRV_PCM_STREAM_CAPTURE,  widget)
+static inline
+void snd_soc_dai_set_widget(struct snd_soc_dai *dai, int stream, struct snd_soc_dapm_widget *widget)
 {
-	return (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
-		dai->playback_dma_data : dai->capture_dma_data;
+	dai->stream[stream].widget = widget;
 }
 
-static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai *dai,
-					    const struct snd_pcm_substream *ss,
-					    void *data)
+#define snd_soc_dai_dma_data_get_playback(dai)	snd_soc_dai_dma_data_get(dai, SNDRV_PCM_STREAM_PLAYBACK)
+#define snd_soc_dai_dma_data_get_capture(dai)	snd_soc_dai_dma_data_get(dai, SNDRV_PCM_STREAM_CAPTURE)
+#define snd_soc_dai_get_dma_data(dai, ss)	snd_soc_dai_dma_data_get(dai, ss->stream)
+static inline void *snd_soc_dai_dma_data_get(const struct snd_soc_dai *dai, int stream)
 {
-	if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		dai->playback_dma_data = data;
-	else
-		dai->capture_dma_data = data;
+	return dai->stream[stream].dma_data;
+}
+
+#define snd_soc_dai_dma_data_set_playback(dai, data)	snd_soc_dai_dma_data_set(dai, SNDRV_PCM_STREAM_PLAYBACK, data)
+#define snd_soc_dai_dma_data_set_capture(dai,  data)	snd_soc_dai_dma_data_set(dai, SNDRV_PCM_STREAM_CAPTURE,  data)
+#define snd_soc_dai_set_dma_data(dai, ss, data)		snd_soc_dai_dma_data_set(dai, ss->stream, data)
+static inline void snd_soc_dai_dma_data_set(struct snd_soc_dai *dai, int stream, void *data)
+{
+	dai->stream[stream].dma_data = data;
 }
 
-static inline void snd_soc_dai_init_dma_data(struct snd_soc_dai *dai,
-					     void *playback, void *capture)
+static inline void snd_soc_dai_init_dma_data(struct snd_soc_dai *dai, void *playback, void *capture)
 {
-	dai->playback_dma_data = playback;
-	dai->capture_dma_data = capture;
+	snd_soc_dai_dma_data_set_playback(dai, playback);
+	snd_soc_dai_dma_data_set_capture(dai,  capture);
+}
+
+static inline unsigned int snd_soc_dai_tdm_mask_get(struct snd_soc_dai *dai, int stream)
+{
+	return dai->stream[stream].tdm_mask;
+}
+
+static inline void snd_soc_dai_tdm_mask_set(struct snd_soc_dai *dai, int stream,
+					    unsigned int tdm_mask)
+{
+	dai->stream[stream].tdm_mask = tdm_mask;
+}
+
+static inline unsigned int snd_soc_dai_stream_active(struct snd_soc_dai *dai, int stream)
+{
+	/* see snd_soc_dai_action() for setup */
+	return dai->stream[stream].active;
 }
 
 static inline void snd_soc_dai_set_drvdata(struct snd_soc_dai *dai,
@@ -561,10 +583,4 @@ static inline void *snd_soc_dai_get_stream(struct snd_soc_dai *dai,
 		return ERR_PTR(-ENOTSUPP);
 }
 
-static inline unsigned int
-snd_soc_dai_stream_active(struct snd_soc_dai *dai, int stream)
-{
-	return dai->stream_active[stream];
-}
-
 #endif

include/sound/soc-dapm.h

@@ -16,6 +16,7 @@
 #include <sound/asoc.h>
 
 struct device;
+struct snd_pcm_substream;
 struct snd_soc_pcm_runtime;
 struct soc_enum;
 

include/sound/soc-dpcm.h

@@ -162,6 +162,8 @@ int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream);
 int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir,
 	int event);
 bool dpcm_end_walk_at_be(struct snd_soc_dapm_widget *widget, enum snd_soc_dapm_direction dir);
+int widget_in_list(struct snd_soc_dapm_widget_list *list,
+		   struct snd_soc_dapm_widget *widget);
 
 #define dpcm_be_dai_startup_rollback(fe, stream, last)	\
 						dpcm_be_dai_stop(fe, stream, 0, last)

include/sound/soc-topology.h

@@ -62,7 +62,7 @@ struct snd_soc_dobj {
 	enum snd_soc_dobj_type type;
 	unsigned int index;	/* objects can belong in different groups */
 	struct list_head list;
-	struct snd_soc_tplg_ops *ops;
+	int (*unload)(struct snd_soc_component *comp, struct snd_soc_dobj *dobj);
 	union {
 		struct snd_soc_dobj_control control;
 		struct snd_soc_dobj_widget widget;

include/sound/soc.h

@@ -1052,6 +1052,12 @@ struct snd_soc_card {
 #define for_each_card_widgets_safe(card, w, _w)	\
 	list_for_each_entry_safe(w, _w, &card->widgets, list)
 
+
+static inline int snd_soc_card_is_instantiated(struct snd_soc_card *card)
+{
+	return card && card->instantiated;
+}
+
 /* SoC machine DAI configuration, glues a codec and cpu DAI together */
 struct snd_soc_pcm_runtime {
 	struct device *dev;

include/sound/sof/ipc4/header.h

@@ -185,6 +185,9 @@ enum sof_ipc4_pipeline_state {
 #define SOF_IPC4_GLB_PIPE_STATE_MASK		GENMASK(15, 0)
 #define SOF_IPC4_GLB_PIPE_STATE(x)		((x) << SOF_IPC4_GLB_PIPE_STATE_SHIFT)
 
+/* pipeline set state IPC msg extension */
+#define SOF_IPC4_GLB_PIPE_STATE_EXT_MULTI	BIT(0)
+
 /* load library ipc msg */
 #define SOF_IPC4_GLB_LOAD_LIBRARY_LIB_ID_SHIFT	16
 #define SOF_IPC4_GLB_LOAD_LIBRARY_LIB_ID(x)	((x) << SOF_IPC4_GLB_LOAD_LIBRARY_LIB_ID_SHIFT)

include/uapi/sound/intel/avs/tokens.h

@@ -108,6 +108,7 @@ enum avs_tplg_token {
 	AVS_TKN_MOD_CORE_ID_U8				= 1704,
 	AVS_TKN_MOD_PROC_DOMAIN_U8			= 1705,
 	AVS_TKN_MOD_MODCFG_EXT_ID_U32			= 1706,
+	AVS_TKN_MOD_KCONTROL_ID_U32			= 1707,
 
 	/* struct avs_tplg_path_template */
 	AVS_TKN_PATH_TMPL_ID_U32			= 1801,
@@ -121,6 +122,9 @@ enum avs_tplg_token {
 	AVS_TKN_PIN_FMT_INDEX_U32			= 2201,
 	AVS_TKN_PIN_FMT_IOBS_U32			= 2202,
 	AVS_TKN_PIN_FMT_AFMT_ID_U32			= 2203,
+
+	/* struct avs_tplg_kcontrol */
+	AVS_TKN_KCONTROL_ID_U32				= 2301,
 };
 
 #endif

sound/soc/amd/acp/acp-legacy-mach.c

@@ -16,6 +16,7 @@
 #include <sound/pcm_params.h>
 #include <sound/soc-acpi.h>
 #include <sound/soc-dapm.h>
+#include <linux/dmi.h>
 #include <linux/module.h>
 
 #include "acp-mach.h"
@@ -27,6 +28,7 @@ static struct acp_card_drvdata rt5682_rt1019_data = {
 	.hs_codec_id = RT5682,
 	.amp_codec_id = RT1019,
 	.dmic_codec_id = DMIC,
+	.tdm_mode = false,
 };
 
 static struct acp_card_drvdata rt5682s_max_data = {
@@ -36,6 +38,7 @@ static struct acp_card_drvdata rt5682s_max_data = {
 	.hs_codec_id = RT5682S,
 	.amp_codec_id = MAX98360A,
 	.dmic_codec_id = DMIC,
+	.tdm_mode = false,
 };
 
 static struct acp_card_drvdata rt5682s_rt1019_data = {
@@ -45,6 +48,7 @@ static struct acp_card_drvdata rt5682s_rt1019_data = {
 	.hs_codec_id = RT5682S,
 	.amp_codec_id = RT1019,
 	.dmic_codec_id = DMIC,
+	.tdm_mode = false,
 };
 
 static struct acp_card_drvdata max_nau8825_data = {
@@ -56,6 +60,7 @@ static struct acp_card_drvdata max_nau8825_data = {
 	.dmic_codec_id = DMIC,
 	.soc_mclk = true,
 	.platform = REMBRANDT,
+	.tdm_mode = false,
 };
 
 static struct acp_card_drvdata rt5682s_rt1019_rmb_data = {
@@ -67,6 +72,7 @@ static struct acp_card_drvdata rt5682s_rt1019_rmb_data = {
 	.dmic_codec_id = DMIC,
 	.soc_mclk = true,
 	.platform = REMBRANDT,
+	.tdm_mode = false,
 };
 
 static const struct snd_kcontrol_new acp_controls[] = {
@@ -90,6 +96,8 @@ static int acp_asoc_probe(struct platform_device *pdev)
 {
 	struct snd_soc_card *card = NULL;
 	struct device *dev = &pdev->dev;
+	const struct dmi_system_id *dmi_id;
+	struct acp_card_drvdata *acp_card_drvdata;
 	int ret;
 
 	if (!pdev->id_entry)
@@ -108,6 +116,11 @@ static int acp_asoc_probe(struct platform_device *pdev)
 	card->num_controls = ARRAY_SIZE(acp_controls);
 	card->drvdata = (struct acp_card_drvdata *)pdev->id_entry->driver_data;
 
+	acp_card_drvdata = card->drvdata;
+	dmi_id = dmi_first_match(acp_quirk_table);
+	if (dmi_id && dmi_id->driver_data)
+		acp_card_drvdata->tdm_mode = dmi_id->driver_data;
+
 	acp_legacy_dai_links_create(card);
 
 	ret = devm_snd_soc_register_card(&pdev->dev, card);

sound/soc/amd/acp/acp-mach.h

@@ -18,6 +18,8 @@
 #include <linux/module.h>
 #include <sound/soc.h>
 
+#define TDM_CHANNELS	8
+
 enum be_id {
 	HEADSET_BE_ID = 0,
 	AMP_BE_ID,
@@ -58,9 +60,11 @@ struct acp_card_drvdata {
 	struct clk *wclk;
 	struct clk *bclk;
 	bool soc_mclk;
+	bool tdm_mode;
 };
 
 int acp_sofdsp_dai_links_create(struct snd_soc_card *card);
 int acp_legacy_dai_links_create(struct snd_soc_card *card);
+extern const struct dmi_system_id acp_quirk_table[];
 
 #endif

sound/soc/amd/acp/acp-sof-mach.c

@@ -16,6 +16,7 @@
 #include <sound/pcm_params.h>
 #include <sound/soc-acpi.h>
 #include <sound/soc-dapm.h>
+#include <linux/dmi.h>
 #include <linux/module.h>
 
 #include "acp-mach.h"
@@ -27,6 +28,7 @@ static struct acp_card_drvdata sof_rt5682_rt1019_data = {
 	.hs_codec_id = RT5682,
 	.amp_codec_id = RT1019,
 	.dmic_codec_id = DMIC,
+	.tdm_mode = false,
 };
 
 static struct acp_card_drvdata sof_rt5682_max_data = {
@@ -36,6 +38,7 @@ static struct acp_card_drvdata sof_rt5682_max_data = {
 	.hs_codec_id = RT5682,
 	.amp_codec_id = MAX98360A,
 	.dmic_codec_id = DMIC,
+	.tdm_mode = false,
 };
 
 static struct acp_card_drvdata sof_rt5682s_rt1019_data = {
@@ -45,6 +48,7 @@ static struct acp_card_drvdata sof_rt5682s_rt1019_data = {
 	.hs_codec_id = RT5682S,
 	.amp_codec_id = RT1019,
 	.dmic_codec_id = DMIC,
+	.tdm_mode = false,
 };
 
 static struct acp_card_drvdata sof_rt5682s_max_data = {
@@ -54,6 +58,7 @@ static struct acp_card_drvdata sof_rt5682s_max_data = {
 	.hs_codec_id = RT5682S,
 	.amp_codec_id = MAX98360A,
 	.dmic_codec_id = DMIC,
+	.tdm_mode = false,
 };
 
 static struct acp_card_drvdata sof_nau8825_data = {
@@ -64,6 +69,7 @@ static struct acp_card_drvdata sof_nau8825_data = {
 	.amp_codec_id = MAX98360A,
 	.dmic_codec_id = DMIC,
 	.soc_mclk = true,
+	.tdm_mode = false,
 };
 
 static struct acp_card_drvdata sof_rt5682s_hs_rt1019_data = {
@@ -74,6 +80,7 @@ static struct acp_card_drvdata sof_rt5682s_hs_rt1019_data = {
 	.amp_codec_id = RT1019,
 	.dmic_codec_id = DMIC,
 	.soc_mclk = true,
+	.tdm_mode = false,
 };
 
 static const struct snd_kcontrol_new acp_controls[] = {
@@ -96,6 +103,8 @@ static int acp_sof_probe(struct platform_device *pdev)
 {
 	struct snd_soc_card *card = NULL;
 	struct device *dev = &pdev->dev;
+	const struct dmi_system_id *dmi_id;
+	struct acp_card_drvdata *acp_card_drvdata;
 	int ret;
 
 	if (!pdev->id_entry)
@@ -114,6 +123,11 @@ static int acp_sof_probe(struct platform_device *pdev)
 	card->num_controls = ARRAY_SIZE(acp_controls);
 	card->drvdata = (struct acp_card_drvdata *)pdev->id_entry->driver_data;
 
+	acp_card_drvdata = card->drvdata;
+	dmi_id = dmi_first_match(acp_quirk_table);
+	if (dmi_id && dmi_id->driver_data)
+		acp_card_drvdata->tdm_mode = dmi_id->driver_data;
+
 	acp_sofdsp_dai_links_create(card);
 
 	ret = devm_snd_soc_register_card(&pdev->dev, card);

sound/soc/amd/ps/acp63.h

@@ -11,7 +11,6 @@
 #define ACP63_REG_START		0x1240000
 #define ACP63_REG_END		0x1250200
 #define ACP63_DEVS		3
-#define ACP63_PDM_MODE		1
 
 #define ACP_SOFT_RESET_SOFTRESET_AUDDONE_MASK	0x00010001
 #define ACP_PGFSM_CNTL_POWER_ON_MASK	1
@@ -30,7 +29,7 @@
 #define ACP_ERROR_STAT	29
 #define PDM_DECIMATION_FACTOR	2
 #define ACP_PDM_CLK_FREQ_MASK	7
-#define ACP_WOV_MISC_CTRL_MASK	0x10
+#define ACP_WOV_GAIN_CONTROL	GENMASK(4, 3)
 #define ACP_PDM_ENABLE		1
 #define ACP_PDM_DISABLE		0
 #define ACP_PDM_DMA_EN_STATUS	2
@@ -54,6 +53,11 @@
 /* time in ms for runtime suspend delay */
 #define ACP_SUSPEND_DELAY_MS	2000
 
+#define ACP63_DMIC_ADDR		2
+#define ACP63_PDM_MODE_DEVS		3
+#define ACP63_PDM_DEV_MASK		1
+#define ACP_DMIC_DEV	2
+
 enum acp_config {
 	ACP_CONFIG_0 = 0,
 	ACP_CONFIG_1,
@@ -84,6 +88,7 @@ struct pdm_stream_instance {
 struct pdm_dev_data {
 	u32 pdm_irq;
 	void __iomem *acp63_base;
+	struct mutex *acp_lock;
 	struct snd_pcm_substream *capture_stream;
 };
 
@@ -100,6 +105,9 @@ static inline void acp63_writel(u32 val, void __iomem *base_addr)
 struct acp63_dev_data {
 	void __iomem *acp63_base;
 	struct resource *res;
-	bool acp63_audio_mode;
 	struct platform_device *pdev[ACP63_DEVS];
+	struct mutex acp_lock; /* protect shared registers */
+	u16 pdev_mask;
+	u16 pdev_count;
+	u16 pdm_dev_index;
 };

sound/soc/amd/ps/pci-ps.c

@@ -116,6 +116,7 @@ static irqreturn_t acp63_irq_handler(int irq, void *dev_id)
 	struct acp63_dev_data *adata;
 	struct pdm_dev_data *ps_pdm_data;
 	u32 val;
+	u16 pdev_index;
 
 	adata = dev_id;
 	if (!adata)
@@ -123,7 +124,8 @@ static irqreturn_t acp63_irq_handler(int irq, void *dev_id)
 
 	val = acp63_readl(adata->acp63_base + ACP_EXTERNAL_INTR_STAT);
 	if (val & BIT(PDM_DMA_STAT)) {
-		ps_pdm_data = dev_get_drvdata(&adata->pdev[0]->dev);
+		pdev_index = adata->pdm_dev_index;
+		ps_pdm_data = dev_get_drvdata(&adata->pdev[pdev_index]->dev);
 		acp63_writel(BIT(PDM_DMA_STAT), adata->acp63_base + ACP_EXTERNAL_INTR_STAT);
 		if (ps_pdm_data->capture_stream)
 			snd_pcm_period_elapsed(ps_pdm_data->capture_stream);
@@ -132,17 +134,124 @@ static irqreturn_t acp63_irq_handler(int irq, void *dev_id)
 	return IRQ_NONE;
 }
 
+static void get_acp63_device_config(u32 config, struct pci_dev *pci,
+				    struct acp63_dev_data *acp_data)
+{
+	struct acpi_device *dmic_dev;
+	const union acpi_object *obj;
+	bool is_dmic_dev = false;
+
+	dmic_dev = acpi_find_child_device(ACPI_COMPANION(&pci->dev), ACP63_DMIC_ADDR, 0);
+	if (dmic_dev) {
+		if (!acpi_dev_get_property(dmic_dev, "acp-audio-device-type",
+					   ACPI_TYPE_INTEGER, &obj) &&
+					   obj->integer.value == ACP_DMIC_DEV)
+			is_dmic_dev = true;
+	}
+
+	switch (config) {
+	case ACP_CONFIG_0:
+	case ACP_CONFIG_1:
+	case ACP_CONFIG_2:
+	case ACP_CONFIG_3:
+	case ACP_CONFIG_9:
+	case ACP_CONFIG_15:
+		dev_dbg(&pci->dev, "Audio Mode %d\n", config);
+		break;
+	default:
+		if (is_dmic_dev) {
+			acp_data->pdev_mask = ACP63_PDM_DEV_MASK;
+			acp_data->pdev_count = ACP63_PDM_MODE_DEVS;
+		}
+		break;
+	}
+}
+
+static void acp63_fill_platform_dev_info(struct platform_device_info *pdevinfo,
+					 struct device *parent,
+					 struct fwnode_handle *fw_node,
+					 char *name, unsigned int id,
+					 const struct resource *res,
+					 unsigned int num_res,
+					 const void *data,
+					 size_t size_data)
+{
+	pdevinfo->name = name;
+	pdevinfo->id = id;
+	pdevinfo->parent = parent;
+	pdevinfo->num_res = num_res;
+	pdevinfo->res = res;
+	pdevinfo->data = data;
+	pdevinfo->size_data = size_data;
+	pdevinfo->fwnode = fw_node;
+}
+
+static int create_acp63_platform_devs(struct pci_dev *pci, struct acp63_dev_data *adata, u32 addr)
+{
+	struct platform_device_info pdevinfo[ACP63_DEVS];
+	struct device *parent;
+	int index;
+	int ret;
+
+	parent = &pci->dev;
+	dev_dbg(&pci->dev,
+		"%s pdev_mask:0x%x pdev_count:0x%x\n", __func__, adata->pdev_mask,
+		adata->pdev_count);
+	if (adata->pdev_mask) {
+		adata->res = devm_kzalloc(&pci->dev, sizeof(struct resource), GFP_KERNEL);
+		if (!adata->res) {
+			ret = -ENOMEM;
+			goto de_init;
+		}
+		adata->res->flags = IORESOURCE_MEM;
+		adata->res->start = addr;
+		adata->res->end = addr + (ACP63_REG_END - ACP63_REG_START);
+		memset(&pdevinfo, 0, sizeof(pdevinfo));
+	}
+
+	switch (adata->pdev_mask) {
+	case ACP63_PDM_DEV_MASK:
+		adata->pdm_dev_index  = 0;
+		acp63_fill_platform_dev_info(&pdevinfo[0], parent, NULL, "acp_ps_pdm_dma",
+					     0, adata->res, 1, &adata->acp_lock,
+					     sizeof(adata->acp_lock));
+		acp63_fill_platform_dev_info(&pdevinfo[1], parent, NULL, "dmic-codec",
+					     0, NULL, 0, NULL, 0);
+		acp63_fill_platform_dev_info(&pdevinfo[2], parent, NULL, "acp_ps_mach",
+					     0, NULL, 0, NULL, 0);
+		break;
+	default:
+		dev_dbg(&pci->dev, "No PDM devices found\n");
+		return 0;
+	}
+
+	for (index = 0; index < adata->pdev_count; index++) {
+		adata->pdev[index] = platform_device_register_full(&pdevinfo[index]);
+		if (IS_ERR(adata->pdev[index])) {
+			dev_err(&pci->dev,
+				"cannot register %s device\n", pdevinfo[index].name);
+			ret = PTR_ERR(adata->pdev[index]);
+			goto unregister_devs;
+		}
+	}
+	return 0;
+unregister_devs:
+	for (--index; index >= 0; index--)
+		platform_device_unregister(adata->pdev[index]);
+de_init:
+	if (acp63_deinit(adata->acp63_base, &pci->dev))
+		dev_err(&pci->dev, "ACP de-init failed\n");
+	return ret;
+}
+
 static int snd_acp63_probe(struct pci_dev *pci,
 			   const struct pci_device_id *pci_id)
 {
 	struct acp63_dev_data *adata;
-	struct platform_device_info pdevinfo[ACP63_DEVS];
-	int index, ret;
-	int val = 0x00;
-	struct acpi_device *adev;
-	const union acpi_object *obj;
 	u32 addr;
-	unsigned int irqflags;
+	u32 irqflags;
+	int val;
+	int ret;
 
 	irqflags = IRQF_SHARED;
 	/* Pink Sardine device check */
@@ -179,86 +288,28 @@ static int snd_acp63_probe(struct pci_dev *pci,
 	}
 	pci_set_master(pci);
 	pci_set_drvdata(pci, adata);
+	mutex_init(&adata->acp_lock);
 	ret = acp63_init(adata->acp63_base, &pci->dev);
 	if (ret)
 		goto release_regions;
+	ret = devm_request_irq(&pci->dev, pci->irq, acp63_irq_handler,
+			       irqflags, "ACP_PCI_IRQ", adata);
+	if (ret) {
+		dev_err(&pci->dev, "ACP PCI IRQ request failed\n");
+		goto de_init;
+	}
 	val = acp63_readl(adata->acp63_base + ACP_PIN_CONFIG);
-	switch (val) {
-	case ACP_CONFIG_0:
-	case ACP_CONFIG_1:
-	case ACP_CONFIG_2:
-	case ACP_CONFIG_3:
-	case ACP_CONFIG_9:
-	case ACP_CONFIG_15:
-		dev_info(&pci->dev, "Audio Mode %d\n", val);
-		break;
-	default:
-
-		/* Checking DMIC hardware*/
-		adev = acpi_find_child_device(ACPI_COMPANION(&pci->dev), 0x02, 0);
-
-		if (!adev)
-			break;
-
-		if (!acpi_dev_get_property(adev, "acp-audio-device-type",
-					   ACPI_TYPE_INTEGER, &obj) &&
-					   obj->integer.value == 2) {
-			adata->res = devm_kzalloc(&pci->dev, sizeof(struct resource), GFP_KERNEL);
-			if (!adata->res) {
-				ret = -ENOMEM;
-				goto de_init;
-			}
-
-			adata->res->name = "acp_iomem";
-			adata->res->flags = IORESOURCE_MEM;
-			adata->res->start = addr;
-			adata->res->end = addr + (ACP63_REG_END - ACP63_REG_START);
-			adata->acp63_audio_mode = ACP63_PDM_MODE;
-
-			memset(&pdevinfo, 0, sizeof(pdevinfo));
-			pdevinfo[0].name = "acp_ps_pdm_dma";
-			pdevinfo[0].id = 0;
-			pdevinfo[0].parent = &pci->dev;
-			pdevinfo[0].num_res = 1;
-			pdevinfo[0].res = adata->res;
-
-			pdevinfo[1].name = "dmic-codec";
-			pdevinfo[1].id = 0;
-			pdevinfo[1].parent = &pci->dev;
-
-			pdevinfo[2].name = "acp_ps_mach";
-			pdevinfo[2].id = 0;
-			pdevinfo[2].parent = &pci->dev;
-
-			for (index = 0; index < ACP63_DEVS; index++) {
-				adata->pdev[index] =
-					platform_device_register_full(&pdevinfo[index]);
-
-				if (IS_ERR(adata->pdev[index])) {
-					dev_err(&pci->dev,
-						"cannot register %s device\n",
-						 pdevinfo[index].name);
-					ret = PTR_ERR(adata->pdev[index]);
-					goto unregister_devs;
-				}
-				ret = devm_request_irq(&pci->dev, pci->irq, acp63_irq_handler,
-						       irqflags, "ACP_PCI_IRQ", adata);
-				if (ret) {
-					dev_err(&pci->dev, "ACP PCI IRQ request failed\n");
-					goto unregister_devs;
-				}
-			}
-		}
-		break;
+	get_acp63_device_config(val, pci, adata);
+	ret = create_acp63_platform_devs(pci, adata, addr);
+	if (ret < 0) {
+		dev_err(&pci->dev, "ACP platform devices creation failed\n");
+		goto de_init;
 	}
 	pm_runtime_set_autosuspend_delay(&pci->dev, ACP_SUSPEND_DELAY_MS);
 	pm_runtime_use_autosuspend(&pci->dev);
 	pm_runtime_put_noidle(&pci->dev);
 	pm_runtime_allow(&pci->dev);
 	return 0;
-unregister_devs:
-	for (--index; index >= 0; index--)
-		platform_device_unregister(adata->pdev[index]);
 de_init:
 	if (acp63_deinit(adata->acp63_base, &pci->dev))
 		dev_err(&pci->dev, "ACP de-init failed\n");
@@ -305,10 +356,8 @@ static void snd_acp63_remove(struct pci_dev *pci)
 	int ret, index;
 
 	adata = pci_get_drvdata(pci);
-	if (adata->acp63_audio_mode == ACP63_PDM_MODE) {
-		for (index = 0; index < ACP63_DEVS; index++)
-			platform_device_unregister(adata->pdev[index]);
-	}
+	for (index = 0; index < adata->pdev_count; index++)
+		platform_device_unregister(adata->pdev[index]);
 	ret = acp63_deinit(adata->acp63_base, &pci->dev);
 	if (ret)
 		dev_err(&pci->dev, "ACP de-init failed\n");

sound/soc/amd/ps/ps-pdm-dma.c

@@ -7,6 +7,7 @@
 
 #include <linux/platform_device.h>
 #include <linux/module.h>
+#include <linux/bitfield.h>
 #include <linux/err.h>
 #include <linux/io.h>
 #include <sound/pcm_params.h>
@@ -18,6 +19,10 @@
 
 #define DRV_NAME "acp_ps_pdm_dma"
 
+static int pdm_gain = 3;
+module_param(pdm_gain, int, 0644);
+MODULE_PARM_DESC(pdm_gain, "Gain control (0-3)");
+
 static const struct snd_pcm_hardware acp63_pdm_hardware_capture = {
 	.info = SNDRV_PCM_INFO_INTERLEAVED |
 		SNDRV_PCM_INFO_BLOCK_TRANSFER |
@@ -55,26 +60,31 @@ static void acp63_enable_pdm_clock(void __iomem *acp_base)
 
 	acp63_writel(pdm_clk_enable, acp_base + ACP_WOV_CLK_CTRL);
 	pdm_ctrl = acp63_readl(acp_base + ACP_WOV_MISC_CTRL);
-	pdm_ctrl |= ACP_WOV_MISC_CTRL_MASK;
+	pdm_ctrl &= ~ACP_WOV_GAIN_CONTROL;
+	pdm_ctrl |= FIELD_PREP(ACP_WOV_GAIN_CONTROL, clamp(pdm_gain, 0, 3));
 	acp63_writel(pdm_ctrl, acp_base + ACP_WOV_MISC_CTRL);
 }
 
-static void acp63_enable_pdm_interrupts(void __iomem *acp_base)
+static void acp63_enable_pdm_interrupts(struct pdm_dev_data *adata)
 {
 	u32 ext_int_ctrl;
 
-	ext_int_ctrl = acp63_readl(acp_base + ACP_EXTERNAL_INTR_CNTL);
+	mutex_lock(adata->acp_lock);
+	ext_int_ctrl = acp63_readl(adata->acp63_base + ACP_EXTERNAL_INTR_CNTL);
 	ext_int_ctrl |= PDM_DMA_INTR_MASK;
-	acp63_writel(ext_int_ctrl, acp_base + ACP_EXTERNAL_INTR_CNTL);
+	acp63_writel(ext_int_ctrl, adata->acp63_base + ACP_EXTERNAL_INTR_CNTL);
+	mutex_unlock(adata->acp_lock);
 }
 
-static void acp63_disable_pdm_interrupts(void __iomem *acp_base)
+static void acp63_disable_pdm_interrupts(struct pdm_dev_data *adata)
 {
 	u32 ext_int_ctrl;
 
-	ext_int_ctrl = acp63_readl(acp_base + ACP_EXTERNAL_INTR_CNTL);
+	mutex_lock(adata->acp_lock);
+	ext_int_ctrl = acp63_readl(adata->acp63_base + ACP_EXTERNAL_INTR_CNTL);
 	ext_int_ctrl &= ~PDM_DMA_INTR_MASK;
-	acp63_writel(ext_int_ctrl, acp_base + ACP_EXTERNAL_INTR_CNTL);
+	acp63_writel(ext_int_ctrl, adata->acp63_base + ACP_EXTERNAL_INTR_CNTL);
+	mutex_unlock(adata->acp_lock);
 }
 
 static bool acp63_check_pdm_dma_status(void __iomem *acp_base)
@@ -196,7 +206,7 @@ static int acp63_pdm_dma_open(struct snd_soc_component *component,
 		return ret;
 	}
 
-	acp63_enable_pdm_interrupts(adata->acp63_base);
+	acp63_enable_pdm_interrupts(adata);
 
 	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
 		adata->capture_stream = substream;
@@ -272,7 +282,7 @@ static int acp63_pdm_dma_close(struct snd_soc_component *component,
 	struct pdm_dev_data *adata = dev_get_drvdata(component->dev);
 	struct snd_pcm_runtime *runtime = substream->runtime;
 
-	acp63_disable_pdm_interrupts(adata->acp63_base);
+	acp63_disable_pdm_interrupts(adata);
 	adata->capture_stream = NULL;
 	kfree(runtime->private_data);
 	return 0;
@@ -353,6 +363,10 @@ static int acp63_pdm_audio_probe(struct platform_device *pdev)
 	struct pdm_dev_data *adata;
 	int status;
 
+	if (!pdev->dev.platform_data) {
+		dev_err(&pdev->dev, "platform_data not retrieved\n");
+		return -ENODEV;
+	}
 	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
 	if (!res) {
 		dev_err(&pdev->dev, "IORESOURCE_MEM FAILED\n");
@@ -368,7 +382,7 @@ static int acp63_pdm_audio_probe(struct platform_device *pdev)
 		return -ENOMEM;
 
 	adata->capture_stream = NULL;
-
+	adata->acp_lock = pdev->dev.platform_data;
 	dev_set_drvdata(&pdev->dev, adata);
 	status = devm_snd_soc_register_component(&pdev->dev,
 						 &acp63_pdm_component,
@@ -408,7 +422,7 @@ static int __maybe_unused acp63_pdm_resume(struct device *dev)
 		acp63_init_pdm_ring_buffer(PDM_MEM_WINDOW_START, buffer_len,
 					   period_bytes, adata->acp63_base);
 	}
-	acp63_enable_pdm_interrupts(adata->acp63_base);
+	acp63_enable_pdm_interrupts(adata);
 	return 0;
 }
 
@@ -417,7 +431,7 @@ static int __maybe_unused acp63_pdm_suspend(struct device *dev)
 	struct pdm_dev_data *adata;
 
 	adata = dev_get_drvdata(dev);
-	acp63_disable_pdm_interrupts(adata->acp63_base);
+	acp63_disable_pdm_interrupts(adata);
 	return 0;
 }
 
@@ -426,7 +440,7 @@ static int __maybe_unused acp63_pdm_runtime_resume(struct device *dev)
 	struct pdm_dev_data *adata;
 
 	adata = dev_get_drvdata(dev);
-	acp63_enable_pdm_interrupts(adata->acp63_base);
+	acp63_enable_pdm_interrupts(adata);
 	return 0;
 }
 

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

@@ -315,16 +315,8 @@ static int acp3x_dai_probe(struct platform_device *pdev)
 	return 0;
 }
 
-static int acp3x_dai_remove(struct platform_device *pdev)
-{
-	/* As we use devm_ memory alloc there is nothing TBD here */
-
-	return 0;
-}
-
 static struct platform_driver acp3x_dai_driver = {
 	.probe = acp3x_dai_probe,
-	.remove = acp3x_dai_remove,
 	.driver = {
 		.name = "acp3x_i2s_playcap",
 	},

sound/soc/amd/renoir/acp3x-pdm-dma.c

@@ -6,6 +6,7 @@
 
 #include <linux/platform_device.h>
 #include <linux/module.h>
+#include <linux/bitfield.h>
 #include <linux/err.h>
 #include <linux/io.h>
 #include <linux/pm_runtime.h>
@@ -17,6 +18,10 @@
 
 #define DRV_NAME "acp_rn_pdm_dma"
 
+static int pdm_gain = 3;
+module_param(pdm_gain, int, 0644);
+MODULE_PARM_DESC(pdm_gain, "Gain control (0-3)");
+
 static const struct snd_pcm_hardware acp_pdm_hardware_capture = {
 	.info = SNDRV_PCM_INFO_INTERLEAVED |
 		SNDRV_PCM_INFO_BLOCK_TRANSFER |
@@ -80,7 +85,8 @@ static void enable_pdm_clock(void __iomem *acp_base)
 
 	rn_writel(pdm_clk_enable, acp_base + ACP_WOV_CLK_CTRL);
 	pdm_ctrl = rn_readl(acp_base + ACP_WOV_MISC_CTRL);
-	pdm_ctrl |= ACP_WOV_MISC_CTRL_MASK;
+	pdm_ctrl &= ~ACP_WOV_GAIN_CONTROL;
+	pdm_ctrl |= FIELD_PREP(ACP_WOV_GAIN_CONTROL, clamp(pdm_gain, 0, 3));
 	rn_writel(pdm_ctrl, acp_base + ACP_WOV_MISC_CTRL);
 }
 

sound/soc/amd/renoir/rn_acp3x.h

@@ -34,7 +34,7 @@
 #define ACP_ERROR_STAT 29
 #define PDM_DECIMATION_FACTOR 0x2
 #define ACP_PDM_CLK_FREQ_MASK 0x07
-#define ACP_WOV_MISC_CTRL_MASK 0x10
+#define ACP_WOV_GAIN_CONTROL	GENMASK(4, 3)
 #define ACP_PDM_ENABLE 0x01
 #define ACP_PDM_DISABLE 0x00
 #define ACP_PDM_DMA_EN_STATUS 0x02

sound/soc/amd/yc/acp6x.h

@@ -31,7 +31,7 @@
 #define ACP_ERROR_STAT	29
 #define PDM_DECIMATION_FACTOR	2
 #define ACP_PDM_CLK_FREQ_MASK	7
-#define ACP_WOV_MISC_CTRL_MASK	0x10
+#define ACP_WOV_GAIN_CONTROL	GENMASK(4, 3)
 #define ACP_PDM_ENABLE		1
 #define ACP_PDM_DISABLE		0
 #define ACP_PDM_DMA_EN_STATUS	2