diff --git a/Bindings/sound/davinci-mcbsp.txt b/Bindings/sound/davinci-mcbsp.txt
deleted file mode 100644
index 3ffc256..0000000
--- a/Bindings/sound/davinci-mcbsp.txt
+++ /dev/null
@@ -1,50 +0,0 @@
-Texas Instruments DaVinci McBSP module
-~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
-
-This binding describes the "Multi-channel Buffered Serial Port" (McBSP)
-audio interface found in some TI DaVinci processors like the OMAP-L138 or AM180x.
-
-
-Required properties:
-~~~~~~~~~~~~~~~~~~~~
-- compatible :
-        "ti,da850-mcbsp" : for DA850, AM180x and OPAM-L138 platforms
-
-- reg : physical base address and length of the controller memory mapped
-        region(s).
-- reg-names : Should contain:
-        * "mpu" for the main registers (required).
-        * "dat" for the data FIFO (optional).
-
-- dmas: three element list of DMA controller phandles, DMA request line and
-	TC channel ordered triplets.
-- dma-names: identifier string for each DMA request line in the dmas property.
-	These strings correspond 1:1 with the ordered pairs in dmas. The dma
-	identifiers must be "rx" and "tx".
-
-Optional properties:
-~~~~~~~~~~~~~~~~~~~~
-- interrupts : Interrupt numbers for McBSP
-- interrupt-names : Known interrupt names are "rx" and "tx"
-
-- pinctrl-0: Should specify pin control group used for this controller.
-- pinctrl-names: Should contain only one value - "default", for more details
-        please refer to pinctrl-bindings.txt
-
-Example (AM1808):
-~~~~~~~~~~~~~~~~~
-
-mcbsp0: mcbsp@1d10000 {
-	compatible = "ti,da850-mcbsp";
-	pinctrl-names = "default";
-	pinctrl-0 = <&mcbsp0_pins>;
-
-	reg = 	<0x00110000 0x1000>,
-		<0x00310000 0x1000>;
-	reg-names = "mpu", "dat";
-	interrupts = <97 98>;
-	interrupt-names = "rx", "tx";
-	dmas = <&edma0 3 1
-		&edma0 2 1>;
-	dma-names = "tx", "rx";
-};
diff --git a/Bindings/sound/davinci-mcbsp.yaml b/Bindings/sound/davinci-mcbsp.yaml
new file mode 100644
index 0000000..4fa6770
--- /dev/null
+++ b/Bindings/sound/davinci-mcbsp.yaml
@@ -0,0 +1,113 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/davinci-mcbsp.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: McBSP Controller for TI SoCs
+
+maintainers:
+  - Bastien Curutchet <bastien.curutchet@bootlin.com>
+
+allOf:
+  - $ref: dai-common.yaml#
+
+properties:
+  compatible:
+    enum:
+      - ti,da850-mcbsp
+
+  reg:
+    minItems: 1
+    items:
+      - description: CFG registers
+      - description: data registers
+
+  reg-names:
+    minItems: 1
+    items:
+      - const: mpu
+      - const: dat
+
+  dmas:
+    items:
+      - description: transmission DMA channel
+      - description: reception DMA channel
+
+  dma-names:
+    items:
+      - const: tx
+      - const: rx
+
+  interrupts:
+    items:
+      - description: RX interrupt
+      - description: TX interrupt
+
+  interrupt-names:
+    items:
+      - const: rx
+      - const: tx
+
+  clocks:
+    minItems: 1
+    items:
+      - description: functional clock
+      - description: external input clock for sample rate generator.
+
+  clock-names:
+    minItems: 1
+    items:
+      - const: fck
+      - const: clks
+
+  power-domains:
+    maxItems: 1
+
+  "#sound-dai-cells":
+    const: 0
+
+  ti,T1-framing-tx:
+    $ref: /schemas/types.yaml#/definitions/flag
+    description:
+      If the property is present, tx data delay is set to 2 bit clock periods.
+      McBSP will insert a blank period (high-impedance period) before the first
+      data bit. This can be used to interface to T1-framing devices.
+
+  ti,T1-framing-rx:
+    $ref: /schemas/types.yaml#/definitions/flag
+    description:
+      If the property is present, rx data delay is set to 2 bit clock periods.
+      McBSP will discard the bit preceding the data stream (called framing bit).
+      This can be used to interface to T1-framing devices.
+
+required:
+  - "#sound-dai-cells"
+  - compatible
+  - reg
+  - reg-names
+  - dmas
+  - dma-names
+  - clocks
+
+unevaluatedProperties: false
+
+examples:
+  - |
+    mcbsp0@1d10000 {
+      #sound-dai-cells = <0>;
+      compatible = "ti,da850-mcbsp";
+      pinctrl-names = "default";
+      pinctrl-0 = <&mcbsp0_pins>;
+
+      reg = <0x111000 0x1000>,
+            <0x311000 0x1000>;
+      reg-names = "mpu", "dat";
+      interrupts = <97>, <98>;
+      interrupt-names = "rx", "tx";
+      dmas = <&edma0 3 1>,
+             <&edma0 2 1>;
+      dma-names = "tx", "rx";
+
+      clocks = <&psc1 14>;
+    };
diff --git a/Bindings/sound/fsl,audmix.txt b/Bindings/sound/fsl,audmix.txt
deleted file mode 100644
index 840b7e0..0000000
--- a/Bindings/sound/fsl,audmix.txt
+++ /dev/null
@@ -1,50 +0,0 @@
-NXP Audio Mixer (AUDMIX).
-
-The Audio Mixer is a on-chip functional module that allows mixing of two
-audio streams into a single audio stream. Audio Mixer has two input serial
-audio interfaces. These are driven by two Synchronous Audio interface
-modules (SAI). Each input serial interface carries 8 audio channels in its
-frame in TDM manner. Mixer mixes audio samples of corresponding channels
-from two interfaces into a single sample. Before mixing, audio samples of
-two inputs can be attenuated based on configuration. The output of the
-Audio Mixer is also a serial audio interface. Like input interfaces it has
-the same TDM frame format. This output is used to drive the serial DAC TDM
-interface of audio codec and also sent to the external pins along with the
-receive path of normal audio SAI module for readback by the CPU.
-
-The output of Audio Mixer can be selected from any of the three streams
- - serial audio input 1
- - serial audio input 2
- - mixed audio
-
-Mixing operation is independent of audio sample rate but the two audio
-input streams must have same audio sample rate with same number of channels
-in TDM frame to be eligible for mixing.
-
-Device driver required properties:
-=================================
-  - compatible		: Compatible list, contains "fsl,imx8qm-audmix"
-
-  - reg			: Offset and length of the register set for the device.
-
-  - clocks		: Must contain an entry for each entry in clock-names.
-
-  - clock-names		: Must include the "ipg" for register access.
-
-  - power-domains	: Must contain the phandle to AUDMIX power domain node
-
-  - dais		: Must contain a list of phandles to AUDMIX connected
-			  DAIs. The current implementation requires two phandles
-			  to SAI interfaces to be provided, the first SAI in the
-			  list being used to route the AUDMIX output.
-
-Device driver configuration example:
-======================================
-  audmix: audmix@59840000 {
-    compatible = "fsl,imx8qm-audmix";
-    reg = <0x0 0x59840000 0x0 0x10000>;
-    clocks = <&clk IMX8QXP_AUD_AUDMIX_IPG>;
-    clock-names = "ipg";
-    power-domains = <&pd_audmix>;
-    dais = <&sai4>, <&sai5>;
-  };
diff --git a/Bindings/sound/fsl,audmix.yaml b/Bindings/sound/fsl,audmix.yaml
new file mode 100644
index 0000000..9413b90
--- /dev/null
+++ b/Bindings/sound/fsl,audmix.yaml
@@ -0,0 +1,83 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/fsl,audmix.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: NXP Audio Mixer (AUDMIX).
+
+maintainers:
+  - Shengjiu Wang <shengjiu.wang@nxp.com>
+  - Frank Li <Frank.Li@nxp.com>
+
+description: |
+  The Audio Mixer is a on-chip functional module that allows mixing of two
+  audio streams into a single audio stream. Audio Mixer has two input serial
+  audio interfaces. These are driven by two Synchronous Audio interface
+  modules (SAI). Each input serial interface carries 8 audio channels in its
+  frame in TDM manner. Mixer mixes audio samples of corresponding channels
+  from two interfaces into a single sample. Before mixing, audio samples of
+  two inputs can be attenuated based on configuration. The output of the
+  Audio Mixer is also a serial audio interface. Like input interfaces it has
+  the same TDM frame format. This output is used to drive the serial DAC TDM
+  interface of audio codec and also sent to the external pins along with the
+  receive path of normal audio SAI module for readback by the CPU.
+
+  The output of Audio Mixer can be selected from any of the three streams
+    - serial audio input 1
+    - serial audio input 2
+    - mixed audio
+
+  Mixing operation is independent of audio sample rate but the two audio
+  input streams must have same audio sample rate with same number of channels
+  in TDM frame to be eligible for mixing.
+
+properties:
+  compatible:
+    const: fsl,imx8qm-audmix
+
+  reg:
+    maxItems: 1
+
+  clocks:
+    maxItems: 1
+
+  clock-names:
+    items:
+      - const: ipg
+
+  power-domains:
+    maxItems: 1
+
+  dais:
+    description: contain a list of phandles to AUDMIX connected DAIs.
+    $ref: /schemas/types.yaml#/definitions/phandle-array
+    minItems: 2
+    items:
+      - description: the AUDMIX output
+        maxItems: 1
+      - description: serial audio input 1
+        maxItems: 1
+      - description: serial audio input 2
+        maxItems: 1
+
+required:
+  - compatible
+  - reg
+  - clocks
+  - clock-names
+  - power-domains
+  - dais
+
+unevaluatedProperties: false
+
+examples:
+  - |
+    audmix@59840000 {
+      compatible = "fsl,imx8qm-audmix";
+      reg = <0x59840000 0x10000>;
+      clocks = <&amix_lpcg 0>;
+      clock-names = "ipg";
+      power-domains = <&pd_audmix>;
+      dais = <&sai4>, <&sai5>;
+    };
diff --git a/Bindings/sound/fsl,esai.txt b/Bindings/sound/fsl,esai.txt
deleted file mode 100644
index 90112ca..0000000
--- a/Bindings/sound/fsl,esai.txt
+++ /dev/null
@@ -1,68 +0,0 @@
-Freescale Enhanced Serial Audio Interface (ESAI) Controller
-
-The Enhanced Serial Audio Interface (ESAI) provides a full-duplex serial port
-for serial communication with a variety of serial devices, including industry
-standard codecs, Sony/Phillips Digital Interface (S/PDIF) transceivers, and
-other DSPs. It has up to six transmitters and four receivers.
-
-Required properties:
-
-  - compatible		: Compatible list, should contain one of the following
-			  compatibles:
-			  "fsl,imx35-esai",
-			  "fsl,vf610-esai",
-			  "fsl,imx6ull-esai",
-			  "fsl,imx8qm-esai",
-
-  - reg			: Offset and length of the register set for the device.
-
-  - interrupts		: Contains the spdif interrupt.
-
-  - dmas		: Generic dma devicetree binding as described in
-			  Documentation/devicetree/bindings/dma/dma.txt.
-
-  - dma-names		: Two dmas have to be defined, "tx" and "rx".
-
-  - clocks		: Contains an entry for each entry in clock-names.
-
-  - clock-names		: Includes the following entries:
-	"core"		  The core clock used to access registers
-	"extal"		  The esai baud clock for esai controller used to
-			  derive HCK, SCK and FS.
-	"fsys"		  The system clock derived from ahb clock used to
-			  derive HCK, SCK and FS.
-	"spba"		  The spba clock is required when ESAI is placed as a
-			  bus slave of the Shared Peripheral Bus and when two
-			  or more bus masters (CPU, DMA or DSP) try to access
-			  it. This property is optional depending on the SoC
-			  design.
-
-  - fsl,fifo-depth	: The number of elements in the transmit and receive
-			  FIFOs. This number is the maximum allowed value for
-			  TFCR[TFWM] or RFCR[RFWM].
-
-  - fsl,esai-synchronous: This is a boolean property. If present, indicating
-			  that ESAI would work in the synchronous mode, which
-			  means all the settings for Receiving would be
-			  duplicated from Transmission related registers.
-
-Optional properties:
-
-  - big-endian		: If this property is absent, the native endian mode
-			  will be in use as default, or the big endian mode
-			  will be in use for all the device registers.
-
-Example:
-
-esai: esai@2024000 {
-	compatible = "fsl,imx35-esai";
-	reg = <0x02024000 0x4000>;
-	interrupts = <0 51 0x04>;
-	clocks = <&clks 208>, <&clks 118>, <&clks 208>;
-	clock-names = "core", "extal", "fsys";
-	dmas = <&sdma 23 21 0>, <&sdma 24 21 0>;
-	dma-names = "rx", "tx";
-	fsl,fifo-depth = <128>;
-	fsl,esai-synchronous;
-	big-endian;
-};
diff --git a/Bindings/sound/fsl,esai.yaml b/Bindings/sound/fsl,esai.yaml
new file mode 100644
index 0000000..f99ed20
--- /dev/null
+++ b/Bindings/sound/fsl,esai.yaml
@@ -0,0 +1,118 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/fsl,esai.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Freescale Enhanced Serial Audio Interface (ESAI) Controller
+
+maintainers:
+  - Shengjiu Wang <shengjiu.wang@nxp.com>
+  - Frank Li <Frank.Li@nxp.com>
+
+description:
+  The Enhanced Serial Audio Interface (ESAI) provides a full-duplex serial port
+  for serial communication with a variety of serial devices, including industry
+  standard codecs, Sony/Phillips Digital Interface (S/PDIF) transceivers, and
+  other DSPs. It has up to six transmitters and four receivers.
+
+properties:
+  compatible:
+    enum:
+      - fsl,imx35-esai
+      - fsl,imx6ull-esai
+      - fsl,imx8qm-esai
+      - fsl,vf610-esai
+
+  reg:
+    maxItems: 1
+
+  interrupts:
+    maxItems: 1
+
+  clocks:
+    minItems: 3
+    items:
+      - description:
+          The core clock used to access registers.
+      - description:
+          The esai baud clock for esai controller used to
+          derive HCK, SCK and FS.
+      - description:
+          The system clock derived from ahb clock used to
+          derive HCK, SCK and FS.
+      - description:
+          The spba clock is required when ESAI is placed as a
+          bus slave of the Shared Peripheral Bus and when two
+          or more bus masters (CPU, DMA or DSP) try to access
+          it. This property is optional depending on the SoC
+          design.
+
+  clock-names:
+    minItems: 3
+    items:
+      - const: core
+      - const: extal
+      - const: fsys
+      - const: spba
+
+  dmas:
+    minItems: 2
+    maxItems: 2
+
+  dma-names:
+    items:
+      - const: rx
+      - const: tx
+
+  fsl,fifo-depth:
+    $ref: /schemas/types.yaml#/definitions/uint32
+    default: 64
+    description:
+      The number of elements in the transmit and receive
+      FIFOs. This number is the maximum allowed value for
+      TFCR[TFWM] or RFCR[RFWM].
+
+  fsl,esai-synchronous:
+    $ref: /schemas/types.yaml#/definitions/flag
+    description:
+      This is a boolean property. If present, indicating
+      that ESAI would work in the synchronous mode, which
+      means all the settings for Receiving would be
+      duplicated from Transmission related registers.
+
+  big-endian:
+    $ref: /schemas/types.yaml#/definitions/flag
+    description:
+      If this property is absent, the native endian mode
+      will be in use as default, or the big endian mode
+      will be in use for all the device registers.
+
+required:
+  - compatible
+  - reg
+  - interrupts
+  - clocks
+  - clock-names
+  - dmas
+  - dma-names
+
+unevaluatedProperties: false
+
+allOf:
+  - $ref: dai-common.yaml#
+
+examples:
+  - |
+    esai@2024000 {
+      compatible = "fsl,imx35-esai";
+      reg = <0x02024000 0x4000>;
+      interrupts = <0 51 0x04>;
+      clocks = <&clks 208>, <&clks 118>, <&clks 208>;
+      clock-names = "core", "extal", "fsys";
+      dmas = <&sdma 23 21 0>, <&sdma 24 21 0>;
+      dma-names = "rx", "tx";
+      fsl,fifo-depth = <128>;
+      fsl,esai-synchronous;
+      big-endian;
+    };
diff --git a/Bindings/sound/fsl,imx-asrc.yaml b/Bindings/sound/fsl,imx-asrc.yaml
index bfef2fc..76aa1f2 100644
--- a/Bindings/sound/fsl,imx-asrc.yaml
+++ b/Bindings/sound/fsl,imx-asrc.yaml
@@ -74,6 +74,9 @@
       - const: asrck_f
       - const: spba
 
+  power-domains:
+    maxItems: 1
+
   fsl,asrc-rate:
     $ref: /schemas/types.yaml#/definitions/uint32
     description: The mutual sample rate used by DPCM Back Ends
@@ -131,6 +134,17 @@
       properties:
         fsl,asrc-clk-map: false
 
+  - if:
+      properties:
+        compatible:
+          contains:
+            enum:
+              - fsl,imx8qm-asrc
+              - fsl,imx8qxp-asrc
+    then:
+      required:
+        - power-domains
+
 additionalProperties: false
 
 examples:
diff --git a/Bindings/sound/fsl,imx-audio-spdif.yaml b/Bindings/sound/fsl,imx-audio-spdif.yaml
new file mode 100644
index 0000000..5fc543d
--- /dev/null
+++ b/Bindings/sound/fsl,imx-audio-spdif.yaml
@@ -0,0 +1,66 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/fsl,imx-audio-spdif.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Freescale i.MX audio complex with S/PDIF transceiver
+
+maintainers:
+  - Shengjiu Wang <shengjiu.wang@nxp.com>
+
+properties:
+  compatible:
+    oneOf:
+      - items:
+          - enum:
+              - fsl,imx-sabreauto-spdif
+              - fsl,imx6sx-sdb-spdif
+          - const: fsl,imx-audio-spdif
+      - enum:
+          - fsl,imx-audio-spdif
+
+  model:
+    $ref: /schemas/types.yaml#/definitions/string
+    description: User specified audio sound card name
+
+  spdif-controller:
+    $ref: /schemas/types.yaml#/definitions/phandle
+    description: The phandle of the i.MX S/PDIF controller
+
+  spdif-out:
+    type: boolean
+    description:
+      If present, the transmitting function of S/PDIF will be enabled,
+      indicating there's a physical S/PDIF out connector or jack on the
+      board or it's connecting to some other IP block, such as an HDMI
+      encoder or display-controller.
+
+  spdif-in:
+    type: boolean
+    description:
+      If present, the receiving function of S/PDIF will be enabled,
+      indicating there is a physical S/PDIF in connector/jack on the board.
+
+required:
+  - compatible
+  - model
+  - spdif-controller
+
+anyOf:
+  - required:
+      - spdif-in
+  - required:
+      - spdif-out
+
+additionalProperties: false
+
+examples:
+  - |
+    sound-spdif {
+        compatible = "fsl,imx-audio-spdif";
+        model = "imx-spdif";
+        spdif-controller = <&spdif>;
+        spdif-out;
+        spdif-in;
+    };
diff --git a/Bindings/sound/fsl,sai.yaml b/Bindings/sound/fsl,sai.yaml
index 2456d95..a5d9c24 100644
--- a/Bindings/sound/fsl,sai.yaml
+++ b/Bindings/sound/fsl,sai.yaml
@@ -81,14 +81,12 @@
 
   dmas:
     minItems: 1
-    items:
-      - description: DMA controller phandle and request line for RX
-      - description: DMA controller phandle and request line for TX
+    maxItems: 2
 
   dma-names:
     minItems: 1
     items:
-      - const: rx
+      - enum: [ rx, tx ]
       - const: tx
 
   interrupts:
diff --git a/Bindings/sound/fsl,spdif.yaml b/Bindings/sound/fsl,spdif.yaml
index 1d64e83..204f361 100644
--- a/Bindings/sound/fsl,spdif.yaml
+++ b/Bindings/sound/fsl,spdif.yaml
@@ -31,7 +31,10 @@
     maxItems: 1
 
   interrupts:
-    maxItems: 1
+    minItems: 1
+    items:
+      - description: Combined or receive interrupt
+      - description: Transmit interrupt
 
   dmas:
     items:
@@ -86,6 +89,9 @@
       registers. Set this flag for HCDs with big endian descriptors and big
       endian registers.
 
+  power-domains:
+    maxItems: 1
+
 required:
   - compatible
   - reg
@@ -97,6 +103,33 @@
 
 additionalProperties: false
 
+allOf:
+  - if:
+      properties:
+        compatible:
+          enum:
+            - fsl,imx8qm-spdif
+            - fsl,imx8qxp-spdif
+    then:
+      properties:
+        interrupts:
+          minItems: 2
+    else:
+      properties:
+        interrupts:
+          maxItems: 1
+
+  - if:
+      properties:
+        compatible:
+          contains:
+            enum:
+              - fsl,imx8qm-spdif
+              - fsl,imx8qxp-spdif
+    then:
+      required:
+        - power-domains
+
 examples:
   - |
     spdif@2004000 {
diff --git a/Bindings/sound/fsl,ssi.txt b/Bindings/sound/fsl,ssi.txt
deleted file mode 100644
index 7e15a85..0000000
--- a/Bindings/sound/fsl,ssi.txt
+++ /dev/null
@@ -1,87 +0,0 @@
-Freescale Synchronous Serial Interface
-
-The SSI is a serial device that communicates with audio codecs.  It can
-be programmed in AC97, I2S, left-justified, or right-justified modes.
-
-Required properties:
-- compatible:       Compatible list, should contain one of the following
-                    compatibles:
-                      fsl,mpc8610-ssi
-                      fsl,imx51-ssi
-                      fsl,imx35-ssi
-                      fsl,imx21-ssi
-- cell-index:       The SSI, <0> = SSI1, <1> = SSI2, and so on.
-- reg:              Offset and length of the register set for the device.
-- interrupts:       <a b> where a is the interrupt number and b is a
-                    field that represents an encoding of the sense and
-                    level information for the interrupt.  This should be
-                    encoded based on the information in section 2)
-                    depending on the type of interrupt controller you
-                    have.
-- fsl,fifo-depth:   The number of elements in the transmit and receive FIFOs.
-                    This number is the maximum allowed value for SFCSR[TFWM0].
- - clocks:          "ipg" - Required clock for the SSI unit
-                    "baud" - Required clock for SSI master mode. Otherwise this
-		      clock is not used
-
-Required are also ac97 link bindings if ac97 is used. See
-Documentation/devicetree/bindings/sound/soc-ac97link.txt for the necessary
-bindings.
-
-Optional properties:
-- codec-handle:     Phandle to a 'codec' node that defines an audio
-                    codec connected to this SSI.  This node is typically
-                    a child of an I2C or other control node.
-- fsl,fiq-stream-filter: Bool property. Disabled DMA and use FIQ instead to
-		    filter the codec stream. This is necessary for some boards
-		    where an incompatible codec is connected to this SSI, e.g.
-		    on pca100 and pcm043.
-- dmas:		    Generic dma devicetree binding as described in
-		    Documentation/devicetree/bindings/dma/dma.txt.
-- dma-names:	    Two dmas have to be defined, "tx" and "rx", if fsl,imx-fiq
-		    is not defined.
-- fsl,mode:         The operating mode for the AC97 interface only.
-                    "ac97-slave" - AC97 mode, SSI is clock slave
-                    "ac97-master" - AC97 mode, SSI is clock master
-- fsl,ssi-asynchronous:
-                    If specified, the SSI is to be programmed in asynchronous
-                    mode.  In this mode, pins SRCK, STCK, SRFS, and STFS must
-                    all be connected to valid signals.  In synchronous mode,
-                    SRCK and SRFS are ignored.  Asynchronous mode allows
-                    playback and capture to use different sample sizes and
-                    sample rates.  Some drivers may require that SRCK and STCK
-                    be connected together, and SRFS and STFS be connected
-                    together.  This would still allow different sample sizes,
-                    but not different sample rates.
-- fsl,playback-dma: Phandle to a node for the DMA channel to use for
-                    playback of audio.  This is typically dictated by SOC
-                    design.  See the notes below.
-                    Only used on Power Architecture.
-- fsl,capture-dma:  Phandle to a node for the DMA channel to use for
-                    capture (recording) of audio.  This is typically dictated
-                    by SOC design.  See the notes below.
-                    Only used on Power Architecture.
-
-Child 'codec' node required properties:
-- compatible:       Compatible list, contains the name of the codec
-
-Child 'codec' node optional properties:
-- clock-frequency:  The frequency of the input clock, which typically comes
-                    from an on-board dedicated oscillator.
-
-Notes on fsl,playback-dma and fsl,capture-dma:
-
-On SOCs that have an SSI, specific DMA channels are hard-wired for playback
-and capture.  On the MPC8610, for example, SSI1 must use DMA channel 0 for
-playback and DMA channel 1 for capture.  SSI2 must use DMA channel 2 for
-playback and DMA channel 3 for capture.  The developer can choose which
-DMA controller to use, but the channels themselves are hard-wired.  The
-purpose of these two properties is to represent this hardware design.
-
-The device tree nodes for the DMA channels that are referenced by
-"fsl,playback-dma" and "fsl,capture-dma" must be marked as compatible with
-"fsl,ssi-dma-channel".  The SOC-specific compatible string (e.g.
-"fsl,mpc8610-dma-channel") can remain.  If these nodes are left as
-"fsl,elo-dma-channel" or "fsl,eloplus-dma-channel", then the generic Elo DMA
-drivers (fsldma) will attempt to use them, and it will conflict with the
-sound drivers.
diff --git a/Bindings/sound/fsl,ssi.yaml b/Bindings/sound/fsl,ssi.yaml
new file mode 100644
index 0000000..4ab10cd
--- /dev/null
+++ b/Bindings/sound/fsl,ssi.yaml
@@ -0,0 +1,194 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/fsl,ssi.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Freescale Synchronous Serial Interface
+
+maintainers:
+  - Shengjiu Wang <shengjiu.wang@nxp.com>
+
+description:
+  Notes on fsl,playback-dma and fsl,capture-dma
+  On SOCs that have an SSI, specific DMA channels are hard-wired for playback
+  and capture.  On the MPC8610, for example, SSI1 must use DMA channel 0 for
+  playback and DMA channel 1 for capture.  SSI2 must use DMA channel 2 for
+  playback and DMA channel 3 for capture.  The developer can choose which
+  DMA controller to use, but the channels themselves are hard-wired.  The
+  purpose of these two properties is to represent this hardware design.
+
+  The device tree nodes for the DMA channels that are referenced by
+  "fsl,playback-dma" and "fsl,capture-dma" must be marked as compatible with
+  "fsl,ssi-dma-channel".  The SOC-specific compatible string (e.g.
+  "fsl,mpc8610-dma-channel") can remain.  If these nodes are left as
+  "fsl,elo-dma-channel" or "fsl,eloplus-dma-channel", then the generic Elo DMA
+  drivers (fsldma) will attempt to use them, and it will conflict with the
+  sound drivers.
+
+properties:
+  compatible:
+    oneOf:
+      - items:
+          - enum:
+              - fsl,imx50-ssi
+              - fsl,imx53-ssi
+          - const: fsl,imx51-ssi
+          - const: fsl,imx21-ssi
+      - items:
+          - enum:
+              - fsl,imx25-ssi
+              - fsl,imx27-ssi
+              - fsl,imx35-ssi
+              - fsl,imx51-ssi
+          - const: fsl,imx21-ssi
+      - items:
+          - enum:
+              - fsl,imx6q-ssi
+              - fsl,imx6sl-ssi
+              - fsl,imx6sx-ssi
+          - const: fsl,imx51-ssi
+      - items:
+          - const: fsl,imx21-ssi
+      - items:
+          - const: fsl,mpc8610-ssi
+
+  reg:
+    maxItems: 1
+
+  interrupts:
+    maxItems: 1
+
+  clocks:
+    items:
+      - description: The ipg clock for register access
+      - description: clock for SSI master mode
+    minItems: 1
+
+  clock-names:
+    items:
+      - const: ipg
+      - const: baud
+    minItems: 1
+
+  dmas:
+    oneOf:
+      - items:
+          - description: DMA controller phandle and request line for RX
+          - description: DMA controller phandle and request line for TX
+      - items:
+          - description: DMA controller phandle and request line for RX0
+          - description: DMA controller phandle and request line for TX0
+          - description: DMA controller phandle and request line for RX1
+          - description: DMA controller phandle and request line for TX1
+
+  dma-names:
+    oneOf:
+      - items:
+          - const: rx
+          - const: tx
+      - items:
+          - const: rx0
+          - const: tx0
+          - const: rx1
+          - const: tx1
+
+  "#sound-dai-cells":
+    const: 0
+    description: optional, some dts node didn't add it.
+
+  cell-index:
+    $ref: /schemas/types.yaml#/definitions/uint32
+    enum: [0, 1, 2]
+    description: The SSI index
+
+  ac97-gpios:
+    $ref: /schemas/types.yaml#/definitions/phandle-array
+    description: Please refer to soc-ac97link.txt
+
+  codec-handle:
+    $ref: /schemas/types.yaml#/definitions/phandle
+    description:
+      Phandle to a 'codec' node that defines an audio
+      codec connected to this SSI.  This node is typically
+      a child of an I2C or other control node.
+
+  fsl,fifo-depth:
+    $ref: /schemas/types.yaml#/definitions/uint32
+    description:
+      The number of elements in the transmit and receive FIFOs.
+      This number is the maximum allowed value for SFCSR[TFWM0].
+    enum: [8, 15]
+
+  fsl,fiq-stream-filter:
+    type: boolean
+    description:
+      Disabled DMA and use FIQ instead to filter the codec stream.
+      This is necessary for some boards where an incompatible codec
+      is connected to this SSI, e.g. on pca100 and pcm043.
+
+  fsl,mode:
+    $ref: /schemas/types.yaml#/definitions/string
+    enum: [ ac97-slave, ac97-master, i2s-slave, i2s-master,
+            lj-slave, lj-master, rj-slave, rj-master ]
+    description: |
+      "ac97-slave" - AC97 mode, SSI is clock slave
+      "ac97-master" - AC97 mode, SSI is clock master
+      "i2s-slave" - I2S mode, SSI is clock slave
+      "i2s-master" - I2S mode, SSI is clock master
+      "lj-slave" - Left justified mode, SSI is clock slave
+      "lj-master" - Left justified mode, SSI is clock master
+      "rj-slave" - Right justified mode, SSI is clock slave
+      "rj-master" - Right justified mode, SSI is clock master
+
+  fsl,ssi-asynchronous:
+    type: boolean
+    description: If specified, the SSI is to be programmed in asynchronous
+      mode.  In this mode, pins SRCK, STCK, SRFS, and STFS must
+      all be connected to valid signals.  In synchronous mode,
+      SRCK and SRFS are ignored.  Asynchronous mode allows
+      playback and capture to use different sample sizes and
+      sample rates.  Some drivers may require that SRCK and STCK
+      be connected together, and SRFS and STFS be connected
+      together.  This would still allow different sample sizes,
+      but not different sample rates.
+
+  fsl,playback-dma:
+    $ref: /schemas/types.yaml#/definitions/phandle
+    description: Phandle to a node for the DMA channel to use for
+      playback of audio.  This is typically dictated by SOC
+      design. Only used on Power Architecture.
+
+  fsl,capture-dma:
+    $ref: /schemas/types.yaml#/definitions/phandle
+    description: Phandle to a node for the DMA channel to use for
+      capture (recording) of audio.  This is typically dictated
+      by SOC design. Only used on Power Architecture.
+
+required:
+  - compatible
+  - reg
+  - interrupts
+  - fsl,fifo-depth
+
+allOf:
+  - $ref: dai-common.yaml#
+
+unevaluatedProperties: false
+
+examples:
+  - |
+    #include <dt-bindings/interrupt-controller/arm-gic.h>
+    #include <dt-bindings/clock/imx6qdl-clock.h>
+    ssi@2028000 {
+        compatible = "fsl,imx6q-ssi", "fsl,imx51-ssi";
+        reg = <0x02028000 0x4000>;
+        interrupts = <GIC_SPI 46 IRQ_TYPE_LEVEL_HIGH>;
+        clocks = <&clks IMX6QDL_CLK_SSI1_IPG>,
+                 <&clks IMX6QDL_CLK_SSI1>;
+        clock-names = "ipg", "baud";
+        dmas = <&sdma 37 1 0>, <&sdma 38 1 0>;
+        dma-names = "rx", "tx";
+        #sound-dai-cells = <0>;
+        fsl,fifo-depth = <15>;
+    };
diff --git a/Bindings/sound/fsl-asoc-card.txt b/Bindings/sound/fsl-asoc-card.txt
deleted file mode 100644
index 4e8dbc5..0000000
--- a/Bindings/sound/fsl-asoc-card.txt
+++ /dev/null
@@ -1,117 +0,0 @@
-Freescale Generic ASoC Sound Card with ASRC support
-
-The Freescale Generic ASoC Sound Card can be used, ideally, for all Freescale
-SoCs connecting with external CODECs.
-
-The idea of this generic sound card is a bit like ASoC Simple Card. However,
-for Freescale SoCs (especially those released in recent years), most of them
-have ASRC (Documentation/devicetree/bindings/sound/fsl,asrc.txt) inside. And
-this is a specific feature that might be painstakingly controlled and merged
-into the Simple Card.
-
-So having this generic sound card allows all Freescale SoC users to benefit
-from the simplification of a new card support and the capability of the wide
-sample rates support through ASRC.
-
-Note: The card is initially designed for those sound cards who use AC'97, I2S
-      and PCM DAI formats. However, it'll be also possible to support those non
-      AC'97/I2S/PCM type sound cards, such as S/PDIF audio and HDMI audio, as
-      long as the driver has been properly upgraded.
-
-
-The compatible list for this generic sound card currently:
- "fsl,imx-audio-ac97"
-
- "fsl,imx-audio-cs42888"
-
- "fsl,imx-audio-cs427x"
- (compatible with CS4271 and CS4272)
-
- "fsl,imx-audio-wm8962"
-
- "fsl,imx-audio-sgtl5000"
- (compatible with Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt)
-
- "fsl,imx-audio-wm8960"
-
- "fsl,imx-audio-mqs"
-
- "fsl,imx-audio-wm8524"
-
- "fsl,imx-audio-tlv320aic32x4"
-
- "fsl,imx-audio-tlv320aic31xx"
-
- "fsl,imx-audio-si476x"
-
- "fsl,imx-audio-wm8958"
-
- "fsl,imx-audio-nau8822"
-
-Required properties:
-
-  - compatible		: Contains one of entries in the compatible list.
-
-  - model		: The user-visible name of this sound complex
-
-  - audio-cpu		: The phandle of an CPU DAI controller
-
-  - audio-codec		: The phandle of an audio codec
-
-Optional properties:
-
-  - audio-asrc		: The phandle of ASRC. It can be absent if there's no
-			  need to add ASRC support via DPCM.
-
-  - audio-routing	: A list of the connections between audio components.
-			  Each entry is a pair of strings, the first being the
-			  connection's sink, the second being the connection's
-			  source. There're a few pre-designed board connectors:
-			   * Line Out Jack
-			   * Line In Jack
-			   * Headphone Jack
-			   * Mic Jack
-			   * Ext Spk
-			   * AMIC (stands for Analog Microphone Jack)
-			   * DMIC (stands for Digital Microphone Jack)
-
-			  Note: The "Mic Jack" and "AMIC" are redundant while
-			        coexisting in order to support the old bindings
-				of wm8962 and sgtl5000.
-
-  - hp-det-gpio		: The GPIO that detect headphones are plugged in
-  - mic-det-gpio	: The GPIO that detect microphones are plugged in
-  - bitclock-master	: Indicates dai-link bit clock master; for details see simple-card.yaml.
-  - frame-master	: Indicates dai-link frame master; for details see simple-card.yaml.
-  - dai-format		: audio format, for details see simple-card.yaml.
-  - frame-inversion	: dai-link uses frame clock inversion, for details see simple-card.yaml.
-  - bitclock-inversion	: dai-link uses bit clock inversion, for details see simple-card.yaml.
-  - mclk-id		: main clock id, specific for each card configuration.
-
-Optional unless SSI is selected as a CPU DAI:
-
-  - mux-int-port	: The internal port of the i.MX audio muxer (AUDMUX)
-
-  - mux-ext-port	: The external port of the i.MX audio muxer
-
-Example:
-sound-cs42888 {
-	compatible = "fsl,imx-audio-cs42888";
-	model = "cs42888-audio";
-	audio-cpu = <&esai>;
-	audio-asrc = <&asrc>;
-	audio-codec = <&cs42888>;
-	audio-routing =
-		"Line Out Jack", "AOUT1L",
-		"Line Out Jack", "AOUT1R",
-		"Line Out Jack", "AOUT2L",
-		"Line Out Jack", "AOUT2R",
-		"Line Out Jack", "AOUT3L",
-		"Line Out Jack", "AOUT3R",
-		"Line Out Jack", "AOUT4L",
-		"Line Out Jack", "AOUT4R",
-		"AIN1L", "Line In Jack",
-		"AIN1R", "Line In Jack",
-		"AIN2L", "Line In Jack",
-		"AIN2R", "Line In Jack";
-};
diff --git a/Bindings/sound/fsl-asoc-card.yaml b/Bindings/sound/fsl-asoc-card.yaml
new file mode 100644
index 0000000..9922664
--- /dev/null
+++ b/Bindings/sound/fsl-asoc-card.yaml
@@ -0,0 +1,197 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/fsl-asoc-card.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Freescale Generic ASoC Sound Card with ASRC support
+
+description:
+  The Freescale Generic ASoC Sound Card can be used, ideally,
+  for all Freescale SoCs connecting with external CODECs.
+
+  The idea of this generic sound card is a bit like ASoC Simple Card.
+  However, for Freescale SoCs (especially those released in recent years),
+  most of them have ASRC inside. And this is a specific feature that might
+  be painstakingly controlled and merged into the Simple Card.
+
+  So having this generic sound card allows all Freescale SoC users to
+  benefit from the simplification of a new card support and the capability
+  of the wide sample rates support through ASRC.
+
+  Note, The card is initially designed for those sound cards who use AC'97, I2S
+  and PCM DAI formats. However, it'll be also possible to support those non
+  AC'97/I2S/PCM type sound cards, such as S/PDIF audio and HDMI audio, as
+  long as the driver has been properly upgraded.
+
+maintainers:
+  - Shengjiu Wang <shengjiu.wang@nxp.com>
+
+properties:
+  compatible:
+    oneOf:
+      - items:
+          - enum:
+              - fsl,imx-sgtl5000
+              - fsl,imx25-pdk-sgtl5000
+              - fsl,imx53-cpuvo-sgtl5000
+              - fsl,imx51-babbage-sgtl5000
+              - fsl,imx53-m53evk-sgtl5000
+              - fsl,imx53-qsb-sgtl5000
+              - fsl,imx53-voipac-sgtl5000
+              - fsl,imx6-armadeus-sgtl5000
+              - fsl,imx6-rex-sgtl5000
+              - fsl,imx6-sabreauto-cs42888
+              - fsl,imx6-wandboard-sgtl5000
+              - fsl,imx6dl-nit6xlite-sgtl5000
+              - fsl,imx6q-ba16-sgtl5000
+              - fsl,imx6q-nitrogen6_max-sgtl5000
+              - fsl,imx6q-nitrogen6_som2-sgtl5000
+              - fsl,imx6q-nitrogen6x-sgtl5000
+              - fsl,imx6q-sabrelite-sgtl5000
+              - fsl,imx6q-sabresd-wm8962
+              - fsl,imx6q-udoo-ac97
+              - fsl,imx6q-ventana-sgtl5000
+              - fsl,imx6sl-evk-wm8962
+              - fsl,imx6sx-sdb-mqs
+              - fsl,imx6sx-sdb-wm8962
+              - fsl,imx7d-evk-wm8960
+              - karo,tx53-audio-sgtl5000
+              - tq,imx53-mba53-sgtl5000
+          - enum:
+              - fsl,imx-audio-ac97
+              - fsl,imx-audio-cs42888
+              - fsl,imx-audio-mqs
+              - fsl,imx-audio-sgtl5000
+              - fsl,imx-audio-wm8960
+              - fsl,imx-audio-wm8962
+      - items:
+          - enum:
+              - fsl,imx-audio-ac97
+              - fsl,imx-audio-cs42888
+              - fsl,imx-audio-cs427x
+              - fsl,imx-audio-mqs
+              - fsl,imx-audio-nau8822
+              - fsl,imx-audio-sgtl5000
+              - fsl,imx-audio-si476x
+              - fsl,imx-audio-tlv320aic31xx
+              - fsl,imx-audio-tlv320aic32x4
+              - fsl,imx-audio-wm8524
+              - fsl,imx-audio-wm8904
+              - fsl,imx-audio-wm8960
+              - fsl,imx-audio-wm8962
+              - fsl,imx-audio-wm8958
+
+  model:
+    $ref: /schemas/types.yaml#/definitions/string
+    description: The user-visible name of this sound complex
+
+  audio-asrc:
+    $ref: /schemas/types.yaml#/definitions/phandle
+    description:
+      The phandle of ASRC. It can be absent if there's no
+      need to add ASRC support via DPCM.
+
+  audio-codec:
+    $ref: /schemas/types.yaml#/definitions/phandle
+    description: The phandle of an audio codec
+
+  audio-cpu:
+    $ref: /schemas/types.yaml#/definitions/phandle
+    description: The phandle of an CPU DAI controller
+
+  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. There're a few pre-designed board
+      connectors. "AMIC" stands for Analog Microphone Jack.
+      "DMIC" stands for Digital Microphone Jack. The "Mic Jack" and "AMIC"
+      are redundant while coexisting in order to support the old bindings
+      of wm8962 and sgtl5000.
+
+  hp-det-gpio:
+    deprecated: true
+    maxItems: 1
+    description: The GPIO that detect headphones are plugged in
+
+  hp-det-gpios:
+    maxItems: 1
+    description: The GPIO that detect headphones are plugged in
+
+  mic-det-gpio:
+    deprecated: true
+    maxItems: 1
+    description: The GPIO that detect microphones are plugged in
+
+  mic-det-gpios:
+    maxItems: 1
+    description: The GPIO that detect microphones are plugged in
+
+  bitclock-master:
+    $ref: simple-card.yaml#/definitions/bitclock-master
+    description: Indicates dai-link bit clock master.
+
+  frame-master:
+    $ref: simple-card.yaml#/definitions/frame-master
+    description: Indicates dai-link frame master.
+
+  format:
+    $ref: simple-card.yaml#/definitions/format
+    description: audio format.
+
+  frame-inversion:
+    $ref: simple-card.yaml#/definitions/frame-inversion
+    description: dai-link uses frame clock inversion.
+
+  bitclock-inversion:
+    $ref: simple-card.yaml#/definitions/bitclock-inversion
+    description: dai-link uses bit clock inversion.
+
+  mclk-id:
+    $ref: /schemas/types.yaml#/definitions/uint32
+    description: main clock id, specific for each card configuration.
+
+  mux-int-port:
+    $ref: /schemas/types.yaml#/definitions/uint32
+    enum: [1, 2, 7]
+    description: The internal port of the i.MX audio muxer (AUDMUX)
+
+  mux-ext-port:
+    $ref: /schemas/types.yaml#/definitions/uint32
+    enum: [3, 4, 5, 6]
+    description: The external port of the i.MX audio muxer
+
+  ssi-controller:
+    $ref: /schemas/types.yaml#/definitions/phandle
+    description: The phandle of an CPU DAI controller
+
+required:
+  - compatible
+  - model
+
+unevaluatedProperties: false
+
+examples:
+  - |
+    sound-cs42888 {
+        compatible = "fsl,imx-audio-cs42888";
+        model = "cs42888-audio";
+        audio-cpu = <&esai>;
+        audio-asrc = <&asrc>;
+        audio-codec = <&cs42888>;
+        audio-routing =
+             "Line Out Jack", "AOUT1L",
+             "Line Out Jack", "AOUT1R",
+             "Line Out Jack", "AOUT2L",
+             "Line Out Jack", "AOUT2R",
+             "Line Out Jack", "AOUT3L",
+             "Line Out Jack", "AOUT3R",
+             "Line Out Jack", "AOUT4L",
+             "Line Out Jack", "AOUT4R",
+             "AIN1L", "Line In Jack",
+             "AIN1R", "Line In Jack",
+             "AIN2L", "Line In Jack",
+             "AIN2R", "Line In Jack";
+    };
diff --git a/Bindings/sound/imx-audio-spdif.txt b/Bindings/sound/imx-audio-spdif.txt
deleted file mode 100644
index da84a44..0000000
--- a/Bindings/sound/imx-audio-spdif.txt
+++ /dev/null
@@ -1,36 +0,0 @@
-Freescale i.MX audio complex with S/PDIF transceiver
-
-Required properties:
-
-  - compatible		: "fsl,imx-audio-spdif"
-
-  - model		: The user-visible name of this sound complex
-
-  - spdif-controller	: The phandle of the i.MX S/PDIF controller
-
-
-Optional properties:
-
-  - spdif-out		: This is a boolean property. If present, the
-			  transmitting function of S/PDIF will be enabled,
-			  indicating there's a physical S/PDIF out connector
-			  or jack on the board or it's connecting to some
-			  other IP block, such as an HDMI encoder or
-			  display-controller.
-
-  - spdif-in		: This is a boolean property. If present, the receiving
-			  function of S/PDIF will be enabled, indicating there
-			  is a physical S/PDIF in connector/jack on the board.
-
-* Note: At least one of these two properties should be set in the DT binding.
-
-
-Example:
-
-sound-spdif {
-	compatible = "fsl,imx-audio-spdif";
-	model = "imx-spdif";
-	spdif-controller = <&spdif>;
-	spdif-out;
-	spdif-in;
-};
diff --git a/Bindings/sound/mediatek,mt2701-wm8960.yaml b/Bindings/sound/mediatek,mt2701-wm8960.yaml
new file mode 100644
index 0000000..cf98546
--- /dev/null
+++ b/Bindings/sound/mediatek,mt2701-wm8960.yaml
@@ -0,0 +1,54 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/mediatek,mt2701-wm8960.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: MediaTek MT2701 with WM8960 CODEC
+
+maintainers:
+  - Kartik Agarwala <agarwala.kartik@gmail.com>
+
+properties:
+  compatible:
+    const: mediatek,mt2701-wm8960-machine
+
+  mediatek,platform:
+    $ref: /schemas/types.yaml#/definitions/phandle
+    description: The phandle of MT2701 ASoC platform.
+
+  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.
+
+  mediatek,audio-codec:
+    $ref: /schemas/types.yaml#/definitions/phandle
+    description: The phandle of the WM8960 audio codec.
+
+unevaluatedProperties: false
+
+required:
+  - compatible
+  - mediatek,platform
+  - audio-routing
+  - mediatek,audio-codec
+  - pinctrl-names
+  - pinctrl-0
+
+examples:
+  - |
+    sound {
+        compatible = "mediatek,mt2701-wm8960-machine";
+        mediatek,platform = <&afe>;
+        audio-routing =
+            "Headphone", "HP_L",
+            "Headphone", "HP_R",
+            "LINPUT1", "AMIC",
+            "RINPUT1", "AMIC";
+        mediatek,audio-codec = <&wm8960>;
+        pinctrl-names = "default";
+        pinctrl-0 = <&aud_pins_default>;
+    };
diff --git a/Bindings/sound/mt2701-wm8960.txt b/Bindings/sound/mt2701-wm8960.txt
deleted file mode 100644
index 809b609..0000000
--- a/Bindings/sound/mt2701-wm8960.txt
+++ /dev/null
@@ -1,24 +0,0 @@
-MT2701 with WM8960 CODEC
-
-Required properties:
-- compatible: "mediatek,mt2701-wm8960-machine"
-- mediatek,platform: the phandle of MT2701 ASoC platform
-- audio-routing: a list of the connections between audio
-- mediatek,audio-codec: the phandles of wm8960 codec
-- pinctrl-names: Should contain only one value - "default"
-- pinctrl-0: Should specify pin control groups used for this controller.
-
-Example:
-
-	sound:sound {
-		compatible = "mediatek,mt2701-wm8960-machine";
-		mediatek,platform = <&afe>;
-		audio-routing =
-			"Headphone", "HP_L",
-			"Headphone", "HP_R",
-			"LINPUT1", "AMIC",
-			"RINPUT1", "AMIC";
-		mediatek,audio-codec = <&wm8960>;
-		pinctrl-names = "default";
-		pinctrl-0 = <&aud_pins_default>;
-	};
diff --git a/Bindings/sound/mt8186-mt6366-da7219-max98357.yaml b/Bindings/sound/mt8186-mt6366-da7219-max98357.yaml
index 9853c11..cbc641e 100644
--- a/Bindings/sound/mt8186-mt6366-da7219-max98357.yaml
+++ b/Bindings/sound/mt8186-mt6366-da7219-max98357.yaml
@@ -12,17 +12,46 @@
 description:
   This binding describes the MT8186 sound card.
 
+allOf:
+  - $ref: sound-card-common.yaml#
+
 properties:
   compatible:
     enum:
       - mediatek,mt8186-mt6366-da7219-max98357-sound
 
+  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.
+      Valid names could be the input or output widgets of audio components,
+      power supplies, MicBias of codec and the software switch.
+    minItems: 2
+    items:
+      enum:
+        # Sinks
+        - HDMI1
+        - Headphones
+        - Line Out
+        - MIC
+        - Speakers
+
+        # Sources
+        - Headset Mic
+        - HPL
+        - HPR
+        - Speaker
+        - TX
+
   mediatek,platform:
     $ref: /schemas/types.yaml#/definitions/phandle
     description: The phandle of MT8186 ASoC platform.
 
   headset-codec:
     type: object
+    deprecated: true
     additionalProperties: false
     properties:
       sound-dai:
@@ -32,6 +61,7 @@
 
   playback-codecs:
     type: object
+    deprecated: true
     additionalProperties: false
     properties:
       sound-dai:
@@ -53,32 +83,115 @@
       A list of the desired dai-links in the sound card. Each entry is a
       name defined in the machine driver.
 
+patternProperties:
+  ".*-dai-link$":
+    type: object
+    additionalProperties: false
+    description:
+      Container for dai-link level properties and CODEC sub-nodes.
+
+    properties:
+      link-name:
+        description: Indicates dai-link name and PCM stream name
+        items:
+          enum:
+            - I2S0
+            - I2S1
+            - I2S2
+            - I2S3
+
+      codec:
+        description: Holds subnode which indicates codec dai.
+        type: object
+        additionalProperties: false
+        properties:
+          sound-dai:
+            minItems: 1
+            maxItems: 2
+        required:
+          - sound-dai
+
-additionalProperties: false
+      dai-format:
+        description: audio format
+        items:
+          enum:
+            - i2s
+            - right_j
+            - left_j
+            - dsp_a
+            - dsp_b
+
+      mediatek,clk-provider:
+        $ref: /schemas/types.yaml#/definitions/string
+        description: Indicates dai-link clock master.
+        items:
+          enum:
+            - cpu
+            - codec
+
+    required:
+      - link-name
+
+unevaluatedProperties: false
 
 required:
   - compatible
   - mediatek,platform
-  - headset-codec
-  - playback-codecs
+
+# Disallow legacy properties if xxx-dai-link nodes are specified
+if:
+  not:
+    patternProperties:
+      ".*-dai-link$": false
+then:
+  properties:
+    headset-codec: false
+    speaker-codecs: false
 
 examples:
   - |
 
     sound: mt8186-sound {
         compatible = "mediatek,mt8186-mt6366-da7219-max98357-sound";
-        mediatek,platform = <&afe>;
+        model = "mt8186_da7219_m98357";
         pinctrl-names = "aud_clk_mosi_off",
                         "aud_clk_mosi_on";
         pinctrl-0 = <&aud_clk_mosi_off>;
         pinctrl-1 = <&aud_clk_mosi_on>;
+        mediatek,platform = <&afe>;
 
-        headset-codec {
-            sound-dai = <&da7219>;
+        audio-routing =
+                "Headphones", "HPL",
+                "Headphones", "HPR",
+                "MIC", "Headset Mic",
+                "Speakers", "Speaker",
+                "HDMI1", "TX";
+
+        hs-playback-dai-link {
+                link-name = "I2S0";
+                dai-format = "i2s";
+                mediatek,clk-provider = "cpu";
+                codec {
+                        sound-dai = <&da7219>;
+                };
+        };
+
+        hs-capture-dai-link {
+                link-name = "I2S1";
+                dai-format = "i2s";
+                mediatek,clk-provider = "cpu";
+                codec {
+                        sound-dai = <&da7219>;
+                };
         };
 
-        playback-codecs {
-            sound-dai = <&anx_bridge_dp>,
-                        <&max98357a>;
+        spk-dp-playback-dai-link {
+                link-name = "I2S3";
+                dai-format = "i2s";
+                mediatek,clk-provider = "cpu";
+                codec {
+                        sound-dai = <&anx_bridge_dp>, <&max98357a>;
+                };
         };
     };
 
diff --git a/Bindings/sound/mt8186-mt6366-rt1019-rt5682s.yaml b/Bindings/sound/mt8186-mt6366-rt1019-rt5682s.yaml
index bdf7b09..ed93f18 100644
--- a/Bindings/sound/mt8186-mt6366-rt1019-rt5682s.yaml
+++ b/Bindings/sound/mt8186-mt6366-rt1019-rt5682s.yaml
@@ -12,6 +12,9 @@
 description:
   This binding describes the MT8186 sound card.
 
+allOf:
+  - $ref: sound-card-common.yaml#
+
 properties:
   compatible:
     enum:
@@ -19,6 +22,34 @@
       - mediatek,mt8186-mt6366-rt5682s-max98360-sound
       - mediatek,mt8186-mt6366-rt5650-sound
 
+  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.
+      Valid names could be the input or output widgets of audio components,
+      power supplies, MicBias of codec and the software switch.
+    minItems: 2
+    items:
+      enum:
+        # Sinks
+        - HDMI1
+        - Headphone
+        - IN1P
+        - IN1N
+        - Line Out
+        - Speakers
+
+        # Sources
+        - Headset Mic
+        - HPOL
+        - HPOR
+        - Speaker
+        - SPOL
+        - SPOR
+        - TX
+
   mediatek,platform:
     $ref: /schemas/types.yaml#/definitions/phandle
     description: The phandle of MT8186 ASoC platform.
@@ -32,6 +63,7 @@
 
   headset-codec:
     type: object
+    deprecated: true
     additionalProperties: false
     properties:
       sound-dai:
@@ -41,6 +73,7 @@
 
   playback-codecs:
     type: object
+    deprecated: true
     additionalProperties: false
     properties:
       sound-dai:
@@ -62,13 +95,56 @@
       A list of the desired dai-links in the sound card. Each entry is a
       name defined in the machine driver.
 
+patternProperties:
+  ".*-dai-link$":
+    type: object
+    additionalProperties: false
+    description:
+      Container for dai-link level properties and CODEC sub-nodes.
+
+    properties:
+      link-name:
+        description: Indicates dai-link name and PCM stream name
+        enum: [ I2S0, I2S1, I2S2, I2S3 ]
+
+      codec:
+        description: Holds subnode which indicates codec dai.
+        type: object
+        additionalProperties: false
+        properties:
+          sound-dai:
+            minItems: 1
+            maxItems: 2
+        required:
+          - sound-dai
+
-additionalProperties: false
+      dai-format:
+        description: audio format
+        enum: [ i2s, right_j, left_j, dsp_a, dsp_b ]
+
+      mediatek,clk-provider:
+        $ref: /schemas/types.yaml#/definitions/string
+        description: Indicates dai-link clock master.
+        enum: [ cpu, codec ]
+
+    required:
+      - link-name
+
+unevaluatedProperties: false
 
 required:
   - compatible
   - mediatek,platform
-  - headset-codec
-  - playback-codecs
+
+# Disallow legacy properties if xxx-dai-link nodes are specified
+if:
+  not:
+    patternProperties:
+      ".*-dai-link$": false
+then:
+  properties:
+    headset-codec: false
+    speaker-codecs: false
 
 examples:
   - |
@@ -76,23 +152,49 @@
 
     sound: mt8186-sound {
         compatible = "mediatek,mt8186-mt6366-rt1019-rt5682s-sound";
-        mediatek,platform = <&afe>;
+        model = "mt8186_rt1019_rt5682s";
         pinctrl-names = "aud_clk_mosi_off",
                         "aud_clk_mosi_on",
                         "aud_gpio_dmic_sec";
         pinctrl-0 = <&aud_clk_mosi_off>;
         pinctrl-1 = <&aud_clk_mosi_on>;
         pinctrl-2 = <&aud_gpio_dmic_sec>;
+        mediatek,platform = <&afe>;
 
         dmic-gpios = <&pio 23 GPIO_ACTIVE_HIGH>;
 
+        audio-routing =
+                "Headphone", "HPOL",
+                "Headphone", "HPOR",
+                "IN1P", "Headset Mic",
+                "Speakers", "Speaker",
+                "HDMI1", "TX";
+
-        headset-codec {
-            sound-dai = <&rt5682s>;
+        hs-playback-dai-link {
+                link-name = "I2S0";
+                dai-format = "i2s";
+                mediatek,clk-provider = "cpu";
+                codec {
+                        sound-dai = <&rt5682s 0>;
+                };
         };
 
-        playback-codecs {
-             sound-dai = <&it6505dptx>,
-                         <&rt1019p>;
+        hs-capture-dai-link {
+                link-name = "I2S1";
+                dai-format = "i2s";
+                mediatek,clk-provider = "cpu";
+                codec {
+                        sound-dai = <&rt5682s 0>;
+                };
+        };
+
+        spk-hdmi-playback-dai-link {
+                link-name = "I2S3";
+                dai-format = "i2s";
+                mediatek,clk-provider = "cpu";
+                codec {
+                        sound-dai = <&it6505dptx>, <&rt1019p>;
+                };
         };
     };
 
diff --git a/Bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml b/Bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml
index 7e50f5d..c4e68f3 100644
--- a/Bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml
+++ b/Bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml
@@ -13,6 +13,9 @@
 description:
   This binding describes the MT8192 sound card.
 
+allOf:
+  - $ref: sound-card-common.yaml#
+
 properties:
   compatible:
     enum:
@@ -20,6 +23,31 @@
       - mediatek,mt8192_mt6359_rt1015p_rt5682
       - mediatek,mt8192_mt6359_rt1015p_rt5682s
 
+  audio-routing:
+    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.
+      Valid names could be the input or output widgets of audio components,
+      power supplies, MicBias of codec and the software switch.
+    minItems: 2
+    items:
+      enum:
+        # Sinks
+        - Speakers
+        - Headphone Jack
+        - IN1P
+        - Left Spk
+        - Right Spk
+
+        # Sources
+        - Headset Mic
+        - HPOL
+        - HPOR
+        - Left SPO
+        - Right SPO
+        - Speaker
+
   mediatek,platform:
     $ref: /schemas/types.yaml#/definitions/phandle
     description: The phandle of MT8192 ASoC platform.
@@ -27,10 +55,12 @@
   mediatek,hdmi-codec:
     $ref: /schemas/types.yaml#/definitions/phandle
     description: The phandle of HDMI codec.
+    deprecated: true
 
   headset-codec:
     type: object
     additionalProperties: false
+    deprecated: true
 
     properties:
       sound-dai:
@@ -41,6 +71,7 @@
   speaker-codecs:
     type: object
     additionalProperties: false
+    deprecated: true
 
     properties:
       sound-dai:
@@ -51,33 +82,121 @@
     required:
       - sound-dai
 
+patternProperties:
+  ".*-dai-link$":
+    type: object
+    additionalProperties: false
+
+    description:
+      Container for dai-link level properties and CODEC sub-nodes.
+
+    properties:
+      link-name:
+        description: Indicates dai-link name and PCM stream name
+        enum:
+          - I2S0
+          - I2S1
+          - I2S2
+          - I2S3
+          - I2S4
+          - I2S5
+          - I2S6
+          - I2S7
+          - I2S8
+          - I2S9
+          - TDM
+
+      codec:
+        description: Holds subnode which indicates codec dai.
+        type: object
+        additionalProperties: false
+        properties:
+          sound-dai:
+            minItems: 1
+            maxItems: 2
+        required:
+          - sound-dai
+
+      dai-format:
+        description: audio format
+        enum: [ i2s, right_j, left_j, dsp_a, dsp_b ]
+
-additionalProperties: false
+      mediatek,clk-provider:
+        $ref: /schemas/types.yaml#/definitions/string
+        description: Indicates dai-link clock master.
+        enum: [ cpu, codec ]
+
+    required:
+      - link-name
+
+unevaluatedProperties: false
 
 required:
   - compatible
   - mediatek,platform
-  - headset-codec
-  - speaker-codecs
+
+# Disallow legacy properties if xxx-dai-link nodes are specified
+if:
+  not:
+    patternProperties:
+      ".*-dai-link$": false
+then:
+  properties:
+    headset-codec: false
+    speaker-codecs: false
+    mediatek,hdmi-codec: false
 
 examples:
   - |
 
     sound: mt8192-sound {
         compatible = "mediatek,mt8192_mt6359_rt1015_rt5682";
-        mediatek,platform = <&afe>;
-        mediatek,hdmi-codec = <&anx_bridge_dp>;
+        model = "mt8192_mt6359_rt1015_rt5682";
         pinctrl-names = "aud_clk_mosi_off",
                         "aud_clk_mosi_on";
         pinctrl-0 = <&aud_clk_mosi_off>;
         pinctrl-1 = <&aud_clk_mosi_on>;
+        mediatek,platform = <&afe>;
 
-        headset-codec {
-            sound-dai = <&rt5682>;
+        audio-routing =
+                "Headphone Jack", "HPOL",
+                "Headphone Jack", "HPOR",
+                "IN1P", "Headset Mic",
+                "Speakers", "Speaker";
+
+        spk-playback-dai-link {
+                link-name = "I2S3";
+                dai-format = "i2s";
+                mediatek,clk-provider = "cpu";
+                codec {
+                        sound-dai = <&rt1015p>;
+                };
+        };
+
+        hs-playback-dai-link {
+                link-name = "I2S8";
+                dai-format = "i2s";
+                mediatek,clk-provider = "cpu";
+                codec {
+                        sound-dai = <&rt5682 0>;
+                };
+        };
+
+        hs-capture-dai-link {
+                link-name = "I2S9";
+                dai-format = "i2s";
+                mediatek,clk-provider = "cpu";
+                codec {
+                        sound-dai = <&rt5682 0>;
+                };
         };
 
-        speaker-codecs {
-            sound-dai = <&rt1015_l>,
-                        <&rt1015_r>;
+        displayport-dai-link {
+                link-name = "TDM";
+                dai-format = "dsp_a";
+                codec {
+                        sound-dai = <&anx_bridge_dp>;
+                };
         };
     };
 
diff --git a/Bindings/sound/mt8195-mt6359.yaml b/Bindings/sound/mt8195-mt6359.yaml
index c1ddbf6..2af1d8f 100644
--- a/Bindings/sound/mt8195-mt6359.yaml
+++ b/Bindings/sound/mt8195-mt6359.yaml
@@ -12,6 +12,9 @@
 description:
   This binding describes the MT8195 sound card.
 
+allOf:
+  - $ref: sound-card-common.yaml#
+
 properties:
   compatible:
     enum:
@@ -23,6 +26,33 @@
     $ref: /schemas/types.yaml#/definitions/string
     description: User specified audio sound card name
 
+  audio-routing:
+    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.
+      Valid names could be the input or output widgets of audio components,
+      power supplies, MicBias of codec and the software switch.
+    minItems: 2
+    items:
+      enum:
+        # Sinks
+        - Ext Spk
+        - Headphone
+        - IN1P
+        - Left Spk
+        - Right Spk
+
+        # Sources
+        - Headset Mic
+        - HPOL
+        - HPOR
+        - Left BE_OUT
+        - Left SPO
+        - Right BE_OUT
+        - Right SPO
+        - Speaker
+
   mediatek,platform:
     $ref: /schemas/types.yaml#/definitions/phandle
     description: The phandle of MT8195 ASoC platform.
@@ -30,10 +60,12 @@
   mediatek,dptx-codec:
     $ref: /schemas/types.yaml#/definitions/phandle
     description: The phandle of MT8195 Display Port Tx codec node.
+    deprecated: true
 
   mediatek,hdmi-codec:
     $ref: /schemas/types.yaml#/definitions/phandle
     description: The phandle of MT8195 HDMI codec node.
+    deprecated: true
 
   mediatek,adsp:
     $ref: /schemas/types.yaml#/definitions/phandle
@@ -45,20 +77,122 @@
       A list of the desired dai-links in the sound card. Each entry is a
       name defined in the machine driver.
 
+patternProperties:
+  ".*-dai-link$":
+    type: object
+    additionalProperties: false
+    description:
+      Container for dai-link level properties and CODEC sub-nodes.
+
+    properties:
+      link-name:
+        description: Indicates dai-link name and PCM stream name
+        enum:
+          - 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
+
+      dai-format:
+        description: audio format
+        enum: [ i2s, right_j, left_j, dsp_a, dsp_b ]
+
+      mediatek,clk-provider:
+        $ref: /schemas/types.yaml#/definitions/string
+        description: Indicates dai-link clock master.
+        enum: [ cpu, codec ]
+
+    required:
+      - link-name
+
 additionalProperties: false
 
 required:
   - compatible
   - mediatek,platform
 
+# Disallow legacy properties if xxx-dai-link nodes are specified
+if:
+  not:
+    patternProperties:
+      ".*-dai-link$": false
+then:
+  properties:
+    mediatek,dptx-codec: false
+    mediatek,hdmi-codec: false
+
 examples:
   - |
 
     sound: mt8195-sound {
         compatible = "mediatek,mt8195_mt6359_rt1019_rt5682";
+        model = "mt8195_r1019_5682";
         mediatek,platform = <&afe>;
         pinctrl-names = "default";
         pinctrl-0 = <&aud_pins_default>;
+
+        audio-routing =
+                "Headphone", "HPOL",
+                "Headphone", "HPOR",
+                "IN1P", "Headset Mic",
+                "Ext Spk", "Speaker";
+
+        mm-dai-link {
+                link-name = "ETDM1_IN_BE";
+                mediatek,clk-provider = "cpu";
+        };
+
+        hs-playback-dai-link {
+                link-name = "ETDM1_OUT_BE";
+                mediatek,clk-provider = "cpu";
+                codec {
+                        sound-dai = <&headset_codec>;
+                };
+        };
+
+        hs-capture-dai-link {
+                link-name = "ETDM2_IN_BE";
+                mediatek,clk-provider = "cpu";
+                codec {
+                        sound-dai = <&headset_codec>;
+                };
+        };
+
+        spk-playback-dai-link {
+                link-name = "ETDM2_OUT_BE";
+                mediatek,clk-provider = "cpu";
+                codec {
+                        sound-dai = <&spk_amplifier>;
+                };
+        };
+
+        hdmi-dai-link {
+                link-name = "ETDM3_OUT_BE";
+                codec {
+                        sound-dai = <&hdmi_tx>;
+                };
+        };
+
+        displayport-dai-link {
+                link-name = "DPTX_BE";
+                codec {
+                        sound-dai = <&dp_tx>;
+                };
+        };
     };
 
 ...
diff --git a/Bindings/sound/nuvoton,nau8325.yaml b/Bindings/sound/nuvoton,nau8325.yaml
new file mode 100644
index 0000000..979be0d
--- /dev/null
+++ b/Bindings/sound/nuvoton,nau8325.yaml
@@ -0,0 +1,80 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/nuvoton,nau8325.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: NAU8325 audio Amplifier
+
+maintainers:
+  - Seven Lee <WTLI@nuvoton.com>
+
+allOf:
+  - $ref: dai-common.yaml#
+
+properties:
+  compatible:
+    const: nuvoton,nau8325
+
+  reg:
+    maxItems: 1
+
+  nuvoton,vref-impedance-ohms:
+    description:
+      The vref impedance to be used in ohms. Middle of voltage enables
+      Tie-Off selection options. Due to the high impedance of the VREF
+      pin, it is important to use a low-leakage capacitor.
+
+    enum: [0, 25000, 125000, 2500]
+
+  nuvoton,dac-vref-microvolt:
+    description:
+      The DAC vref to be used in voltage. DAC reference voltage setting. Can
+      be used for minor tuning of the output level. Since the VDDA is range
+      between 1.62 to 1.98 voltage, the typical value for design is 1.8V. After
+      the minor tuning, the final microvolt are as the below.
+
+    enum: [1800000, 2700000, 2880000, 3060000]
+
+  nuvoton,alc-enable:
+    description:
+      Enable digital automatic level control (ALC) function.
+    type: boolean
+
+  nuvoton,clock-detection-disable:
+    description:
+      When clock detection is enabled, it will detect whether MCLK
+      and FS are within the range. MCLK range is from 2.048MHz to 24.576MHz.
+      FS range is from 8kHz to 96kHz. And also needs to detect the ratio
+      MCLK_SRC/LRCK of 256, 400 or 500, and needs to detect the BCLK
+      to make sure data is present. There needs to be at least 8 BCLK
+      cycles per Frame Sync.
+    type: boolean
+
+  nuvoton,clock-det-data:
+    description:
+      Request clock detection to require 2048 non-zero samples before enabling
+      the audio paths. If set then non-zero samples is required, otherwise it
+      doesn't matter.
+    type: boolean
+
+required:
+  - compatible
+  - reg
+
+unevaluatedProperties: false
+
+examples:
+  - |
+    i2c {
+        #address-cells = <1>;
+        #size-cells = <0>;
+        codec@21 {
+            compatible = "nuvoton,nau8325";
+            reg = <0x21>;
+            nuvoton,vref-impedance-ohms = <125000>;
+            nuvoton,dac-vref-microvolt = <2880000>;
+            nuvoton,alc-enable;
+            nuvoton,clock-det-data;
+        };
+    };
diff --git a/Bindings/sound/nuvoton,nau8821.yaml b/Bindings/sound/nuvoton,nau8821.yaml
index 054b539..9f44168 100644
--- a/Bindings/sound/nuvoton,nau8821.yaml
+++ b/Bindings/sound/nuvoton,nau8821.yaml
@@ -103,6 +103,12 @@
         just limited to the left adc for design demand.
     type: boolean
 
+  nuvoton,adc-delay-ms:
+    description: Delay (in ms) to make input path stable and avoid pop noise.
+    minimum: 125
+    maximum: 500
+    default: 125
+
   '#sound-dai-cells':
     const: 0
 
@@ -136,6 +142,7 @@
             nuvoton,jack-eject-debounce = <0>;
             nuvoton,dmic-clk-threshold = <3072000>;
             nuvoton,dmic-slew-rate = <0>;
+            nuvoton,adc-delay-ms = <125>;
             #sound-dai-cells = <0>;
         };
     };
diff --git a/Bindings/sound/nvidia,tegra20-ac97.txt b/Bindings/sound/nvidia,tegra20-ac97.txt
deleted file mode 100644
index eaf0010..0000000
--- a/Bindings/sound/nvidia,tegra20-ac97.txt
+++ /dev/null
@@ -1,36 +0,0 @@
-NVIDIA Tegra 20 AC97 controller
-
-Required properties:
-- compatible : "nvidia,tegra20-ac97"
-- reg : Should contain AC97 controller registers location and length
-- interrupts : Should contain AC97 interrupt
-- resets : Must contain an entry for each entry in reset-names.
-  See ../reset/reset.txt for details.
-- reset-names : Must include the following entries:
-  - ac97
-- dmas : Must contain an entry for each entry in clock-names.
-  See ../dma/dma.txt for details.
-- dma-names : Must include the following entries:
-  - rx
-  - tx
-- clocks : Must contain one entry, for the module clock.
-  See ../clocks/clock-bindings.txt for details.
-- nvidia,codec-reset-gpio : The Tegra GPIO controller's phandle and the number
-  of the GPIO used to reset the external AC97 codec
-- nvidia,codec-sync-gpio : The Tegra GPIO controller's phandle and the number
-  of the GPIO corresponding with the AC97 DAP _FS line
-
-Example:
-
-ac97@70002000 {
-	compatible = "nvidia,tegra20-ac97";
-	reg = <0x70002000 0x200>;
-	interrupts = <0 81 0x04>;
-	nvidia,codec-reset-gpio = <&gpio 170 0>;
-	nvidia,codec-sync-gpio = <&gpio 120 0>;
-	clocks = <&tegra_car 3>;
-	resets = <&tegra_car 3>;
-	reset-names = "ac97";
-	dmas = <&apbdma 12>, <&apbdma 12>;
-	dma-names = "rx", "tx";
-};
diff --git a/Bindings/sound/nvidia,tegra20-ac97.yaml b/Bindings/sound/nvidia,tegra20-ac97.yaml
new file mode 100644
index 0000000..4ea0a30
--- /dev/null
+++ b/Bindings/sound/nvidia,tegra20-ac97.yaml
@@ -0,0 +1,82 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/nvidia,tegra20-ac97.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: NVIDIA Tegra20 AC97 controller
+
+maintainers:
+  - Thierry Reding <treding@nvidia.com>
+  - Jon Hunter <jonathanh@nvidia.com>
+
+properties:
+  compatible:
+    const: nvidia,tegra20-ac97
+
+  reg:
+    maxItems: 1
+
+  resets:
+    maxItems: 1
+
+  reset-names:
+    const: ac97
+
+  interrupts:
+    maxItems: 1
+
+  clocks:
+    maxItems: 1
+
+  dmas:
+    maxItems: 2
+
+  dma-names:
+    items:
+      - const: rx
+      - const: tx
+
+  nvidia,codec-reset-gpios:
+    description: Reset pin of external AC97 codec
+    maxItems: 1
+
+  nvidia,codec-sync-gpios:
+    description: AC97 DAP _FS line
+    maxItems: 1
+
+required:
+  - compatible
+  - reg
+  - resets
+  - reset-names
+  - interrupts
+  - clocks
+  - dmas
+  - dma-names
+  - nvidia,codec-reset-gpios
+  - nvidia,codec-sync-gpios
+
+additionalProperties: false
+
+examples:
+  - |
+    #include <dt-bindings/clock/tegra20-car.h>
+    #include <dt-bindings/gpio/tegra-gpio.h>
+    #include <dt-bindings/interrupt-controller/arm-gic.h>
+    #include <dt-bindings/interrupt-controller/irq.h>
+    #include <dt-bindings/gpio/gpio.h>
+
+    ac97@70002000 {
+        compatible = "nvidia,tegra20-ac97";
+        reg = <0x70002000 0x200>;
+        resets = <&tegra_car 3>;
+        reset-names = "ac97";
+        interrupts = <GIC_SPI 81 IRQ_TYPE_LEVEL_HIGH>;
+        clocks = <&tegra_car 3>;
+        dmas = <&apbdma 12>, <&apbdma 12>;
+        dma-names = "rx", "tx";
+        nvidia,codec-reset-gpios = <&gpio TEGRA_GPIO(V, 2) GPIO_ACTIVE_HIGH>;
+        nvidia,codec-sync-gpios = <&gpio TEGRA_GPIO(P, 0) GPIO_ACTIVE_HIGH>;
+    };
+...
diff --git a/Bindings/sound/nvidia,tegra20-das.txt b/Bindings/sound/nvidia,tegra20-das.txt
deleted file mode 100644
index 6de3a7e..0000000
--- a/Bindings/sound/nvidia,tegra20-das.txt
+++ /dev/null
@@ -1,12 +0,0 @@
-NVIDIA Tegra 20 DAS (Digital Audio Switch) controller
-
-Required properties:
-- compatible : "nvidia,tegra20-das"
-- reg : Should contain DAS registers location and length
-
-Example:
-
-das@70000c00 {
-	compatible = "nvidia,tegra20-das";
-	reg = <0x70000c00 0x80>;
-};
diff --git a/Bindings/sound/nvidia,tegra20-das.yaml b/Bindings/sound/nvidia,tegra20-das.yaml
new file mode 100644
index 0000000..44c5ce8
--- /dev/null
+++ b/Bindings/sound/nvidia,tegra20-das.yaml
@@ -0,0 +1,36 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/nvidia,tegra20-das.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: NVIDIA Tegra 20 DAS (Digital Audio Switch) controller
+
+maintainers:
+  - Thierry Reding <treding@nvidia.com>
+  - Jon Hunter <jonathanh@nvidia.com>
+
+properties:
+  compatible:
+    const: nvidia,tegra20-das
+
+  reg:
+    maxItems: 1
+
+required:
+  - compatible
+  - reg
+
+additionalProperties: false
+
+examples:
+  - |
+    bus {
+        #address-cells = <1>;
+        #size-cells = <1>;
+        das@70000c00 {
+            compatible = "nvidia,tegra20-das";
+            reg = <0x70000c00 0x80>;
+        };
+    };
+...
diff --git a/Bindings/sound/nvidia,tegra30-i2s.txt b/Bindings/sound/nvidia,tegra30-i2s.txt
deleted file mode 100644
index 38caa93..0000000
--- a/Bindings/sound/nvidia,tegra30-i2s.txt
+++ /dev/null
@@ -1,27 +0,0 @@
-NVIDIA Tegra30 I2S controller
-
-Required properties:
-- compatible : For Tegra30, must contain "nvidia,tegra30-i2s".  For Tegra124,
-  must contain "nvidia,tegra124-i2s".  Otherwise, must contain
-  "nvidia,<chip>-i2s" plus at least one of the above, where <chip> is
-  tegra114 or tegra132.
-- reg : Should contain I2S registers location and length
-- clocks : Must contain one entry, for the module clock.
-  See ../clocks/clock-bindings.txt for details.
-- resets : Must contain an entry for each entry in reset-names.
-  See ../reset/reset.txt for details.
-- reset-names : Must include the following entries:
-  - i2s
-- nvidia,ahub-cif-ids : The list of AHUB CIF IDs for this port, rx (playback)
-  first, tx (capture) second. See nvidia,tegra30-ahub.txt for values.
-
-Example:
-
-i2s@70080300 {
-	compatible = "nvidia,tegra30-i2s";
-	reg = <0x70080300 0x100>;
-	nvidia,ahub-cif-ids = <4 4>;
-	clocks = <&tegra_car 11>;
-	resets = <&tegra_car 11>;
-	reset-names = "i2s";
-};
diff --git a/Bindings/sound/nvidia,tegra30-i2s.yaml b/Bindings/sound/nvidia,tegra30-i2s.yaml
new file mode 100644
index 0000000..89c3c64
--- /dev/null
+++ b/Bindings/sound/nvidia,tegra30-i2s.yaml
@@ -0,0 +1,67 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/nvidia,tegra30-i2s.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: NVIDIA Tegra30 I2S controller
+
+maintainers:
+  - Thierry Reding <treding@nvidia.com>
+  - Jon Hunter <jonathanh@nvidia.com>
+
+properties:
+  compatible:
+    oneOf:
+      - enum:
+          - nvidia,tegra124-i2s
+          - nvidia,tegra30-i2s
+      - items:
+          - const: nvidia,tegra114-i2s
+          - const: nvidia,tegra30-i2s
+
+  reg:
+    maxItems: 1
+
+  clocks:
+    maxItems: 1
+
+  clock-names:
+    const: i2s
+
+  resets:
+    maxItems: 1
+
+  reset-names:
+    const: i2s
+
+  nvidia,ahub-cif-ids:
+    description: list of AHUB CIF IDs
+    $ref: /schemas/types.yaml#/definitions/uint32-array
+    items:
+      - description: rx (playback)
+      - description: tx (capture)
+
+required:
+  - compatible
+  - reg
+  - clocks
+  - resets
+  - reset-names
+  - nvidia,ahub-cif-ids
+
+additionalProperties: false
+
+examples:
+  - |
+    #include <dt-bindings/clock/tegra30-car.h>
+
+    i2s@70080300 {
+        compatible = "nvidia,tegra30-i2s";
+        reg = <0x70080300 0x100>;
+        nvidia,ahub-cif-ids = <4 4>;
+        clocks = <&tegra_car TEGRA30_CLK_I2S0>;
+        resets = <&tegra_car 30>;
+        reset-names = "i2s";
+    };
+...
diff --git a/Bindings/sound/qcom,sm8250.yaml b/Bindings/sound/qcom,sm8250.yaml
index 2ab6871..b2e15eb 100644
--- a/Bindings/sound/qcom,sm8250.yaml
+++ b/Bindings/sound/qcom,sm8250.yaml
@@ -29,6 +29,8 @@
       - enum:
           - qcom,apq8016-sbc-sndcard
           - qcom,msm8916-qdsp6-sndcard
+          - qcom,qcm6490-idp-sndcard
+          - qcom,qcs6490-rb3gen2-sndcard
           - qcom,qrb5165-rb5-sndcard
           - qcom,sc7180-qdsp6-sndcard
           - qcom,sc8280xp-sndcard
diff --git a/Bindings/sound/renesas,rsnd.yaml b/Bindings/sound/renesas,rsnd.yaml
index 0d7a6b5..07ec624 100644
--- a/Bindings/sound/renesas,rsnd.yaml
+++ b/Bindings/sound/renesas,rsnd.yaml
@@ -48,13 +48,16 @@
           - const: renesas,rcar_sound-gen3
       # for Gen4 SoC
       - items:
-          - const: renesas,rcar_sound-r8a779g0  # R-Car V4H
+          - enum:
+              - renesas,rcar_sound-r8a779g0  # R-Car V4H
+              - renesas,rcar_sound-r8a779h0  # R-Car V4M
           - const: renesas,rcar_sound-gen4
       # for Generic
       - enum:
           - renesas,rcar_sound-gen1
           - renesas,rcar_sound-gen2
           - renesas,rcar_sound-gen3
+          - renesas,rcar_sound-gen4
 
   reg:
     minItems: 1
diff --git a/Bindings/sound/rockchip,rk3308-codec.yaml b/Bindings/sound/rockchip,rk3308-codec.yaml
new file mode 100644
index 0000000..ecf3d7d
--- /dev/null
+++ b/Bindings/sound/rockchip,rk3308-codec.yaml
@@ -0,0 +1,98 @@
+# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/rockchip,rk3308-codec.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Rockchip RK3308 Internal Codec
+
+description: |
+  This is the audio codec embedded in the Rockchip RK3308
+  SoC. It has 8 24-bit ADCs and 2 24-bit DACs. The maximum supported
+  sampling rate is 192 kHz.
+
+  It is connected internally to one out of a selection of the internal I2S
+  controllers.
+
+  The RK3308 audio codec has 8 independent capture channels, but some
+  features work on stereo pairs called groups:
+    * grp 0 -- MIC1 / MIC2
+    * grp 1 -- MIC3 / MIC4
+    * grp 2 -- MIC5 / MIC6
+    * grp 3 -- MIC7 / MIC8
+
+maintainers:
+  - Luca Ceresoli <luca.ceresoli@bootlin.com>
+
+properties:
+  compatible:
+    const: rockchip,rk3308-codec
+
+  reg:
+    maxItems: 1
+
+  rockchip,grf:
+    $ref: /schemas/types.yaml#/definitions/phandle
+    description:
+      Phandle to the General Register Files (GRF)
+
+  clocks:
+    items:
+      - description: clock for TX
+      - description: clock for RX
+      - description: AHB clock driving the interface
+
+  clock-names:
+    items:
+      - const: mclk_tx
+      - const: mclk_rx
+      - const: hclk
+
+  resets:
+    maxItems: 1
+
+  reset-names:
+    items:
+      - const: codec
+
+  "#sound-dai-cells":
+    const: 0
+
+  rockchip,micbias-avdd-percent:
+    description: |
+      Voltage setting for the MICBIAS pins expressed as a percentage of
+      AVDD.
+
+      E.g. if rockchip,micbias-avdd-percent = 85 and AVDD = 3v3, then the
+      MIC BIAS voltage will be 3.3 V * 85% = 2.805 V.
+
+    enum: [ 50, 55, 60, 65, 70, 75, 80, 85 ]
+
+required:
+  - compatible
+  - reg
+  - rockchip,grf
+  - clocks
+  - resets
+  - "#sound-dai-cells"
+
+additionalProperties: false
+
+examples:
+  - |
+    #include <dt-bindings/clock/rk3308-cru.h>
+
+    audio_codec: audio-codec@ff560000 {
+        compatible = "rockchip,rk3308-codec";
+        reg = <0xff560000 0x10000>;
+        rockchip,grf = <&grf>;
+        clock-names = "mclk_tx", "mclk_rx", "hclk";
+        clocks = <&cru SCLK_I2S2_8CH_TX_OUT>,
+                 <&cru SCLK_I2S2_8CH_RX_OUT>,
+                 <&cru PCLK_ACODEC>;
+        reset-names = "codec";
+        resets = <&cru SRST_ACODEC_P>;
+        #sound-dai-cells = <0>;
+    };
+
+...
diff --git a/Bindings/sound/st,stm32-i2s.yaml b/Bindings/sound/st,stm32-i2s.yaml
index b9111d3..8978f6b 100644
--- a/Bindings/sound/st,stm32-i2s.yaml
+++ b/Bindings/sound/st,stm32-i2s.yaml
@@ -65,6 +65,10 @@
     $ref: audio-graph-port.yaml#
     unevaluatedProperties: false
 
+  access-controllers:
+    minItems: 1
+    maxItems: 2
+
 required:
   - compatible
   - "#sound-dai-cells"
diff --git a/Bindings/sound/st,stm32-sai.yaml b/Bindings/sound/st,stm32-sai.yaml
index 59df8a8..68f97b4 100644
--- a/Bindings/sound/st,stm32-sai.yaml
+++ b/Bindings/sound/st,stm32-sai.yaml
@@ -48,6 +48,10 @@
   clock-names:
     maxItems: 3
 
+  access-controllers:
+    minItems: 1
+    maxItems: 2
+
 required:
   - compatible
   - reg
@@ -68,7 +72,7 @@
     properties:
       compatible:
         description: Compatible for SAI sub-block A or B.
-        pattern: "st,stm32-sai-sub-[ab]"
+        pattern: "^st,stm32-sai-sub-[ab]$"
 
       "#sound-dai-cells":
         const: 0
diff --git a/Bindings/sound/st,stm32-spdifrx.yaml b/Bindings/sound/st,stm32-spdifrx.yaml
index bc48151..3dedc81 100644
--- a/Bindings/sound/st,stm32-spdifrx.yaml
+++ b/Bindings/sound/st,stm32-spdifrx.yaml
@@ -50,6 +50,10 @@
   resets:
     maxItems: 1
 
+  access-controllers:
+    minItems: 1
+    maxItems: 2
+
 required:
   - compatible
   - "#sound-dai-cells"
diff --git a/Bindings/sound/ti,pcm1681.txt b/Bindings/sound/ti,pcm1681.txt
deleted file mode 100644
index 4df1718..0000000
--- a/Bindings/sound/ti,pcm1681.txt
+++ /dev/null
@@ -1,15 +0,0 @@
-Texas Instruments PCM1681 8-channel PWM Processor
-
-Required properties:
-
- - compatible:		Should contain "ti,pcm1681".
- - reg:			The i2c address. Should contain <0x4c>.
-
-Examples:
-
-	i2c_bus {
-		pcm1681@4c {
-			compatible = "ti,pcm1681";
-			reg = <0x4c>;
-		};
-	};
diff --git a/Bindings/sound/ti,pcm1681.yaml b/Bindings/sound/ti,pcm1681.yaml
new file mode 100644
index 0000000..5aa0061
--- /dev/null
+++ b/Bindings/sound/ti,pcm1681.yaml
@@ -0,0 +1,43 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/ti,pcm1681.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Texas Instruments PCM1681 8-channel PWM Processor
+
+maintainers:
+  - Shenghao Ding <shenghao-ding@ti.com>
+  - Kevin Lu <kevin-lu@ti.com>
+  - Baojun Xu <baojun.xu@ti.com>
+
+allOf:
+  - $ref: dai-common.yaml#
+
+properties:
+  compatible:
+    const: ti,pcm1681
+
+  reg:
+    maxItems: 1
+
+  "#sound-dai-cells":
+    const: 0
+
+required:
+  - compatible
+  - reg
+
+unevaluatedProperties: false
+
+examples:
+  - |
+    i2c {
+        #address-cells = <1>;
+        #size-cells = <0>;
+
+        pcm1681: audio-codec@4c {
+            compatible = "ti,pcm1681";
+            reg = <0x4c>;
+        };
+    };
diff --git a/Bindings/sound/ti,pcm6240.yaml b/Bindings/sound/ti,pcm6240.yaml
new file mode 100644
index 0000000..dd5b08e
--- /dev/null
+++ b/Bindings/sound/ti,pcm6240.yaml
@@ -0,0 +1,177 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+# Copyright (C) 2022 - 2024 Texas Instruments Incorporated
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/ti,pcm6240.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Texas Instruments PCM6240 Family Audio ADC/DAC
+
+maintainers:
+  - Shenghao Ding <shenghao-ding@ti.com>
+
+description: |
+  The PCM6240 Family is a big family of Audio ADC/DAC for
+  different Specifications, range from Personal Electric
+  to Automotive Electric, even some professional fields.
+
+  Specifications about the audio chip can be found at:
+    https://www.ti.com/lit/gpn/tlv320adc3120
+    https://www.ti.com/lit/gpn/tlv320adc5120
+    https://www.ti.com/lit/gpn/tlv320adc6120
+    https://www.ti.com/lit/gpn/dix4192
+    https://www.ti.com/lit/gpn/pcm1690
+    https://www.ti.com/lit/gpn/pcm3120-q1
+    https://www.ti.com/lit/gpn/pcm3140-q1
+    https://www.ti.com/lit/gpn/pcm5120-q1
+    https://www.ti.com/lit/gpn/pcm6120-q1
+    https://www.ti.com/lit/gpn/pcm6260-q1
+    https://www.ti.com/lit/gpn/pcm9211
+    https://www.ti.com/lit/gpn/pcmd3140
+    https://www.ti.com/lit/gpn/pcmd3180
+    https://www.ti.com/lit/gpn/taa5212
+    https://www.ti.com/lit/gpn/tad5212
+
+properties:
+  compatible:
+    description: |
+      ti,adc3120: Stereo-channel, 768-kHz, Burr-Brown™ audio analog-to-
+      digital converter (ADC) with 106-dB SNR.
+
+      ti,adc5120: 2-Channel, 768-kHz, Burr-Brown™ Audio ADC with 120-dB SNR.
+
+      ti,adc6120: Stereo-channel, 768-kHz, Burr-Brown™ audio analog-to-
+      digital converter (ADC) with 123-dB SNR.
+
+      ti,dix4192: 216-kHz digital audio converter with Quad-Channel In
+      and One-Channel Out.
+
+      ti,pcm1690: Automotive Catalog 113dB SNR 8-Channel Audio DAC with
+      Differential Outputs.
+
+      ti,pcm3120: Automotive, stereo, 106-dB SNR, 768-kHz, low-power
+      software-controlled audio ADC.
+
+      ti,pcm3140: Automotive, Quad-Channel, 768-kHz, Burr-Brown™ Audio ADC
+      with 106-dB SNR.
+
+      ti,pcm5120: Automotive, stereo, 120-dB SNR, 768-kHz, low-power
+      software-controlled audio ADC.
+
+      ti,pcm5140: Automotive, Quad-Channel, 768-kHz, Burr-Brown™ Audio ADC
+      with 120-dB SNR.
+
+      ti,pcm6120: Automotive, stereo, 123-dB SNR, 768-kHz, low-power
+      software-controlled audio ADC.
+
+      ti,pcm6140: Automotive, Quad-Channel, 768-kHz, Burr-Brown™ Audio ADC
+      with 123-dB SNR.
+
+      ti,pcm6240: Automotive 4-ch audio ADC with integrated programmable mic
+      bias, boost and input diagnostics.
+
+      ti,pcm6260: Automotive 6-ch audio ADC with integrated programmable mic
+      bias, boost and input diagnostics.
+
+      ti,pcm9211: 216-kHz digital audio converter With Stereo ADC and
+      Routing.
+
+      ti,pcmd3140: Four-channel PDM-input to TDM or I2S output converter.
+
+      ti,pcmd3180: Eight-channel pulse-density-modulation input to TDM or
+      I2S output converter.
+
+      ti,taa5212: Low-power high-performance stereo audio ADC with 118-dB
+      dynamic range.
+
+      ti,tad5212: Low-power stereo audio DAC with 120-dB dynamic range.
+    oneOf:
+      - items:
+          - enum:
+              - ti,adc3120
+              - ti,adc5120
+              - ti,pcm3120
+              - ti,pcm5120
+              - ti,pcm6120
+          - const: ti,adc6120
+      - items:
+          - enum:
+              - ti,pcmd512x
+              - ti,pcm9211
+              - ti,taa5212
+              - ti,tad5212
+          - const: ti,adc6120
+      - items:
+          - enum:
+              - ti,pcm3140
+              - ti,pcm5140
+              - ti,dix4192
+              - ti,pcm6140
+              - ti,pcm6260
+          - const: ti,pcm6240
+      - items:
+          - enum:
+              - ti,pcmd3140
+              - ti,pcmd3180
+              - ti,pcm1690
+              - ti,taa5412
+              - ti,tad5412
+          - const: ti,pcm6240
+      - enum:
+          - ti,adc6120
+          - ti,pcm6240
+
+  reg:
+    description:
+      I2C address, in multiple pcmdevices case, all the i2c address
+      aggregate as one Audio Device to support multiple audio slots.
+    minItems: 1
+    maxItems: 4
+
+  reset-gpios:
+    maxItems: 1
+
+  interrupts:
+    maxItems: 1
+    description:
+      Invalid only for ti,pcm1690 because of no INT pin.
+
+  '#sound-dai-cells':
+    const: 0
+
+required:
+  - compatible
+  - reg
+
+allOf:
+  - $ref: dai-common.yaml#
+  - if:
+      properties:
+        compatible:
+          contains:
+            enum:
+              - ti,pcm1690
+    then:
+      properties:
+        interrupts: false
+
+additionalProperties: false
+
+examples:
+  - |
+   #include <dt-bindings/gpio/gpio.h>
+   i2c {
+     /* example for two devices with interrupt support */
+     #address-cells = <1>;
+     #size-cells = <0>;
+     pcm6240: audio-codec@48 {
+       compatible = "ti,pcm6240";
+       reg = <0x48>, /* primary-device */
+             <0x4b>; /* secondary-device */
+       #sound-dai-cells = <0>;
+       reset-gpios = <&gpio1 10 GPIO_ACTIVE_HIGH>;
+       interrupt-parent = <&gpio1>;
+       interrupts = <15>;
+     };
+   };
+...
diff --git a/Bindings/sound/wlf,wm8776.yaml b/Bindings/sound/wlf,wm8776.yaml
new file mode 100644
index 0000000..7bbc96e
--- /dev/null
+++ b/Bindings/sound/wlf,wm8776.yaml
@@ -0,0 +1,41 @@
+# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/wlf,wm8776.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: WM8776 audio CODEC
+
+maintainers:
+  - patches@opensource.cirrus.com
+
+allOf:
+  - $ref: dai-common.yaml#
+
+properties:
+  compatible:
+    const: wlf,wm8776
+
+  reg:
+    maxItems: 1
+
+  "#sound-dai-cells":
+    const: 0
+
+required:
+  - compatible
+  - reg
+
+unevaluatedProperties: false
+
+examples:
+  - |
+    i2c {
+        #address-cells = <1>;
+        #size-cells = <0>;
+
+        codec@1a {
+            compatible = "wlf,wm8776";
+            reg = <0x1a>;
+        };
+    };
diff --git a/Bindings/sound/wlf,wm8974.txt b/Bindings/sound/wlf,wm8974.txt
deleted file mode 100644
index 01d3a7c..0000000
--- a/Bindings/sound/wlf,wm8974.txt
+++ /dev/null
@@ -1,15 +0,0 @@
-WM8974 audio CODEC
-
-This device supports both I2C and SPI (configured with pin strapping
-on the board).
-
-Required properties:
-  - compatible: "wlf,wm8974"
-  - reg: the I2C address or SPI chip select number of the device
-
-Examples:
-
-codec: wm8974@1a {
-	compatible = "wlf,wm8974";
-	reg = <0x1a>;
-};
diff --git a/Bindings/sound/wlf,wm8974.yaml b/Bindings/sound/wlf,wm8974.yaml
new file mode 100644
index 0000000..d273002
--- /dev/null
+++ b/Bindings/sound/wlf,wm8974.yaml
@@ -0,0 +1,41 @@
+# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/wlf,wm8974.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: WM8974 audio CODEC
+
+maintainers:
+  - patches@opensource.cirrus.com
+
+allOf:
+  - $ref: dai-common.yaml#
+
+properties:
+  compatible:
+    const: wlf,wm8974
+
+  reg:
+    maxItems: 1
+
+  "#sound-dai-cells":
+    const: 0
+
+required:
+  - compatible
+  - reg
+
+unevaluatedProperties: false
+
+examples:
+  - |
+    i2c {
+        #address-cells = <1>;
+        #size-cells = <0>;
+
+        codec@1a {
+            compatible = "wlf,wm8974";
+            reg = <0x1a>;
+        };
+    };
diff --git a/Bindings/sound/wm8776.txt b/Bindings/sound/wm8776.txt
deleted file mode 100644
index 0117336..0000000
--- a/Bindings/sound/wm8776.txt
+++ /dev/null
@@ -1,18 +0,0 @@
-WM8776 audio CODEC
-
-This device supports both I2C and SPI (configured with pin strapping
-on the board).
-
-Required properties:
-
-  - compatible : "wlf,wm8776"
-
-  - reg : the I2C address of the device for I2C, the chip select
-          number for SPI.
-
-Example:
-
-wm8776: codec@1a {
-	compatible = "wlf,wm8776";
-	reg = <0x1a>;
-};
diff --git a/Bindings/sound/xmos,xvf3500.yaml b/Bindings/sound/xmos,xvf3500.yaml
new file mode 100644
index 0000000..fb77a61
--- /dev/null
+++ b/Bindings/sound/xmos,xvf3500.yaml
@@ -0,0 +1,63 @@
+# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/xmos,xvf3500.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: XMOS XVF3500 VocalFusion Voice Processor
+
+maintainers:
+  - Javier Carrasco <javier.carrasco@wolfvision.net>
+
+description:
+  The XMOS XVF3500 VocalFusion Voice Processor is a low-latency, 32-bit
+  multicore controller for voice processing.
+  https://www.xmos.com/xvf3500/
+
+allOf:
+  - $ref: /schemas/usb/usb-device.yaml#
+
+properties:
+  compatible:
+    const: usb20b1,0013
+
+  reg: true
+
+  reset-gpios:
+    maxItems: 1
+
+  vdd-supply:
+    description:
+      Regulator for the 1V0 supply.
+
+  vddio-supply:
+    description:
+      Regulator for the 3V3 supply.
+
+required:
+  - compatible
+  - reg
+  - reset-gpios
+  - vdd-supply
+  - vddio-supply
+
+additionalProperties: false
+
+examples:
+  - |
+    #include <dt-bindings/gpio/gpio.h>
+
+    usb {
+        #address-cells = <1>;
+        #size-cells = <0>;
+
+        voice_processor: voice-processor@1 {
+            compatible = "usb20b1,0013";
+            reg = <1>;
+            reset-gpios = <&gpio 5 GPIO_ACTIVE_LOW>;
+            vdd-supply = <&vcc1v0>;
+            vddio-supply = <&vcc3v3>;
+        };
+    };
+
+...