ASoC: Updates for v5.18

Quite a quiet release for ASoC, lots of work on drivers and platforms
 but nothing too groundbreaking but not much on the core itself:
 
  - Start of moving SoF to support multiple IPC mechanisms.
  - Use of NHLT ACPI table to reduce the amount of quirking required for
    Intel systems.
  - Some building blocks for use in forthcoming Intel AVS driver for
    legacy Intel DSP firmwares.
  - Support for AMD PDM, Atmel PDMC, Awinic AW8738, i.MX cards with
    TLV320AIC31xx, Intel machines with CS35L41 and ESSX8336, Mediatek
    MT8181 wideband bluetooth, nVidia Tegra234, Qualcomm SC7280, Renesas
    RZ/V2L, Texas Instruments TAS585M
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Merge tag 'asoc-v5.18' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: Updates for v5.18

Quite a quiet release for ASoC, lots of work on drivers and platforms
but nothing too groundbreaking but not much on the core itself:

 - Start of moving SoF to support multiple IPC mechanisms.
 - Use of NHLT ACPI table to reduce the amount of quirking required for
   Intel systems.
 - Some building blocks for use in forthcoming Intel AVS driver for
   legacy Intel DSP firmwares.
 - Support for AMD PDM, Atmel PDMC, Awinic AW8738, i.MX cards with
   TLV320AIC31xx, Intel machines with CS35L41 and ESSX8336, Mediatek
   MT8181 wideband bluetooth, nVidia Tegra234, Qualcomm SC7280, Renesas
   RZ/V2L, Texas Instruments TAS585M

Documentation/devicetree/bindings/bus/nvidia,tegra210-aconnect.yaml

@@ -21,6 +21,7 @@ properties:
       - const: nvidia,tegra210-aconnect
       - items:
           - enum:
+              - nvidia,tegra234-aconnect
               - nvidia,tegra186-aconnect
               - nvidia,tegra194-aconnect
           - const: nvidia,tegra210-aconnect

Documentation/devicetree/bindings/dma/nvidia,tegra210-adma.yaml

@@ -23,7 +23,9 @@ properties:
           - nvidia,tegra210-adma
           - nvidia,tegra186-adma
       - items:
-          - const: nvidia,tegra194-adma
+          - enum:
+              - nvidia,tegra234-adma
+              - nvidia,tegra194-adma
           - const: nvidia,tegra186-adma
 
   reg:

Documentation/devicetree/bindings/interrupt-controller/arm,gic.yaml

@@ -58,6 +58,7 @@ properties:
               - enum:
                   - nvidia,tegra186-agic
                   - nvidia,tegra194-agic
+                  - nvidia,tegra234-agic
               - const: nvidia,tegra210-agic
 
   interrupt-controller: true

Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-i2s.yaml

@@ -31,6 +31,10 @@ properties:
           - const: allwinner,sun50i-a64-i2s
           - const: allwinner,sun8i-h3-i2s
       - const: allwinner,sun50i-h6-i2s
+      - const: allwinner,sun50i-r329-i2s
+      - items:
+          - const: allwinner,sun20i-d1-i2s
+          - const: allwinner,sun50i-r329-i2s
 
   reg:
     maxItems: 1
@@ -67,6 +71,7 @@ allOf:
               - allwinner,sun8i-h3-i2s
               - allwinner,sun50i-a64-codec-i2s
               - allwinner,sun50i-h6-i2s
+              - allwinner,sun50i-r329-i2s
 
     then:
       required:

Documentation/devicetree/bindings/sound/arndale.txt

@@ -1,25 +0,0 @@
-Audio Binding for Arndale boards
-
-Required properties:
-- compatible : Can be one of the following:
-		"samsung,arndale-rt5631",
-		"samsung,arndale-wm1811"
-
-- samsung,audio-cpu: The phandle of the Samsung I2S controller
-- samsung,audio-codec: The phandle of the audio codec
-
-Optional:
-- samsung,model: The name of the sound-card
-
-Arndale Boards has many audio daughter cards, one of them is
-rt5631/alc5631. Below example shows audio bindings for rt5631/
-alc5631 based codec.
-
-Example:
-
-sound {
-		compatible = "samsung,arndale-rt5631";
-
-		samsung,audio-cpu = <&i2s0>
-		samsung,audio-codec = <&rt5631>;
-};

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

@@ -71,4 +71,24 @@ patternProperties:
         description: CPU to Codec rate channels.
         $ref: /schemas/types.yaml#/definitions/uint32
 
+      dai-tdm-slot-width-map:
+        description: Mapping of sample widths to slot widths. For hardware
+          that cannot support a fixed slot width or a slot width always
+          equal to sample width. A matrix of one or more 3-tuples.
+        $ref: /schemas/types.yaml#/definitions/uint32-matrix
+        items:
+          items:
+            -
+              description: Sample width in bits
+              minimum: 8
+              maximum: 64
+            -
+              description: Slot width in bits
+              minimum: 8
+              maximum: 256
+            -
+              description: Slot count
+              minimum: 1
+              maximum: 64
+
 additionalProperties: true

Documentation/devicetree/bindings/sound/awinic,aw8738.yaml

@@ -0,0 +1,54 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/awinic,aw8738.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Awinic AW8738 Audio Amplifier
+
+maintainers:
+  - Stephan Gerhold <stephan@gerhold.net>
+
+description:
+  The Awinic AW8738 is a simple audio amplifier with different operation modes
+  (set using one-wire pulse control). The mode configures the speaker-guard
+  function (primarily the power limit for the amplifier).
+
+allOf:
+  - $ref: name-prefix.yaml#
+
+properties:
+  compatible:
+    const: awinic,aw8738
+
+  mode-gpios:
+    description:
+      GPIO used for one-wire pulse control. The pin is typically called SHDN
+      (active-low), but this is misleading since it is actually more than
+      just a simple shutdown/enable control.
+    maxItems: 1
+
+  awinic,mode:
+    description: Operation mode (number of pulses for one-wire pulse control)
+    $ref: /schemas/types.yaml#/definitions/uint32
+    minimum: 1
+    maximum: 7
+
+  sound-name-prefix: true
+
+required:
+  - compatible
+  - mode-gpios
+  - awinic,mode
+
+additionalProperties: false
+
+examples:
+  - |
+    #include <dt-bindings/gpio/gpio.h>
+    audio-amplifier {
+        compatible = "awinic,aw8738";
+        mode-gpios = <&msmgpio 114 GPIO_ACTIVE_HIGH>;
+        awinic,mode = <5>;
+        sound-name-prefix = "Speaker Amp";
+    };

Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt

@@ -1,86 +0,0 @@
-Texas Instruments McASP controller
-
-Required properties:
-- compatible :
-	"ti,dm646x-mcasp-audio"	: for DM646x platforms
-	"ti,da830-mcasp-audio"	: for both DA830 & DA850 platforms
-	"ti,am33xx-mcasp-audio"	: for AM33xx platforms (AM33xx, AM43xx, TI81xx)
-	"ti,dra7-mcasp-audio"	: for DRA7xx platforms
-	"ti,omap4-mcasp-audio"	: for OMAP4
-
-- reg : Should contain reg specifiers for the entries in the reg-names property.
-- reg-names : Should contain:
-         * "mpu" for the main registers (required). For compatibility with
-           existing software, it is recommended this is the first entry.
-         * "dat" for separate data port register access (optional).
-- op-mode : I2S/DIT ops mode. 0 for I2S mode. 1 for DIT mode used for S/PDIF,
-  	    IEC60958-1, and AES-3 formats.
-- tdm-slots : Slots for TDM operation. Indicates number of channels transmitted
-  	      or received over one serializer.
-- serial-dir : A list of serializer configuration. Each entry is a number
-               indication for serializer pin direction.
-               (0 - INACTIVE, 1 - TX, 2 - RX)
-- dmas: two element list of DMA controller phandles and DMA request line
-        ordered pairs.
-- 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:
-
-- ti,hwmods : Must be "mcasp<n>", n is controller instance starting 0
-- tx-num-evt : FIFO levels.
-- rx-num-evt : FIFO levels.
-- dismod : Specify the drive on TX pin during inactive slots
-	0 : 3-state
-	2 : logic low
-	3 : logic high
-	Defaults to 'logic low' when the property is not present
-- sram-size-playback : size of sram to be allocated during playback
-- sram-size-capture  : size of sram to be allocated during capture
-- interrupts : Interrupt numbers for McASP
-- interrupt-names : Known interrupt names are "tx" and "rx"
-- 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
-- fck_parent : Should contain a valid clock name which will be used as parent
-	       for the McASP fck
-- auxclk-fs-ratio: When McASP is bus master indicates the ratio between AUCLK
-		   and FS rate if applicable:
-		   AUCLK rate = auxclk-fs-ratio * FS rate
-
-Optional GPIO support:
-If any McASP pin need to be used as GPIO then the McASP node must have:
-...
-  gpio-controller
-  #gpio-cells = <2>;
-...
-
-When requesting a GPIO, the first parameter is the PIN index in McASP_P*
-registers.
-For example to request the AXR2 pin of mcasp8:
-function-gpios = <&mcasp8 2 0>;
-
-Or to request the ACLKR pin of mcasp8:
-function-gpios = <&mcasp8 29 0>;
-
-For generic gpio information, please refer to bindings/gpio/gpio.txt
-
-Example:
-
-mcasp0: mcasp0@1d00000 {
-	compatible = "ti,da830-mcasp-audio";
-	reg = <0x100000 0x3000>;
-	reg-names "mpu";
-	interrupts = <82>, <83>;
-	interrupt-names = "tx", "rx";
-	op-mode = <0>;		/* MCASP_IIS_MODE */
-	tdm-slots = <2>;
-	serial-dir = <
-			0 0 0 0	/* 0: INACTIVE, 1: TX, 2: RX */
-			0 0 0 0
-			0 0 0 1
-			2 0 0 0 >;
-	tx-num-evt = <1>;
-	rx-num-evt = <1>;
-};

Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml

@@ -0,0 +1,201 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/davinci-mcasp-audio.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: McASP Controller for TI SoCs
+
+maintainers:
+  - Jayesh Choudhary <j-choudhary@ti.com>
+
+properties:
+  compatible:
+    enum:
+      - ti,dm646x-mcasp-audio
+      - ti,da830-mcasp-audio
+      - ti,am33xx-mcasp-audio
+      - ti,dra7-mcasp-audio
+      - ti,omap4-mcasp-audio
+
+  reg:
+    minItems: 1
+    items:
+      - description: CFG registers
+      - description: data registers
+
+  reg-names:
+    minItems: 1
+    items:
+      - const: mpu
+      - const: dat
+
+  op-mode:
+    $ref: /schemas/types.yaml#/definitions/uint32
+    description: 0 - I2S or 1 - DIT operation mode
+    enum:
+      - 0
+      - 1
+
+  tdm-slots:
+    $ref: /schemas/types.yaml#/definitions/uint32
+    description:
+      number of channels over one serializer
+      the property is ignored in DIT mode
+    minimum: 2
+    maximum: 32
+
+  serial-dir:
+    description:
+      A list of serializer configuration
+      Entry is indication for serializer pin direction
+      0 - Inactive, 1 - TX, 2 - RX
+      All AXR pins should be present in the array even if inactive
+    $ref: /schemas/types.yaml#/definitions/uint32-array
+    minItems: 1
+    maxItems: 25
+    items:
+      minimum: 0
+      maximum: 2
+
+  dmas:
+    minItems: 1
+    items:
+      - description: transmission DMA channel
+      - description: reception DMA channel
+
+  dma-names:
+    minItems: 1
+    items:
+      - const: tx
+      - const: rx
+
+  ti,hwmods:
+    $ref: /schemas/types.yaml#/definitions/string
+    description: Name of hwmod associated with McASP
+    maxItems: 1
+    deprecated: true
+
+  tx-num-evt:
+    $ref: /schemas/types.yaml#/definitions/uint32
+    description:
+      configures WFIFO threshold
+      0 disables the FIFO use
+      if property is missing, then also FIFO use is disabled
+
+  rx-num-evt:
+    $ref: /schemas/types.yaml#/definitions/uint32
+    description:
+      configures RFIFO threshold
+      0 disables the FIFO use
+      if property is missing, then also FIFO use is disabled
+
+  dismod:
+    $ref: /schemas/types.yaml#/definitions/uint32
+    description:
+      specify the drive on TX pin during inactive time slots
+      0 - 3-state, 2 - logic low, 3 - logic high
+    enum:
+      - 0
+      - 2
+      - 3
+    default: 2
+
+  interrupts:
+    anyOf:
+      - minItems: 1
+        items:
+          - description: TX interrupt
+          - description: RX interrupt
+      - items:
+          - description: common/combined interrupt
+
+  interrupt-names:
+    oneOf:
+      - minItems: 1
+        items:
+          - const: tx
+          - const: rx
+      - const: common
+
+  fck_parent:
+    $ref: /schemas/types.yaml#/definitions/string
+    description: parent clock name for McASP fck
+    maxItems: 1
+
+  auxclk-fs-ratio:
+    $ref: /schemas/types.yaml#/definitions/uint32
+    description: ratio of AUCLK and FS rate if applicable
+
+  gpio-controller: true
+
+  "#gpio-cells":
+    const: 2
+
+  clocks:
+    minItems: 1
+    items:
+      - description: functional clock
+      - description: module specific optional ahclkx clock
+      - description: module specific optional ahclkr clock
+
+  clock-names:
+    minItems: 1
+    items:
+      - const: fck
+      - const: ahclkx
+      - const: ahclkr
+
+  power-domains:
+    description: phandle to the corresponding power-domain
+    maxItems: 1
+
+  "#sound-dai-cells":
+    const: 0
+
+  port:
+    description: connection for when McASP is used via graph card
+    type: object
+
+required:
+  - compatible
+  - reg
+  - reg-names
+  - dmas
+  - dma-names
+  - interrupts
+  - interrupt-names
+
+allOf:
+  - if:
+      properties:
+        opmode:
+          enum:
+            - 0
+
+    then:
+      required:
+        - tdm-slots
+
+additionalProperties: false
+
+examples:
+  - |
+    mcasp0: mcasp0@1d00000 {
+      compatible = "ti,da830-mcasp-audio";
+      reg = <0x100000 0x3000>;
+      reg-names = "mpu";
+      interrupts = <82>, <83>;
+      interrupt-names = "tx", "rx";
+      op-mode = <0>;		/* MCASP_IIS_MODE */
+      tdm-slots = <2>;
+      dmas = <&main_udmap 0xc400>, <&main_udmap 0x4400>;
+      dma-names = "tx", "rx";
+      serial-dir = <
+          0 0 0 0	/* 0: INACTIVE, 1: TX, 2: RX */
+          0 0 0 0
+          0 0 0 1
+          2 0 0 0 >;
+      tx-num-evt = <1>;
+      rx-num-evt = <1>;
+    };

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

@@ -40,6 +40,8 @@ The compatible list for this generic sound card currently:
 
  "fsl,imx-audio-tlv320aic32x4"
 
+ "fsl,imx-audio-tlv320aic31xx"
+
  "fsl,imx-audio-si476x"
 
  "fsl,imx-audio-wm8958"
@@ -82,6 +84,7 @@ Optional properties:
   - 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:
 

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

@@ -0,0 +1,180 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/google,sc7280-herobrine.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Google SC7280-Herobrine ASoC sound card driver
+
+maintainers:
+  - Srinivasa Rao Mandadapu <srivasam@codeaurora.org>
+  - Judy Hsiao <judyhsiao@chromium.org>
+
+description:
+  This binding describes the SC7280 sound card which uses LPASS for audio.
+
+properties:
+  compatible:
+    enum:
+      - google,sc7280-herobrine
+
+  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.
+
+  model:
+    $ref: /schemas/types.yaml#/definitions/string
+    description: User specified audio sound card name
+
+  "#address-cells":
+    const: 1
+
+  "#size-cells":
+    const: 0
+
+patternProperties:
+  "^dai-link@[0-9a-f]$":
+    description:
+      Each subnode represents a dai link. Subnodes of each dai links would be
+      cpu/codec dais.
+
+    type: object
+
+    properties:
+      link-name:
+        description: Indicates dai-link name and PCM stream name.
+        $ref: /schemas/types.yaml#/definitions/string
+        maxItems: 1
+
+      reg:
+        maxItems: 1
+        description: dai link address.
+
+      cpu:
+        description: Holds subnode which indicates cpu dai.
+        type: object
+        properties:
+          sound-dai: true
+
+        required:
+          - sound-dai
+
+        additionalProperties: false
+
+      codec:
+        description: Holds subnode which indicates codec dai.
+        type: object
+        properties:
+          sound-dai: true
+
+        required:
+          - sound-dai
+
+        additionalProperties: false
+
+    required:
+      - link-name
+      - cpu
+      - codec
+      - reg
+
+    additionalProperties: false
+
+required:
+  - compatible
+  - model
+  - "#address-cells"
+  - "#size-cells"
+
+additionalProperties: false
+
+examples:
+
+  - |
+    #include <dt-bindings/sound/qcom,lpass.h>
+    sound {
+        compatible = "google,sc7280-herobrine";
+        model = "sc7280-wcd938x-max98360a-4dmic";
+
+        audio-routing =
+            "IN1_HPHL", "HPHL_OUT",
+            "IN2_HPHR", "HPHR_OUT",
+            "AMIC1", "MIC BIAS1",
+            "AMIC2", "MIC BIAS2",
+            "VA DMIC0", "MIC BIAS3",
+            "VA DMIC1", "MIC BIAS3",
+            "VA DMIC2", "MIC BIAS4",
+            "VA DMIC3", "MIC BIAS4",
+            "TX SWR_ADC0", "ADC1_OUTPUT",
+            "TX SWR_ADC1", "ADC2_OUTPUT",
+            "TX SWR_ADC2", "ADC3_OUTPUT",
+            "TX SWR_DMIC0", "DMIC1_OUTPUT",
+            "TX SWR_DMIC1", "DMIC2_OUTPUT",
+            "TX SWR_DMIC2", "DMIC3_OUTPUT",
+            "TX SWR_DMIC3", "DMIC4_OUTPUT";
+
+        #address-cells = <1>;
+        #size-cells = <0>;
+
+        dai-link@0 {
+            link-name = "WCD Playback";
+            reg = <LPASS_CDC_DMA_RX0>;
+            cpu {
+                sound-dai = <&lpass_cpu LPASS_CDC_DMA_RX0>;
+            };
+
+            codec {
+                sound-dai = <&wcd938x 0>, <&swr0 0>, <&rxmacro 0>;
+            };
+        };
+        dai-link@1 {
+            link-name = "WCD Capture";
+            reg = <LPASS_CDC_DMA_TX3>;
+            cpu {
+                sound-dai = <&lpass_cpu LPASS_CDC_DMA_TX3>;
+            };
+
+            codec {
+                sound-dai = <&wcd938x 1>, <&swr1 0>, <&txmacro 0>;
+            };
+        };
+
+        dai-link@2 {
+            link-name = "MI2S Playback";
+            reg = <MI2S_SECONDARY>;
+            cpu {
+                sound-dai = <&lpass_cpu MI2S_SECONDARY>;
+            };
+
+            codec {
+                sound-dai = <&max98360a>;
+            };
+        };
+
+        dai-link@3 {
+            link-name = "DMIC Capture";
+            reg = <LPASS_CDC_DMA_VA_TX0>;
+            cpu {
+                sound-dai = <&lpass_cpu LPASS_CDC_DMA_VA_TX0>;
+            };
+
+            codec {
+                sound-dai = <&vamacro 0>;
+            };
+        };
+
+        dai-link@5 {
+            link-name = "DP Playback";
+            reg = <LPASS_DP_RX>;
+            cpu {
+                sound-dai = <&lpass_cpu LPASS_DP_RX>;
+            };
+
+            codec {
+                sound-dai = <&mdss_dp>;
+            };
+        };
+    };

Documentation/devicetree/bindings/sound/microchip,pdmc.yaml

@@ -0,0 +1,100 @@
+# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/microchip,pdmc.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Microchip Pulse Density Microphone Controller
+
+maintainers:
+  - Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
+
+description:
+  The Microchip Pulse Density Microphone Controller (PDMC) interfaces up to 4
+  digital microphones having Pulse Density Modulated (PDM) outputs.
+
+properties:
+  compatible:
+    const: microchip,sama7g5-pdmc
+
+  reg:
+    maxItems: 1
+
+  "#sound-dai-cells":
+    const: 0
+
+  interrupts:
+    maxItems: 1
+
+  clocks:
+    items:
+      - description: Peripheral Bus Clock
+      - description: Generic Clock
+
+  clock-names:
+    items:
+      - const: pclk
+      - const: gclk
+
+  dmas:
+    description: RX DMA Channel
+    maxItems: 1
+
+  dma-names:
+    const: rx
+
+  microchip,mic-pos:
+    description: |
+      Position of PDM microphones on the DS line and the sampling edge (rising
+      or falling) of the CLK line. A microphone is represented as a pair of DS
+      line and the sampling edge. The first microphone is mapped to channel 0,
+      the second to channel 1, etc.
+    $ref: /schemas/types.yaml#/definitions/uint32-matrix
+    items:
+      items:
+        - description: value for DS line
+        - description: value for sampling edge
+      anyOf:
+        - enum:
+            - [0, 0]
+            - [0, 1]
+            - [1, 0]
+            - [1, 1]
+    minItems: 1
+    maxItems: 4
+    uniqueItems: true
+
+required:
+  - compatible
+  - reg
+  - "#sound-dai-cells"
+  - interrupts
+  - clocks
+  - clock-names
+  - dmas
+  - dma-names
+  - microchip,mic-pos
+
+additionalProperties: false
+
+examples:
+  - |
+    #include <dt-bindings/clock/at91.h>
+    #include <dt-bindings/dma/at91.h>
+    #include <dt-bindings/interrupt-controller/arm-gic.h>
+    #include <dt-bindings/sound/microchip,pdmc.h>
+
+    pdmc: sound@e1608000 {
+        compatible = "microchip,sama7g5-pdmc";
+        reg = <0xe1608000 0x4000>;
+        #sound-dai-cells = <0>;
+        interrupts = <GIC_SPI 68 IRQ_TYPE_LEVEL_HIGH>;
+        dmas = <&dma0 AT91_XDMAC_DT_PERID(37)>;
+        dma-names = "rx";
+        clocks = <&pmc PMC_TYPE_PERIPHERAL 68>, <&pmc PMC_TYPE_GCK 68>;
+        clock-names = "pclk", "gclk";
+        microchip,mic-pos = <MCHP_PDMC_DS0 MCHP_PDMC_CLK_POSITIVE>,
+                            <MCHP_PDMC_DS0 MCHP_PDMC_CLK_NEGATIVE>,
+                            <MCHP_PDMC_DS1 MCHP_PDMC_CLK_POSITIVE>,
+                            <MCHP_PDMC_DS1 MCHP_PDMC_CLK_NEGATIVE>;
+    };

Documentation/devicetree/bindings/sound/mt8195-afe-pcm.yaml

@@ -19,6 +19,12 @@ properties:
   interrupts:
     maxItems: 1
 
+  resets:
+    maxItems: 1
+
+  reset-names:
+    const: audiosys
+
   memory-region:
     maxItems: 1
     description: |
@@ -127,6 +133,8 @@ required:
   - compatible
   - reg
   - interrupts
+  - resets
+  - reset-names
   - mediatek,topckgen
   - power-domains
   - clocks
@@ -144,6 +152,8 @@ examples:
         compatible = "mediatek,mt8195-audio";
         reg = <0x10890000 0x10000>;
         interrupts = <GIC_SPI 822 IRQ_TYPE_LEVEL_HIGH 0>;
+        resets = <&watchdog 14>;
+        reset-names = "audiosys";
         mediatek,topckgen = <&topckgen>;
         power-domains = <&spm 7>; //MT8195_POWER_DOMAIN_AUDIO
         memory-region = <&snd_dma_mem_reserved>;

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

@@ -28,7 +28,9 @@ properties:
     oneOf:
       - const: nvidia,tegra186-dspk
       - items:
-          - const: nvidia,tegra194-dspk
+          - enum:
+              - nvidia,tegra234-dspk
+              - nvidia,tegra194-dspk
           - const: nvidia,tegra186-dspk
 
   reg:

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

@@ -27,7 +27,9 @@ properties:
           - nvidia,tegra210-admaif
           - nvidia,tegra186-admaif
       - items:
-          - const: nvidia,tegra194-admaif
+          - enum:
+              - nvidia,tegra234-admaif
+              - nvidia,tegra194-admaif
           - const: nvidia,tegra186-admaif
 
   reg:

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

@@ -30,6 +30,7 @@ properties:
       - const: nvidia,tegra210-adx
       - items:
           - enum:
+              - nvidia,tegra234-adx
               - nvidia,tegra194-adx
               - nvidia,tegra186-adx
           - const: nvidia,tegra210-adx

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

@@ -26,6 +26,7 @@ properties:
       - enum:
           - nvidia,tegra210-ahub
           - nvidia,tegra186-ahub
+          - nvidia,tegra234-ahub
       - items:
           - const: nvidia,tegra194-ahub
           - const: nvidia,tegra186-ahub

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

@@ -31,6 +31,9 @@ properties:
           - const: nvidia,tegra186-amx
           - const: nvidia,tegra210-amx
       - const: nvidia,tegra194-amx
+      - items:
+          - const: nvidia,tegra234-amx
+          - const: nvidia,tegra194-amx
 
   reg:
     maxItems: 1

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

@@ -28,6 +28,7 @@ properties:
       - const: nvidia,tegra210-dmic
       - items:
           - enum:
+              - nvidia,tegra234-dmic
               - nvidia,tegra194-dmic
               - nvidia,tegra186-dmic
           - const: nvidia,tegra210-dmic

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

@@ -28,6 +28,7 @@ properties:
       - const: nvidia,tegra210-i2s
       - items:
           - enum:
+              - nvidia,tegra234-i2s
               - nvidia,tegra194-i2s
               - nvidia,tegra186-i2s
           - const: nvidia,tegra210-i2s

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

@@ -28,6 +28,7 @@ properties:
       - const: nvidia,tegra210-amixer
       - items:
           - enum:
+              - nvidia,tegra234-amixer
               - nvidia,tegra194-amixer
               - nvidia,tegra186-amixer
           - const: nvidia,tegra210-amixer

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

@@ -31,6 +31,7 @@ properties:
       - const: nvidia,tegra210-mvc
       - items:
           - enum:
+              - nvidia,tegra234-mvc
               - nvidia,tegra194-mvc
               - nvidia,tegra186-mvc
           - const: nvidia,tegra210-mvc

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

@@ -28,6 +28,7 @@ properties:
       - const: nvidia,tegra210-sfc
       - items:
           - enum:
+              - nvidia,tegra234-sfc
               - nvidia,tegra194-sfc
               - nvidia,tegra186-sfc
           - const: nvidia,tegra210-sfc

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

@@ -22,35 +22,41 @@ properties:
       - qcom,lpass-cpu
       - qcom,apq8016-lpass-cpu
       - qcom,sc7180-lpass-cpu
+      - qcom,sc7280-lpass-cpu
 
   reg:
-    maxItems: 2
+    minItems: 2
+    maxItems: 6
     description: LPAIF core registers
 
   reg-names:
-    maxItems: 2
+    minItems: 2
+    maxItems: 6
 
   clocks:
     minItems: 3
-    maxItems: 6
+    maxItems: 7
 
   clock-names:
     minItems: 3
-    maxItems: 6
+    maxItems: 7
 
   interrupts:
-    maxItems: 2
+    minItems: 2
+    maxItems: 4
     description: LPAIF DMA buffer interrupt
 
   interrupt-names:
-    maxItems: 2
+    minItems: 2
+    maxItems: 4
 
   qcom,adsp:
     $ref: /schemas/types.yaml#/definitions/phandle
     description: Phandle for the audio DSP node
 
   iommus:
-    maxItems: 2
+    minItems: 2
+    maxItems: 3
     description: Phandle to apps_smmu node with sid mask
 
   power-domains:
@@ -69,7 +75,7 @@ patternProperties:
   "^dai-link@[0-9a-f]$":
     type: object
     description: |
-      LPASS CPU dai node for each I2S device. Bindings of each node
+      LPASS CPU dai node for each I2S device or Soundwire device. Bindings of each node
       depends on the specific driver providing the functionality and
       properties.
     properties:
@@ -174,6 +180,59 @@ allOf:
         - iommus
         - power-domains
 
+  - if:
+      properties:
+        compatible:
+          contains:
+            const: qcom,sc7280-lpass-cpu
+
+    then:
+      properties:
+        clock-names:
+          oneOf:
+            - items:   #for I2S
+                - const: aon_cc_audio_hm_h
+                - const: core_cc_sysnoc_mport_core
+                - const: core_cc_ext_if1_ibit
+            - items:   #for Soundwire
+                - const: aon_cc_audio_hm_h
+                - const: audio_cc_codec_mem0
+                - const: audio_cc_codec_mem1
+                - const: audio_cc_codec_mem2
+            - items:   #for HDMI
+                - const: aon_cc_audio_hm_h
+
+        reg-names:
+          anyOf:
+            - items:   #for I2S
+                - const: lpass-lpaif
+            - items:   #for I2S and HDMI
+                - const: lpass-hdmiif
+                - const: lpass-lpaif
+            - items:   #for I2S, soundwire and HDMI
+                - const: lpass-hdmiif
+                - const: lpass-lpaif
+                - const: lpass-rxtx-cdc-dma-lpm
+                - const: lpass-rxtx-lpaif
+                - const: lpass-va-lpaif
+                - const: lpass-va-cdc-dma-lpm
+        interrupt-names:
+          anyOf:
+            - items:   #for I2S
+                - const: lpass-irq-lpaif
+            - items:   #for I2S and HDMI
+                - const: lpass-irq-lpaif
+                - const: lpass-irq-hdmi
+            - items:   #for I2S, soundwire and HDMI
+                - const: lpass-irq-lpaif
+                - const: lpass-irq-hdmi
+                - const: lpass-irq-vaif
+                - const: lpass-irq-rxtxif
+
+      required:
+        - iommus
+        - power-domains
+
 examples:
   - |
     #include <dt-bindings/sound/sc7180-lpass.h>

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

@@ -39,6 +39,14 @@ properties:
     items:
       - const: mclk
 
+  power-domains:
+    maxItems: 2
+
+  power-domain-names:
+    items:
+      - const: macro
+      - const: dcodec
+
 required:
   - compatible
   - reg

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

@@ -39,6 +39,14 @@ properties:
     items:
       - const: mclk
 
+  power-domains:
+    maxItems: 2
+
+  power-domain-names:
+    items:
+      - const: macro
+      - const: dcodec
+
   qcom,dmic-sample-rate:
     description: dmic sample rate
     $ref: /schemas/types.yaml#/definitions/uint32

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

@@ -37,6 +37,14 @@ properties:
     items:
       - const: fsgen
 
+  power-domains:
+    maxItems: 2
+
+  power-domain-names:
+    items:
+      - const: macro
+      - const: dcodec
+
   qcom,dmic-sample-rate:
     description: dmic sample rate
     $ref: /schemas/types.yaml#/definitions/uint32

Documentation/devicetree/bindings/sound/qcom,wcd938x.yaml

@@ -23,6 +23,10 @@ properties:
     description: GPIO spec for reset line to use
     maxItems: 1
 
+  us-euro-gpios:
+    description: GPIO spec for swapping gnd and mic segments
+    maxItems: 1
+
   vdd-buck-supply:
     description: A reference to the 1.8V buck supply
 

Documentation/devicetree/bindings/sound/realtek,rt5682s.yaml

@@ -21,6 +21,7 @@ properties:
     description: I2C address of the device.
 
   interrupts:
+    maxItems: 1
     description: The CODEC's interrupt output.
 
   realtek,dmic1-data-pin:
@@ -94,7 +95,7 @@ required:
 
 examples:
   - |
-    #include <dt-bindings/gpio/tegra-gpio.h>
+    #include <dt-bindings/gpio/gpio.h>
     #include <dt-bindings/interrupt-controller/irq.h>
 
     i2c {
@@ -104,10 +105,9 @@ examples:
         codec@1a {
             compatible = "realtek,rt5682s";
             reg = <0x1a>;
-            interrupt-parent = <&gpio>;
+            interrupts = <6 IRQ_TYPE_LEVEL_HIGH>;
-            interrupts = <TEGRA_GPIO(U, 6) IRQ_TYPE_LEVEL_HIGH>;
             realtek,ldo1-en-gpios =
-                <&gpio TEGRA_GPIO(R, 2) GPIO_ACTIVE_HIGH>;
+                <&gpio 2 GPIO_ACTIVE_HIGH>;
             realtek,dmic1-data-pin = <1>;
             realtek,dmic1-clk-pin = <1>;
             realtek,jd-src = <1>;

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

@@ -123,9 +123,7 @@ properties:
     $ref: audio-graph-port.yaml#
     unevaluatedProperties: false
 
-# use patternProperties to avoid naming "xxx,yyy" issue
+  rcar_sound,dvc:
-patternProperties:
-  "^rcar_sound,dvc$":
     description: DVC subnode.
     type: object
     patternProperties:
@@ -141,7 +139,7 @@ patternProperties:
           - dma-names
     additionalProperties: false
 
-  "^rcar_sound,mix$":
+  rcar_sound,mix:
     description: MIX subnode.
     type: object
     patternProperties:
@@ -150,7 +148,7 @@ patternProperties:
         # no properties
     additionalProperties: false
 
-  "^rcar_sound,ctu$":
+  rcar_sound,ctu:
     description: CTU subnode.
     type: object
     patternProperties:
@@ -159,7 +157,7 @@ patternProperties:
         # no properties
     additionalProperties: false
 
-  "^rcar_sound,src$":
+  rcar_sound,src:
     description: SRC subnode.
     type: object
     patternProperties:
@@ -182,7 +180,7 @@ patternProperties:
           - dma-names
     additionalProperties: false
 
-  "^rcar_sound,ssiu$":
+  rcar_sound,ssiu:
     description: SSIU subnode.
     type: object
     patternProperties:
@@ -202,7 +200,7 @@ patternProperties:
           - dma-names
     additionalProperties: false
 
-  "^rcar_sound,ssi$":
+  rcar_sound,ssi:
     description: SSI subnode.
     type: object
     patternProperties:
@@ -239,7 +237,7 @@ patternProperties:
     additionalProperties: false
 
   # For DAI base
-  "^rcar_sound,dai$":
+  rcar_sound,dai:
     description: DAI subnode.
     type: object
     patternProperties:

Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml

@@ -4,7 +4,7 @@
 $id: http://devicetree.org/schemas/sound/renesas,rz-ssi.yaml#
 $schema: http://devicetree.org/meta-schemas/core.yaml#
 
-title: Renesas RZ/G2L ASoC Sound Serial Interface (SSIF-2)
+title: Renesas RZ/{G2L,V2L} ASoC Sound Serial Interface (SSIF-2)
 
 maintainers:
   - Biju Das <biju.das.jz@bp.renesas.com>
@@ -14,6 +14,7 @@ properties:
     items:
       - enum:
           - renesas,r9a07g044-ssi  # RZ/G2{L,LC}
+          - renesas,r9a07g054-ssi  # RZ/V2L
       - const: renesas,rz-ssi
 
   reg:

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

@@ -31,6 +31,8 @@ properties:
         description: |
           phandles to the I2S controller and bluetooth codec,
           in that order
+    required:
+      - sound-dai
 
   codec:
     type: object
@@ -38,6 +40,8 @@ properties:
       sound-dai:
         $ref: /schemas/types.yaml#/definitions/phandle-array
         description: phandle to the WM8994 CODEC
+    required:
+      - sound-dai
 
   samsung,audio-routing:
     $ref: /schemas/types.yaml#/definitions/non-unique-string-array

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

@@ -0,0 +1,45 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/samsung,arndale.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Insignal Arndale boards audio complex
+
+maintainers:
+  - Krzysztof Kozlowski <krzysztof.kozlowski@canonical.com>
+  - Sylwester Nawrocki <s.nawrocki@samsung.com>
+
+properties:
+  compatible:
+    enum:
+      - samsung,arndale-alc5631
+      - samsung,arndale-rt5631
+      - samsung,arndale-wm1811
+
+  samsung,audio-codec:
+    description: Phandle to the audio codec.
+    $ref: /schemas/types.yaml#/definitions/phandle
+
+  samsung,audio-cpu:
+    description: Phandle to the Samsung I2S controller.
+    $ref: /schemas/types.yaml#/definitions/phandle
+
+  samsung,model:
+    description: The user-visible name of this sound complex.
+    $ref: /schemas/types.yaml#/definitions/string
+
+required:
+  - compatible
+  - samsung,audio-codec
+  - samsung,audio-cpu
+
+additionalProperties: false
+
+examples:
+  - |
+    sound {
+        compatible = "samsung,arndale-rt5631";
+        samsung,audio-cpu = <&i2s0>;
+        samsung,audio-codec = <&rt5631>;
+    };

Documentation/devicetree/bindings/sound/samsung,smdk-wm8994.txt

@@ -1,14 +0,0 @@
-Samsung SMDK audio complex
-
-Required properties:
-- compatible : "samsung,smdk-wm8994"
-- samsung,i2s-controller: The phandle of the Samsung I2S0 controller
-- samsung,audio-codec: The phandle of the WM8994 audio codec
-Example:
-
-sound {
-		compatible = "samsung,smdk-wm8994";
-
-		samsung,i2s-controller = <&i2s0>;
-		samsung,audio-codec = <&wm8994>;
-};

Documentation/devicetree/bindings/sound/samsung,smdk5250.yaml

@@ -0,0 +1,38 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/samsung,smdk5250.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Samsung SMDK5250 audio complex with WM8994 codec
+
+maintainers:
+  - Krzysztof Kozlowski <krzysztof.kozlowski@canonical.com>
+  - Sylwester Nawrocki <s.nawrocki@samsung.com>
+
+properties:
+  compatible:
+    const: samsung,smdk-wm8994
+
+  samsung,audio-codec:
+    description: Phandle to the audio codec.
+    $ref: /schemas/types.yaml#/definitions/phandle
+
+  samsung,i2s-controller:
+    description: Phandle to the Samsung I2S controller.
+    $ref: /schemas/types.yaml#/definitions/phandle
+
+required:
+  - compatible
+  - samsung,audio-codec
+  - samsung,i2s-controller
+
+additionalProperties: false
+
+examples:
+  - |
+    sound {
+        compatible = "samsung,smdk-wm8994";
+        samsung,i2s-controller = <&i2s0>;
+        samsung,audio-codec = <&wm8994>;
+    };

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

@@ -0,0 +1,74 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/samsung,snow.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Google Snow audio complex with MAX9809x codec
+
+maintainers:
+  - Krzysztof Kozlowski <krzysztof.kozlowski@canonical.com>
+  - Sylwester Nawrocki <s.nawrocki@samsung.com>
+
+properties:
+  compatible:
+    enum:
+      - google,snow-audio-max98090
+      - google,snow-audio-max98091
+      - google,snow-audio-max98095
+
+  codec:
+    type: object
+    properties:
+      sound-dai:
+        description: List of phandles to the CODEC and HDMI IP nodes.
+        items:
+          - description: Phandle to the MAX98090, MAX98091 or MAX98095 CODEC.
+          - description: Phandle to the HDMI IP block node.
+    required:
+      - sound-dai
+
+  cpu:
+    type: object
+    properties:
+      sound-dai:
+        description: Phandle to the Samsung I2S controller.
+        maxItems: 1
+    required:
+      - sound-dai
+
+  samsung,audio-codec:
+    description: Phandle to the audio codec.
+    $ref: /schemas/types.yaml#/definitions/phandle
+    deprecated: true
+
+  samsung,i2s-controller:
+    description: Phandle to the Samsung I2S controller.
+    $ref: /schemas/types.yaml#/definitions/phandle
+    deprecated: true
+
+  samsung,model:
+    description: The user-visible name of this sound complex.
+    $ref: /schemas/types.yaml#/definitions/string
+
+required:
+  - compatible
+  - codec
+  - cpu
+
+additionalProperties: false
+
+examples:
+  - |
+    sound {
+        compatible = "google,snow-audio-max98095";
+        samsung,model = "Snow-I2S-MAX98095";
+
+        cpu {
+            sound-dai = <&i2s0 0>;
+        };
+
+        codec {
+            sound-dai = <&max98095 0>, <&hdmi>;
+        };
+    };

Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt

@@ -1,42 +0,0 @@
-Samsung Exynos5433 TM2(E) audio complex with WM5110 codec
-
-Required properties:
-
- - compatible		 : "samsung,tm2-audio"
- - model		 : the user-visible name of this sound complex
- - audio-codec		 : the first entry should be phandle of the wm5110 audio
-			   codec node, as described in ../mfd/arizona.txt;
-			   the second entry should be phandle of the HDMI
-			   transmitter node
- - i2s-controller	 : the list of phandle and argument tuples pointing to
-			   I2S controllers, the first entry should be I2S0 and
-			   the second one I2S1
- - audio-amplifier	 : the phandle of the MAX98504 amplifier
- - samsung,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; valid names for sources and sinks are the
-			   WM5110's and MAX98504's pins and the jacks on the
-			   board: HP, SPK, Main Mic, Sub Mic, Third Mic,
-			   Headset Mic
- - mic-bias-gpios	 : GPIO pin that enables the Main Mic bias regulator
-
-
-Example:
-
-sound {
-	compatible = "samsung,tm2-audio";
-	audio-codec = <&wm5110>, <&hdmi>;
-	i2s-controller = <&i2s0 0>, <&i2s1 0>;
-	audio-amplifier = <&max98504>;
-	mic-bias-gpios = <&gpr3 2 0>;
-	model = "wm5110";
-	samsung,audio-routing =
-		"HP", "HPOUT1L",
-		"HP", "HPOUT1R",
-		"SPK", "SPKOUT",
-		"SPKOUT", "HPOUT2L",
-		"SPKOUT", "HPOUT2R",
-		"Main Mic", "MICBIAS2",
-		"IN1R", "Main Mic";
-};

Documentation/devicetree/bindings/sound/samsung,tm2.yaml

@@ -0,0 +1,80 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/samsung,tm2.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Samsung Exynos5433 TM2(E) audio complex with WM5110 codec
+
+maintainers:
+  - Krzysztof Kozlowski <krzysztof.kozlowski@canonical.com>
+  - Sylwester Nawrocki <s.nawrocki@samsung.com>
+
+properties:
+  compatible:
+    const: samsung,tm2-audio
+
+  audio-amplifier:
+    description: Phandle to the MAX98504 amplifier.
+    $ref: /schemas/types.yaml#/definitions/phandle
+
+  audio-codec:
+    description: Phandles to the codecs.
+    $ref: /schemas/types.yaml#/definitions/phandle-array
+    items:
+      - description: Phandle to the WM5110 audio codec.
+      - description: Phandle to the HDMI transmitter node.
+
+  samsung,audio-routing:
+    description: |
+      List of the connections between audio components; each entry is
+      a pair of strings, the first being the connection's sink, the second
+      being the connection's source; valid names for sources and sinks are the
+      WM5110's and MAX98504's pins and the jacks on the board: HP, SPK, Main
+      Mic, Sub Mic, Third Mic, Headset Mic.
+    $ref: /schemas/types.yaml#/definitions/non-unique-string-array
+
+  i2s-controller:
+    description: Phandles to the I2S controllers.
+    $ref: /schemas/types.yaml#/definitions/phandle-array
+    items:
+      - description: Phandle to I2S0.
+      - description: Phandle to I2S1.
+
+  mic-bias-gpios:
+    description: GPIO pin that enables the Main Mic bias regulator.
+
+  model:
+    description: The user-visible name of this sound complex.
+    $ref: /schemas/types.yaml#/definitions/string
+
+required:
+  - compatible
+  - audio-amplifier
+  - audio-codec
+  - samsung,audio-routing
+  - i2s-controller
+  - mic-bias-gpios
+  - model
+
+additionalProperties: false
+
+examples:
+  - |
+    #include <dt-bindings/gpio/gpio.h>
+
+    sound {
+        compatible = "samsung,tm2-audio";
+        audio-codec = <&wm5110>, <&hdmi>;
+        i2s-controller = <&i2s0 0>, <&i2s1 0>;
+        audio-amplifier = <&max98504>;
+        mic-bias-gpios = <&gpr3 2 GPIO_ACTIVE_HIGH>;
+        model = "wm5110";
+        samsung,audio-routing = "HP", "HPOUT1L",
+                                "HP", "HPOUT1R",
+                                "SPK", "SPKOUT",
+                                "SPKOUT", "HPOUT2L",
+                                "SPKOUT", "HPOUT2R",
+                                "RCV", "HPOUT3L",
+                                "RCV", "HPOUT3R";
+    };

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

@@ -48,6 +48,15 @@ definitions:
       It is useful for some aCPUs with fixed clocks.
     $ref: /schemas/types.yaml#/definitions/flag
 
+  system-clock-fixed:
+    description: |
+      Specifies that the clock frequency should not be modified.
+      Implied when system-clock-frequency is specified, but can be used when
+      a clock is mapped to the device whose frequency cannot or should not be
+      changed. When mclk-fs is also specified, this restricts the device to a
+      single fixed sampling rate.
+    $ref: /schemas/types.yaml#/definitions/flag
+
   mclk-fs:
     description: |
       Multiplication factor between stream rate and codec mclk.
@@ -134,6 +143,8 @@ definitions:
         $ref: "#/definitions/system-clock-frequency"
       system-clock-direction-out:
         $ref: "#/definitions/system-clock-direction-out"
+      system-clock-fixed:
+        $ref: "#/definitions/system-clock-fixed"
     required:
       - sound-dai
 
@@ -156,45 +167,45 @@ properties:
     description: User specified audio sound card name.
     $ref: /schemas/types.yaml#/definitions/string
 
-# use patternProperties to avoid naming "xxx,yyy" issue
+  simple-audio-card,widgets:
-patternProperties:
-  "^simple-audio-card,widgets$":
     $ref: "#/definitions/widgets"
-  "^simple-audio-card,routing$":
+  simple-audio-card,routing:
     $ref: "#/definitions/routing"
-  "^simple-audio-card,cpu(@[0-9a-f]+)?":
-    $ref: "#/definitions/dai"
-  "^simple-audio-card,codec(@[0-9a-f]+)?":
-    $ref: "#/definitions/dai"
 
   # common properties
-  "^simple-audio-card,frame-master$":
+  simple-audio-card,frame-master:
     $ref: "#/definitions/frame-master"
-  "^simple-audio-card,bitclock-master$":
+  simple-audio-card,bitclock-master:
     $ref: "#/definitions/bitclock-master"
-  "^simple-audio-card,frame-inversion$":
+  simple-audio-card,frame-inversion:
     $ref: "#/definitions/frame-inversion"
-  "^simple-audio-card,bitclock-inversion$":
+  simple-audio-card,bitclock-inversion:
     $ref: "#/definitions/bitclock-inversion"
-  "^simple-audio-card,format$":
+  simple-audio-card,format:
     $ref: "#/definitions/format"
-  "^simple-audio-card,mclk-fs$":
+  simple-audio-card,mclk-fs:
     $ref: "#/definitions/mclk-fs"
-  "^simple-audio-card,aux-devs$":
+  simple-audio-card,aux-devs:
     $ref: "#/definitions/aux-devs"
-  "^simple-audio-card,convert-rate$":
+  simple-audio-card,convert-rate:
     $ref: "#/definitions/convert-rate"
-  "^simple-audio-card,convert-channels$":
+  simple-audio-card,convert-channels:
     $ref: "#/definitions/convert-channels"
-  "^simple-audio-card,prefix$":
+  simple-audio-card,prefix:
     $ref: "#/definitions/prefix"
-  "^simple-audio-card,pin-switches$":
+  simple-audio-card,pin-switches:
     $ref: "#/definitions/pin-switches"
-  "^simple-audio-card,hp-det-gpio$":
+  simple-audio-card,hp-det-gpio:
     maxItems: 1
-  "^simple-audio-card,mic-det-gpio$":
+  simple-audio-card,mic-det-gpio:
     maxItems: 1
 
+patternProperties:
+  "^simple-audio-card,cpu(@[0-9a-f]+)?$":
+    $ref: "#/definitions/dai"
+  "^simple-audio-card,codec(@[0-9a-f]+)?$":
+    $ref: "#/definitions/dai"
+
   "^simple-audio-card,dai-link(@[0-9a-f]+)?$":
     description: |
       Container for dai-link level properties and the CPU and CODEC sub-nodes.

Documentation/devicetree/bindings/sound/snow.txt

@@ -1,31 +0,0 @@
-Audio Binding for Snow boards
-
-Required properties:
-- compatible : Can be one of the following,
-			"google,snow-audio-max98090" or
-			"google,snow-audio-max98091" or
-			"google,snow-audio-max98095"
-- samsung,i2s-controller (deprecated): The phandle of the Samsung I2S controller
-- samsung,audio-codec (deprecated): The phandle of the audio codec
-
-Required sub-nodes:
-
- - 'cpu' subnode with a 'sound-dai' property containing the phandle of the I2S
-    controller
- - 'codec' subnode with a 'sound-dai' property containing list of phandles
-    to the CODEC nodes, first entry must be the phandle of the MAX98090,
-    MAX98091 or MAX98095 CODEC (exact device type is indicated by the compatible
-    string) and the second entry must be the phandle of the HDMI IP block node
-
-Optional:
-- samsung,model: The name of the sound-card
-
-Example:
-
-sound {
-		compatible = "google,snow-audio-max98095";
-
-		samsung,model = "Snow-I2S-MAX98095";
-		samsung,i2s-controller = <&i2s0>;
-		samsung,audio-codec = <&max98095>;
-};

Documentation/devicetree/bindings/sound/tas5805m.yaml

@@ -0,0 +1,56 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/tas5805m.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: TAS5805M audio amplifier
+
+maintainers:
+  - Daniel Beer <daniel.beer@igorinstitute.com>
+
+description: |
+  The TAS5805M is a class D audio amplifier with a built-in DSP.
+
+properties:
+  compatible:
+    enum:
+      - ti,tas5805m
+
+  reg:
+    maxItems: 1
+    description: |
+      I2C address of the amplifier. See the datasheet for possible values.
+
+  pvdd-supply:
+    description: |
+      Regulator for audio power supply (PVDD in the datasheet).
+
+  pdn-gpios:
+    description: |
+      Power-down control GPIO (PDN pin in the datasheet).
+
+  ti,dsp-config-name:
+    description: |
+      The name of the DSP configuration that should be loaded for this
+      instance. Configuration blobs are sequences of register writes
+      generated from TI's PPC3 tool.
+    $ref: /schemas/types.yaml#/definitions/string
+
+examples:
+  - |
+    i2c0 {
+        #address-cells = <1>;
+        #size-cells = <0>;
+        tas5805m: tas5805m@2c {
+                reg = <0x2c>;
+                compatible = "ti,tas5805m";
+
+                pvdd-supply = <&audiopwr>;
+                pdn-gpios = <&tlmm 160 0>;
+
+                ti,dsp-config-name = "mono_pbtl_48khz";
+        };
+    };
+
+additionalProperties: true

Documentation/devicetree/bindings/sound/tlv320aic31xx.txt

@@ -58,7 +58,7 @@ The pins can be used in referring sound node's audio-routing property.
 
 Example:
 #include <dt-bindings/gpio/gpio.h>
-#include <dt-bindings/sound/tlv320aic31xx-micbias.h>
+#include <dt-bindings/sound/tlv320aic31xx.h>
 
 tlv320aic31xx: tlv320aic31xx@18 {
 	compatible = "ti,tlv320aic311x";

MAINTAINERS

@@ -7779,10 +7779,10 @@ F:	drivers/net/ethernet/freescale/fs_enet/
 F:	include/linux/fs_enet_pd.h
 
 FREESCALE SOC SOUND DRIVERS
-M:	Nicolin Chen <nicoleotsuka@gmail.com>
+M:	Shengjiu Wang <shengjiu.wang@gmail.com>
 M:	Xiubo Li <Xiubo.Lee@gmail.com>
 R:	Fabio Estevam <festevam@gmail.com>
-R:	Shengjiu Wang <shengjiu.wang@gmail.com>
+R:	Nicolin Chen <nicoleotsuka@gmail.com>
 L:	alsa-devel@alsa-project.org (moderated for non-subscribers)
 L:	linuxppc-dev@lists.ozlabs.org
 S:	Maintained
@@ -19043,6 +19043,7 @@ TEXAS INSTRUMENTS ASoC DRIVERS
 M:	Peter Ujfalusi <peter.ujfalusi@gmail.com>
 L:	alsa-devel@alsa-project.org (moderated for non-subscribers)
 S:	Maintained
+F:	Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml
 F:	sound/soc/ti/
 
 TEXAS INSTRUMENTS' DAC7612 DAC DRIVER

arch/arm/boot/dts/am43x-epos-evm.dts

@@ -11,7 +11,7 @@
 #include <dt-bindings/pinctrl/am43xx.h>
 #include <dt-bindings/gpio/gpio.h>
 #include <dt-bindings/pwm/pwm.h>
-#include <dt-bindings/sound/tlv320aic31xx-micbias.h>
+#include <dt-bindings/sound/tlv320aic31xx.h>
 
 / {
 	model = "TI AM43x EPOS EVM";

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

@@ -586,6 +586,13 @@ static bool cdn_dp_check_link_status(struct cdn_dp_device *dp)
 	return drm_dp_channel_eq_ok(link_status, min(port->lanes, sink_lanes));
 }
 
+static void cdn_dp_audio_handle_plugged_change(struct cdn_dp_device *dp,
+					       bool plugged)
+{
+	if (dp->codec_dev)
+		dp->plugged_cb(dp->codec_dev, plugged);
+}
+
 static void cdn_dp_encoder_enable(struct drm_encoder *encoder)
 {
 	struct cdn_dp_device *dp = encoder_to_dp(encoder);
@@ -641,6 +648,9 @@ static void cdn_dp_encoder_enable(struct drm_encoder *encoder)
 		DRM_DEV_ERROR(dp->dev, "Failed to valid video %d\n", ret);
 		goto out;
 	}
+
+	cdn_dp_audio_handle_plugged_change(dp, true);
+
 out:
 	mutex_unlock(&dp->lock);
 }
@@ -651,6 +661,8 @@ static void cdn_dp_encoder_disable(struct drm_encoder *encoder)
 	int ret;
 
 	mutex_lock(&dp->lock);
+	cdn_dp_audio_handle_plugged_change(dp, false);
+
 	if (dp->active) {
 		ret = cdn_dp_disable(dp);
 		if (ret) {
@@ -846,11 +858,27 @@ static int cdn_dp_audio_get_eld(struct device *dev, void *data,
 	return 0;
 }
 
+static int cdn_dp_audio_hook_plugged_cb(struct device *dev, void *data,
+					hdmi_codec_plugged_cb fn,
+					struct device *codec_dev)
+{
+	struct cdn_dp_device *dp = dev_get_drvdata(dev);
+
+	mutex_lock(&dp->lock);
+	dp->plugged_cb = fn;
+	dp->codec_dev = codec_dev;
+	cdn_dp_audio_handle_plugged_change(dp, dp->connected);
+	mutex_unlock(&dp->lock);
+
+	return 0;
+}
+
 static const struct hdmi_codec_ops audio_codec_ops = {
 	.hw_params = cdn_dp_audio_hw_params,
 	.audio_shutdown = cdn_dp_audio_shutdown,
 	.mute_stream = cdn_dp_audio_mute_stream,
 	.get_eld = cdn_dp_audio_get_eld,
+	.hook_plugged_cb = cdn_dp_audio_hook_plugged_cb,
 	.no_capture_mute = 1,
 };
 

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

@@ -10,6 +10,7 @@
 #include <drm/drm_dp_helper.h>
 #include <drm/drm_panel.h>
 #include <drm/drm_probe_helper.h>
+#include <sound/hdmi-codec.h>
 
 #include "rockchip_drm_drv.h"
 
@@ -101,5 +102,8 @@ struct cdn_dp_device {
 
 	u8 dpcd[DP_RECEIVER_CAP_SIZE];
 	bool sink_has_audio;
+
+	hdmi_codec_plugged_cb plugged_cb;
+	struct device *codec_dev;
 };
 #endif  /* _CDN_DP_CORE_H */

drivers/soundwire/dmi-quirks.c

@@ -59,7 +59,7 @@ static const struct dmi_system_id adr_remap_quirk_table[] = {
 	{
 		.matches = {
 			DMI_MATCH(DMI_SYS_VENDOR, "HP"),
-			DMI_MATCH(DMI_PRODUCT_NAME, "HP Spectre x360 Convertible"),
+			DMI_MATCH(DMI_PRODUCT_NAME, "HP Spectre x360 Conv"),
 		},
 		.driver_data = (void *)intel_tgl_bios,
 	},

include/dt-bindings/sound/microchip,pdmc.h

@@ -0,0 +1,13 @@
+/* SPDX-License-Identifier: GPL-2.0 */
+#ifndef __DT_BINDINGS_MICROCHIP_PDMC_H__
+#define __DT_BINDINGS_MICROCHIP_PDMC_H__
+
+/* PDM microphone's pin placement */
+#define MCHP_PDMC_DS0 0
+#define MCHP_PDMC_DS1 1
+
+/* PDM microphone clock edge sampling */
+#define MCHP_PDMC_CLK_POSITIVE 0
+#define MCHP_PDMC_CLK_NEGATIVE 1
+
+#endif /* __DT_BINDINGS_MICROCHIP_PDMC_H__ */

include/dt-bindings/sound/tlv320aic31xx-micbias.h

@@ -1,9 +0,0 @@
-/* SPDX-License-Identifier: GPL-2.0 */
-#ifndef __DT_TLV320AIC31XX_MICBIAS_H
-#define __DT_TLV320AIC31XX_MICBIAS_H
-
-#define MICBIAS_2_0V		1
-#define MICBIAS_2_5V		2
-#define MICBIAS_AVDDV		3
-
-#endif /* __DT_TLV320AIC31XX_MICBIAS_H */

include/dt-bindings/sound/tlv320aic31xx.h

@@ -0,0 +1,14 @@
+/* SPDX-License-Identifier: GPL-2.0 */
+#ifndef __DT_TLV320AIC31XX_H
+#define __DT_TLV320AIC31XX_H
+
+#define MICBIAS_2_0V		1
+#define MICBIAS_2_5V		2
+#define MICBIAS_AVDDV		3
+
+#define PLL_CLKIN_MCLK		0x00
+#define PLL_CLKIN_BCLK		0x01
+#define PLL_CLKIN_GPIO1		0x02
+#define PLL_CLKIN_DIN		0x03
+
+#endif /* __DT_TLV320AIC31XX_H */

include/sound/hdaudio.h

@@ -9,6 +9,7 @@
 #include <linux/device.h>
 #include <linux/interrupt.h>
 #include <linux/io.h>
+#include <linux/io-64-nonatomic-lo-hi.h>
 #include <linux/pm_runtime.h>
 #include <linux/timecounter.h>
 #include <sound/core.h>
@@ -448,6 +449,8 @@ static inline u16 snd_hdac_reg_readw(struct hdac_bus *bus, void __iomem *addr)
 
 #define snd_hdac_reg_writel(bus, addr, val)	writel(val, addr)
 #define snd_hdac_reg_readl(bus, addr)	readl(addr)
+#define snd_hdac_reg_writeq(bus, addr, val)	writeq(val, addr)
+#define snd_hdac_reg_readq(bus, addr)		readq(addr)
 
 /*
  * macros for easy use

include/sound/hdaudio_ext.h

@@ -2,6 +2,8 @@
 #ifndef __SOUND_HDAUDIO_EXT_H
 #define __SOUND_HDAUDIO_EXT_H
 
+#include <linux/io-64-nonatomic-lo-hi.h>
+#include <linux/iopoll.h>
 #include <sound/hdaudio.h>
 
 int snd_hdac_ext_bus_init(struct hdac_bus *bus, struct device *dev,
@@ -144,6 +146,54 @@ void snd_hdac_ext_bus_link_power(struct hdac_device *codec, bool enable);
 	writew(((readw(addr + reg) & ~(mask)) | (val)), \
 		addr + reg)
 
+#define snd_hdac_adsp_writeb(chip, reg, value) \
+	snd_hdac_reg_writeb(chip, (chip)->dsp_ba + (reg), value)
+#define snd_hdac_adsp_readb(chip, reg) \
+	snd_hdac_reg_readb(chip, (chip)->dsp_ba + (reg))
+#define snd_hdac_adsp_writew(chip, reg, value) \
+	snd_hdac_reg_writew(chip, (chip)->dsp_ba + (reg), value)
+#define snd_hdac_adsp_readw(chip, reg) \
+	snd_hdac_reg_readw(chip, (chip)->dsp_ba + (reg))
+#define snd_hdac_adsp_writel(chip, reg, value) \
+	snd_hdac_reg_writel(chip, (chip)->dsp_ba + (reg), value)
+#define snd_hdac_adsp_readl(chip, reg) \
+	snd_hdac_reg_readl(chip, (chip)->dsp_ba + (reg))
+#define snd_hdac_adsp_writeq(chip, reg, value) \
+	snd_hdac_reg_writeq(chip, (chip)->dsp_ba + (reg), value)
+#define snd_hdac_adsp_readq(chip, reg) \
+	snd_hdac_reg_readq(chip, (chip)->dsp_ba + (reg))
+
+#define snd_hdac_adsp_updateb(chip, reg, mask, val) \
+	snd_hdac_adsp_writeb(chip, reg, \
+			(snd_hdac_adsp_readb(chip, reg) & ~(mask)) | (val))
+#define snd_hdac_adsp_updatew(chip, reg, mask, val) \
+	snd_hdac_adsp_writew(chip, reg, \
+			(snd_hdac_adsp_readw(chip, reg) & ~(mask)) | (val))
+#define snd_hdac_adsp_updatel(chip, reg, mask, val) \
+	snd_hdac_adsp_writel(chip, reg, \
+			(snd_hdac_adsp_readl(chip, reg) & ~(mask)) | (val))
+#define snd_hdac_adsp_updateq(chip, reg, mask, val) \
+	snd_hdac_adsp_writeq(chip, reg, \
+			(snd_hdac_adsp_readq(chip, reg) & ~(mask)) | (val))
+
+#define snd_hdac_adsp_readb_poll(chip, reg, val, cond, delay_us, timeout_us) \
+	readb_poll_timeout((chip)->dsp_ba + (reg), val, cond, \
+			   delay_us, timeout_us)
+#define snd_hdac_adsp_readw_poll(chip, reg, val, cond, delay_us, timeout_us) \
+	readw_poll_timeout((chip)->dsp_ba + (reg), val, cond, \
+			   delay_us, timeout_us)
+#define snd_hdac_adsp_readl_poll(chip, reg, val, cond, delay_us, timeout_us) \
+	readl_poll_timeout((chip)->dsp_ba + (reg), val, cond, \
+			   delay_us, timeout_us)
+#define snd_hdac_adsp_readq_poll(chip, reg, val, cond, delay_us, timeout_us) \
+	readq_poll_timeout((chip)->dsp_ba + (reg), val, cond, \
+			   delay_us, timeout_us)
+#define snd_hdac_stream_readb_poll(strm, reg, val, cond, delay_us, timeout_us) \
+	readb_poll_timeout((strm)->sd_addr + AZX_REG_ ## reg, val, cond, \
+			   delay_us, timeout_us)
+#define snd_hdac_stream_readl_poll(strm, reg, val, cond, delay_us, timeout_us) \
+	readl_poll_timeout((strm)->sd_addr + AZX_REG_ ## reg, val, cond, \
+			   delay_us, timeout_us)
 
 struct hdac_ext_device;
 

include/sound/intel-nhlt.h

@@ -18,6 +18,13 @@ enum nhlt_link_type {
 	NHLT_LINK_INVALID
 };
 
+enum nhlt_device_type {
+	NHLT_DEVICE_BT = 0,
+	NHLT_DEVICE_DMIC = 1,
+	NHLT_DEVICE_I2S = 4,
+	NHLT_DEVICE_INVALID
+};
+
 #if IS_ENABLED(CONFIG_ACPI) && IS_ENABLED(CONFIG_SND_INTEL_NHLT)
 
 struct wav_fmt {
@@ -41,13 +48,6 @@ struct wav_fmt_ext {
 	u8 sub_fmt[16];
 } __packed;
 
-enum nhlt_device_type {
-	NHLT_DEVICE_BT = 0,
-	NHLT_DEVICE_DMIC = 1,
-	NHLT_DEVICE_I2S = 4,
-	NHLT_DEVICE_INVALID
-};
-
 struct nhlt_specific_cfg {
 	u32 size;
 	u8 caps[];
@@ -133,6 +133,9 @@ void intel_nhlt_free(struct nhlt_acpi_table *addr);
 int intel_nhlt_get_dmic_geo(struct device *dev, struct nhlt_acpi_table *nhlt);
 
 bool intel_nhlt_has_endpoint_type(struct nhlt_acpi_table *nhlt, u8 link_type);
+
+int intel_nhlt_ssp_endpoint_mask(struct nhlt_acpi_table *nhlt, u8 device_type);
+
 struct nhlt_specific_cfg *
 intel_nhlt_get_endpoint_blob(struct device *dev, struct nhlt_acpi_table *nhlt,
 			     u32 bus_id, u8 link_type, u8 vbps, u8 bps,
@@ -163,6 +166,11 @@ static inline bool intel_nhlt_has_endpoint_type(struct nhlt_acpi_table *nhlt,
 	return false;
 }
 
+static inline int intel_nhlt_ssp_endpoint_mask(struct nhlt_acpi_table *nhlt, u8 device_type)
+{
+	return 0;
+}
+
 static inline struct nhlt_specific_cfg *
 intel_nhlt_get_endpoint_blob(struct device *dev, struct nhlt_acpi_table *nhlt,
 			     u32 bus_id, u8 link_type, u8 vbps, u8 bps,

include/sound/simple_card_utils.h

@@ -16,6 +16,12 @@
 #define asoc_simple_init_mic(card, sjack, prefix) \
 	asoc_simple_init_jack(card, sjack, 0, prefix, NULL)
 
+struct asoc_simple_tdm_width_map {
+	u8 sample_bits;
+	u8 slot_count;
+	u16 slot_width;
+};
+
 struct asoc_simple_dai {
 	const char *name;
 	unsigned int sysclk;
@@ -25,6 +31,9 @@ struct asoc_simple_dai {
 	unsigned int tx_slot_mask;
 	unsigned int rx_slot_mask;
 	struct clk *clk;
+	bool clk_fixed;
+	struct asoc_simple_tdm_width_map *tdm_width_map;
+	int n_tdm_widths;
 };
 
 struct asoc_simple_data {
@@ -131,6 +140,9 @@ int asoc_simple_parse_daifmt(struct device *dev,
 			     struct device_node *codec,
 			     char *prefix,
 			     unsigned int *retfmt);
+int asoc_simple_parse_tdm_width_map(struct device *dev, struct device_node *np,
+				    struct asoc_simple_dai *dai);
+
 __printf(3, 4)
 int asoc_simple_set_dailink_name(struct device *dev,
 				 struct snd_soc_dai_link *dai_link,

include/sound/soc-acpi.h

@@ -60,9 +60,11 @@ static inline struct snd_soc_acpi_mach *snd_soc_acpi_codec_list(void *arg)
  * @acpi_ipc_irq_index: used for BYT-CR detection
  * @platform: string used for HDAudio codec support
  * @codec_mask: used for HDAudio support
+ * @dmic_num: number of SoC- or chipset-attached PDM digital microphones
  * @common_hdmi_codec_drv: use commom HDAudio HDMI codec driver
- * @link_mask: links enabled on the board
+ * @link_mask: SoundWire links enabled on the board
- * @links: array of link _ADR descriptors, null terminated
+ * @links: array of SoundWire link _ADR descriptors, null terminated
+ * @i2s_link_mask: I2S/TDM links enabled on the board
  * @num_dai_drivers: number of elements in @dai_drivers
  * @dai_drivers: pointer to dai_drivers, used e.g. in nocodec mode
  */
@@ -74,6 +76,7 @@ struct snd_soc_acpi_mach_params {
 	bool common_hdmi_codec_drv;
 	u32 link_mask;
 	const struct snd_soc_acpi_link_adr *links;
+	u32 i2s_link_mask;
 	u32 num_dai_drivers;
 	struct snd_soc_dai_driver *dai_drivers;
 };
@@ -122,6 +125,24 @@ struct snd_soc_acpi_link_adr {
 	const struct snd_soc_acpi_adr_device *adr_d;
 };
 
+/*
+ * when set the topology uses the -ssp<N> suffix, where N is determined based on
+ * BIOS or DMI information
+ */
+#define SND_SOC_ACPI_TPLG_INTEL_SSP_NUMBER BIT(0)
+
+/*
+ * when more than one SSP is reported in the link mask, use the most significant.
+ * This choice was found to be valid on platforms with ES8336 codecs.
+ */
+#define SND_SOC_ACPI_TPLG_INTEL_SSP_MSB BIT(1)
+
+/*
+ * when set the topology uses the -dmic<N>ch suffix, where N is determined based on
+ * BIOS or DMI information
+ */
+#define SND_SOC_ACPI_TPLG_INTEL_DMIC_NUMBER BIT(2)
+
 /**
  * snd_soc_acpi_mach: ACPI-based machine descriptor. Most of the fields are
  * related to the hardware, except for the firmware and topology file names.
@@ -142,8 +163,8 @@ struct snd_soc_acpi_link_adr {
  * audio codecs whose presence if checked with ACPI
  * @pdata: intended for platform data or machine specific-ops. This structure
  *  is not constant since this field may be updated at run-time
- * @sof_fw_filename: Sound Open Firmware file name, if enabled
  * @sof_tplg_filename: Sound Open Firmware topology file name, if enabled
+ * @tplg_quirk_mask: quirks to select different topology files dynamically
  */
 /* Descriptor for SST ASoC machine driver */
 struct snd_soc_acpi_mach {
@@ -158,8 +179,8 @@ struct snd_soc_acpi_mach {
 	const void *quirk_data;
 	void *pdata;
 	struct snd_soc_acpi_mach_params mach_params;
-	const char *sof_fw_filename;
 	const char *sof_tplg_filename;
+	const u32 tplg_quirk_mask;
 };
 
 #define SND_SOC_ACPI_MAX_CODECS 3

include/sound/soc-dapm.h

@@ -429,6 +429,7 @@ struct snd_soc_dapm_widget *snd_soc_dapm_new_control_unlocked(
 		const struct snd_soc_dapm_widget *widget);
 int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm,
 				 struct snd_soc_dai *dai);
+void snd_soc_dapm_free_widget(struct snd_soc_dapm_widget *w);
 int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card);
 void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card);
 

include/sound/sof.h

@@ -39,6 +39,14 @@ enum sof_fw_state {
 	SOF_FW_CRASHED,
 };
 
+/* DSP power states */
+enum sof_dsp_power_states {
+	SOF_DSP_PM_D0,
+	SOF_DSP_PM_D1,
+	SOF_DSP_PM_D2,
+	SOF_DSP_PM_D3,
+};
+
 /*
  * SOF Platform data.
  */

include/sound/sof/dai.h

@@ -116,4 +116,9 @@ struct sof_ipc_dai_config {
 	};
 } __packed;
 
+struct sof_dai_private_data {
+	struct sof_ipc_comp_dai *comp_dai;
+	struct sof_ipc_dai_config *dai_config;
+};
+
 #endif

include/sound/sof/header.h

@@ -31,7 +31,7 @@
 
 /* Global Message - Generic */
 #define SOF_GLB_TYPE_SHIFT			28
-#define SOF_GLB_TYPE_MASK			(0xfL << SOF_GLB_TYPE_SHIFT)
+#define SOF_GLB_TYPE_MASK			(0xfUL << SOF_GLB_TYPE_SHIFT)
 #define SOF_GLB_TYPE(x)				((x) << SOF_GLB_TYPE_SHIFT)
 
 /* Command Message - Generic */

include/sound/sof/info.h

@@ -25,6 +25,7 @@
 #define SOF_IPC_INFO_LOCKS		BIT(1)
 #define SOF_IPC_INFO_LOCKSV		BIT(2)
 #define SOF_IPC_INFO_GDB		BIT(3)
+#define SOF_IPC_INFO_D3_PERSISTENT	BIT(4)
 
 /* extended data types that can be appended onto end of sof_ipc_fw_ready */
 enum sof_ipc_ext_data {

include/sound/sof/topology.h

@@ -87,9 +87,6 @@ struct sof_ipc_comp {
  */
 #define SOF_BUF_UNDERRUN_PERMITTED	BIT(1)
 
-/* the UUID size in bytes, shared between FW and host */
-#define SOF_UUID_SIZE	16
-
 /* create new component buffer - SOF_IPC_TPLG_BUFFER_NEW */
 struct sof_ipc_buffer {
 	struct sof_ipc_comp comp;
@@ -237,7 +234,7 @@ struct sof_ipc_comp_process {
 	/* reserved for future use */
 	uint32_t reserved[7];
 
-	uint8_t data[0];
+	uint8_t data[];
 } __packed;
 
 /* frees components, buffers and pipelines
@@ -303,9 +300,4 @@ enum sof_event_types {
 	SOF_KEYWORD_DETECT_DAPM_EVENT,
 };
 
-/* extended data struct for UUID components */
-struct sof_ipc_comp_ext {
-	uint8_t uuid[SOF_UUID_SIZE];
-}  __packed;
-
 #endif

include/uapi/sound/sof/abi.h

@@ -26,8 +26,8 @@
 
 /* SOF ABI version major, minor and patch numbers */
 #define SOF_ABI_MAJOR 3
-#define SOF_ABI_MINOR 18
+#define SOF_ABI_MINOR 19
-#define SOF_ABI_PATCH 0
+#define SOF_ABI_PATCH 1
 
 /* SOF ABI version number. Format within 32bit word is MMmmmppp */
 #define SOF_ABI_MAJOR_SHIFT	24

sound/hda/intel-dsp-config.c

@@ -11,6 +11,7 @@
 #include <sound/core.h>
 #include <sound/intel-dsp-config.h>
 #include <sound/intel-nhlt.h>
+#include <sound/soc-acpi.h>
 
 static int dsp_driver;
 
@@ -31,7 +32,12 @@ struct config_entry {
 	u16 device;
 	u8 acpi_hid[ACPI_ID_LEN];
 	const struct dmi_system_id *dmi_table;
-	u8 codec_hid[ACPI_ID_LEN];
+	const struct snd_soc_acpi_codecs *codec_hid;
+};
+
+static const struct snd_soc_acpi_codecs __maybe_unused essx_83x6 = {
+	.num_codecs = 3,
+	.codecs = { "ESSX8316", "ESSX8326", "ESSX8336"},
 };
 
 /*
@@ -77,7 +83,7 @@ static const struct config_entry config_table[] = {
 	{
 		.flags = FLAG_SOF,
 		.device = 0x5a98,
-		.codec_hid = "ESSX8336",
+		.codec_hid =  &essx_83x6,
 	},
 #endif
 #if IS_ENABLED(CONFIG_SND_SOC_INTEL_APL)
@@ -163,7 +169,7 @@ static const struct config_entry config_table[] = {
 	{
 		.flags = FLAG_SOF,
 		.device = 0x3198,
-		.codec_hid = "ESSX8336",
+		.codec_hid =  &essx_83x6,
 	},
 #endif
 
@@ -193,6 +199,11 @@ static const struct config_entry config_table[] = {
 			{}
 		}
 	},
+	{
+		.flags = FLAG_SOF,
+		.device = 0x09dc8,
+		.codec_hid =  &essx_83x6,
+	},
 	{
 		.flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE,
 		.device = 0x9dc8,
@@ -251,7 +262,7 @@ static const struct config_entry config_table[] = {
 	{
 		.flags = FLAG_SOF,
 		.device = 0x02c8,
-		.codec_hid = "ESSX8336",
+		.codec_hid =  &essx_83x6,
 	},
 	{
 		.flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE,
@@ -280,7 +291,7 @@ static const struct config_entry config_table[] = {
 	{
 		.flags = FLAG_SOF,
 		.device = 0x06c8,
-		.codec_hid = "ESSX8336",
+		.codec_hid =  &essx_83x6,
 	},
 	{
 		.flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE,
@@ -327,7 +338,7 @@ static const struct config_entry config_table[] = {
 	{
 		.flags = FLAG_SOF,
 		.device = 0x4dc8,
-		.codec_hid = "ESSX8336",
+		.codec_hid =  &essx_83x6,
 	},
 	{
 		.flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC,
@@ -353,7 +364,7 @@ static const struct config_entry config_table[] = {
 	{
 		.flags = FLAG_SOF,
 		.device = 0xa0c8,
-		.codec_hid = "ESSX8336",
+		.codec_hid =  &essx_83x6,
 	},
 	{
 		.flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE,
@@ -414,8 +425,15 @@ static const struct config_entry *snd_intel_dsp_find_config
 			continue;
 		if (table->dmi_table && !dmi_check_system(table->dmi_table))
 			continue;
-		if (table->codec_hid[0] && !acpi_dev_present(table->codec_hid, NULL, -1))
+		if (table->codec_hid) {
-			continue;
+			int i;
+
+			for (i = 0; i < table->codec_hid->num_codecs; i++)
+				if (acpi_dev_present(table->codec_hid->codecs[i], NULL, -1))
+					break;
+			if (i == table->codec_hid->num_codecs)
+				continue;
+		}
 		return table;
 	}
 	return NULL;

sound/hda/intel-nhlt.c

@@ -130,6 +130,28 @@ bool intel_nhlt_has_endpoint_type(struct nhlt_acpi_table *nhlt, u8 link_type)
 }
 EXPORT_SYMBOL(intel_nhlt_has_endpoint_type);
 
+int intel_nhlt_ssp_endpoint_mask(struct nhlt_acpi_table *nhlt, u8 device_type)
+{
+	struct nhlt_endpoint *epnt;
+	int ssp_mask = 0;
+	int i;
+
+	if (!nhlt || (device_type != NHLT_DEVICE_BT && device_type != NHLT_DEVICE_I2S))
+		return 0;
+
+	epnt = (struct nhlt_endpoint *)nhlt->desc;
+	for (i = 0; i < nhlt->endpoint_count; i++) {
+		if (epnt->linktype == NHLT_LINK_SSP && epnt->device_type == device_type) {
+			/* for SSP the virtual bus id is the SSP port */
+			ssp_mask |= BIT(epnt->virtual_bus_id);
+		}
+		epnt = (struct nhlt_endpoint *)((u8 *)epnt + epnt->length);
+	}
+
+	return ssp_mask;
+}
+EXPORT_SYMBOL(intel_nhlt_ssp_endpoint_mask);
+
 static struct nhlt_specific_cfg *
 nhlt_get_specific_cfg(struct device *dev, struct nhlt_fmt *fmt, u8 num_ch,
 		      u32 rate, u8 vbps, u8 bps)

sound/soc/amd/Kconfig

@@ -44,6 +44,7 @@ config SND_SOC_AMD_RV_RT5682_MACH
 
 config SND_SOC_AMD_RENOIR
 	tristate "AMD Audio Coprocessor - Renoir support"
+	select SND_AMD_ACP_CONFIG
 	depends on X86 && PCI
 	help
 	 This option enables ACP support for Renoir platform

sound/soc/amd/acp-config.c

@@ -110,6 +110,15 @@ struct snd_soc_acpi_mach snd_soc_acpi_amd_sof_machines[] = {
 		.fw_filename = "sof-rn.ri",
 		.sof_tplg_filename = "sof-rn-rt5682-max98360.tplg",
 	},
+	{
+		.id = "RTL5682",
+		.drv_name = "rt5682s-rt1019",
+		.pdata = (void *)&acp_quirk_data,
+		.machine_quirk = snd_soc_acpi_codec_list,
+		.quirk_data = &amp_rt1019,
+		.fw_filename = "sof-rn.ri",
+		.sof_tplg_filename = "sof-rn-rt5682-rt1019.tplg",
+	},
 	{
 		.id = "AMDI1019",
 		.drv_name = "renoir-dsp",

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

@@ -1217,9 +1217,8 @@ static const struct snd_soc_component_driver acp_asoc_platform = {
 
 static int acp_audio_probe(struct platform_device *pdev)
 {
-	int status;
+	int status, irq;
 	struct audio_drv_data *audio_drv_data;
-	struct resource *res;
 	const u32 *pdata = pdev->dev.platform_data;
 
 	if (!pdata) {
@@ -1249,13 +1248,11 @@ static int acp_audio_probe(struct platform_device *pdev)
 
 	audio_drv_data->asic_type =  *pdata;
 
-	res = platform_get_resource(pdev, IORESOURCE_IRQ, 0);
+	irq = platform_get_irq(pdev, 0);
-	if (!res) {
+	if (irq < 0)
-		dev_err(&pdev->dev, "IORESOURCE_IRQ FAILED\n");
 		return -ENODEV;
-	}
 
-	status = devm_request_irq(&pdev->dev, res->start, dma_irq_handler,
+	status = devm_request_irq(&pdev->dev, irq, dma_irq_handler,
 				  0, "ACP_IRQ", &pdev->dev);
 	if (status) {
 		dev_err(&pdev->dev, "ACP IRQ request failed\n");

sound/soc/amd/acp/Kconfig

@@ -15,6 +15,9 @@ config SND_SOC_AMD_ACP_COMMON
 
 if SND_SOC_AMD_ACP_COMMON
 
+config SND_SOC_AMD_ACP_PDM
+	tristate
+
 config SND_SOC_AMD_ACP_I2S
 	tristate
 
@@ -22,10 +25,17 @@ config SND_SOC_AMD_ACP_PCM
 	tristate
 	select SND_SOC_ACPI if ACPI
 
+config SND_SOC_AMD_ACP_PCI
+	tristate "AMD ACP PCI Driver Support"
+	depends on X86 && PCI
+	help
+	  This options enables generic PCI driver for ACP device.
+
 config SND_AMD_ASOC_RENOIR
 	tristate "AMD ACP ASOC Renoir Support"
 	select SND_SOC_AMD_ACP_PCM
 	select SND_SOC_AMD_ACP_I2S
+	select SND_SOC_AMD_ACP_PDM
 	depends on X86 && PCI
 	help
 	  This option enables Renoir I2S support on AMD platform.

sound/soc/amd/acp/Makefile

@@ -7,6 +7,8 @@
 #common acp driver
 snd-acp-pcm-objs     := acp-platform.o
 snd-acp-i2s-objs     := acp-i2s.o
+snd-acp-pdm-objs     := acp-pdm.o
+snd-acp-pci-objs     := acp-pci.o
 
 #platform specific driver
 snd-acp-renoir-objs     := acp-renoir.o
@@ -18,6 +20,8 @@ snd-acp-sof-mach-objs     := acp-sof-mach.o
 
 obj-$(CONFIG_SND_SOC_AMD_ACP_PCM) += snd-acp-pcm.o
 obj-$(CONFIG_SND_SOC_AMD_ACP_I2S) += snd-acp-i2s.o
+obj-$(CONFIG_SND_SOC_AMD_ACP_PDM) += snd-acp-pdm.o
+obj-$(CONFIG_SND_SOC_AMD_ACP_PCI) += snd-acp-pci.o
 
 obj-$(CONFIG_SND_AMD_ASOC_RENOIR) += snd-acp-renoir.o
 

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

@@ -23,13 +23,33 @@
 static struct acp_card_drvdata rt5682_rt1019_data = {
 	.hs_cpu_id = I2S_SP,
 	.amp_cpu_id = I2S_SP,
-	.dmic_cpu_id = NONE,
+	.dmic_cpu_id = DMIC,
 	.hs_codec_id = RT5682,
 	.amp_codec_id = RT1019,
-	.dmic_codec_id = NONE,
+	.dmic_codec_id = DMIC,
 	.gpio_spkr_en = EN_SPKR_GPIO_GB,
 };
 
+static struct acp_card_drvdata rt5682s_max_data = {
+	.hs_cpu_id = I2S_SP,
+	.amp_cpu_id = I2S_SP,
+	.dmic_cpu_id = DMIC,
+	.hs_codec_id = RT5682S,
+	.amp_codec_id = MAX98360A,
+	.dmic_codec_id = DMIC,
+	.gpio_spkr_en = EN_SPKR_GPIO_NONE,
+};
+
+static struct acp_card_drvdata rt5682s_rt1019_data = {
+	.hs_cpu_id = I2S_SP,
+	.amp_cpu_id = I2S_SP,
+	.dmic_cpu_id = DMIC,
+	.hs_codec_id = RT5682S,
+	.amp_codec_id = RT1019,
+	.dmic_codec_id = DMIC,
+	.gpio_spkr_en = EN_SPKR_GPIO_NONE,
+};
+
 static const struct snd_kcontrol_new acp_controls[] = {
 	SOC_DAPM_PIN_SWITCH("Headphone Jack"),
 	SOC_DAPM_PIN_SWITCH("Headset Mic"),
@@ -96,9 +116,17 @@ static int acp_asoc_probe(struct platform_device *pdev)
 
 static const struct platform_device_id board_ids[] = {
 	{
-		.name = "rn_rt5682_rt1019",
+		.name = "acp3xalc56821019",
 		.driver_data = (kernel_ulong_t)&rt5682_rt1019_data,
 	},
+	{
+		.name = "acp3xalc5682sm98360",
+		.driver_data = (kernel_ulong_t)&rt5682s_max_data,
+	},
+	{
+		.name = "acp3xalc5682s1019",
+		.driver_data = (kernel_ulong_t)&rt5682s_rt1019_data,
+	},
 	{ }
 };
 static struct platform_driver acp_asoc_audio = {
@@ -113,5 +141,7 @@ module_platform_driver(acp_asoc_audio);
 
 MODULE_IMPORT_NS(SND_SOC_AMD_MACH);
 MODULE_DESCRIPTION("ACP chrome audio support");
-MODULE_ALIAS("platform:rn_rt5682_rt1019");
+MODULE_ALIAS("platform:acp3xalc56821019");
+MODULE_ALIAS("platform:acp3xalc5682sm98360");
+MODULE_ALIAS("platform:acp3xalc5682s1019");
 MODULE_LICENSE("GPL v2");

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

@@ -291,6 +291,32 @@ static const struct snd_soc_ops acp_card_rt5682s_ops = {
 	.shutdown = acp_card_shutdown,
 };
 
+static const unsigned int dmic_channels[] = {
+	DUAL_CHANNEL, FOUR_CHANNEL,
+};
+
+static const struct snd_pcm_hw_constraint_list dmic_constraints_channels = {
+	.count = ARRAY_SIZE(dmic_channels),
+	.list = dmic_channels,
+	.mask = 0,
+};
+
+static int acp_card_dmic_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+
+	snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+				   &dmic_constraints_channels);
+	snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+				   &constraints_rates);
+
+	return 0;
+}
+
+static const struct snd_soc_ops acp_card_dmic_ops = {
+	.startup = acp_card_dmic_startup,
+};
+
 /* Declare RT1019 codec components */
 SND_SOC_DAILINK_DEF(rt1019,
 	DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10EC1019:00", "rt1019-aif"),
@@ -438,6 +464,8 @@ SND_SOC_DAILINK_DEF(sof_sp,
 	DAILINK_COMP_ARRAY(COMP_CPU("acp-sof-sp")));
 SND_SOC_DAILINK_DEF(sof_dmic,
 	DAILINK_COMP_ARRAY(COMP_CPU("acp-sof-dmic")));
+SND_SOC_DAILINK_DEF(pdm_dmic,
+	DAILINK_COMP_ARRAY(COMP_CPU("acp-pdm-dmic")));
 
 int acp_sofdsp_dai_links_create(struct snd_soc_card *card)
 {
@@ -556,6 +584,8 @@ int acp_legacy_dai_links_create(struct snd_soc_card *card)
 		num_links++;
 
 	links = devm_kzalloc(dev, sizeof(struct snd_soc_dai_link) * num_links, GFP_KERNEL);
+	if (!links)
+		return -ENOMEM;
 
 	if (drv_data->hs_cpu_id == I2S_SP) {
 		links[i].name = "acp-headset-codec";
@@ -613,6 +643,26 @@ int acp_legacy_dai_links_create(struct snd_soc_card *card)
 			links[i].ops = &acp_card_maxim_ops;
 			links[i].init = acp_card_maxim_init;
 		}
+		i++;
+	}
+
+	if (drv_data->dmic_cpu_id == DMIC) {
+		links[i].name = "acp-dmic-codec";
+		links[i].id = DMIC_BE_ID;
+		if (drv_data->dmic_codec_id == DMIC) {
+			links[i].codecs = dmic_codec;
+			links[i].num_codecs = ARRAY_SIZE(dmic_codec);
+		} else {
+			/* Use dummy codec if codec id not specified */
+			links[i].codecs = dummy_codec;
+			links[i].num_codecs = ARRAY_SIZE(dummy_codec);
+		}
+		links[i].cpus = pdm_dmic;
+		links[i].num_cpus = ARRAY_SIZE(pdm_dmic);
+		links[i].platforms = platform_component;
+		links[i].num_platforms = ARRAY_SIZE(platform_component);
+		links[i].ops = &acp_card_dmic_ops;
+		links[i].dpcm_capture = 1;
 	}
 
 	card->dai_link = links;

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

@@ -0,0 +1,160 @@
+// SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause)
+//
+// This file is provided under a dual BSD/GPLv2 license. When using or
+// redistributing this file, you may do so under either license.
+//
+// Copyright(c) 2022 Advanced Micro Devices, Inc. All rights reserved.
+//
+// Authors: Ajit Kumar Pandey <AjitKumar.Pandey@amd.com>
+
+/*
+ * Generic PCI interface for ACP device
+ */
+
+#include <linux/delay.h>
+#include <linux/interrupt.h>
+#include <linux/pci.h>
+#include <linux/platform_device.h>
+#include <linux/pm_runtime.h>
+
+#include "amd.h"
+#include "../mach-config.h"
+
+#define DRV_NAME "acp_pci"
+
+#define ACP3x_REG_START	0x1240000
+#define ACP3x_REG_END	0x125C000
+
+static struct platform_device *dmic_dev;
+static struct platform_device *pdev;
+
+static const struct resource acp3x_res[] = {
+	{
+		.start = 0,
+		.end = ACP3x_REG_END - ACP3x_REG_START,
+		.name = "acp_mem",
+		.flags = IORESOURCE_MEM,
+	},
+	{
+		.start = 0,
+		.end = 0,
+		.name = "acp_dai_irq",
+		.flags = IORESOURCE_IRQ,
+	},
+};
+
+static int acp_pci_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
+{
+	struct platform_device_info pdevinfo;
+	struct device *dev = &pci->dev;
+	const struct resource *res_acp;
+	struct acp_chip_info *chip;
+	struct resource *res;
+	unsigned int flag, addr, num_res, i;
+	int ret;
+
+	flag = snd_amd_acp_find_config(pci);
+	if (flag != FLAG_AMD_LEGACY)
+		return -ENODEV;
+
+	chip = devm_kzalloc(&pci->dev, sizeof(*chip), GFP_KERNEL);
+	if (!chip)
+		return -ENOMEM;
+
+	if (pci_enable_device(pci)) {
+		dev_err(&pci->dev, "pci_enable_device failed\n");
+		return -ENODEV;
+	}
+
+	ret = pci_request_regions(pci, "AMD ACP3x audio");
+	if (ret < 0) {
+		dev_err(&pci->dev, "pci_request_regions failed\n");
+		return -ENOMEM;
+	}
+
+	pci_set_master(pci);
+
+	switch (pci->revision) {
+	case 0x01:
+		res_acp = acp3x_res;
+		num_res = ARRAY_SIZE(acp3x_res);
+		chip->name = "acp_asoc_renoir";
+		chip->acp_rev = ACP3X_DEV;
+		break;
+	default:
+		dev_err(dev, "Unsupported device revision:0x%x\n", pci->revision);
+		return -EINVAL;
+	}
+
+	dmic_dev = platform_device_register_data(dev, "dmic-codec", PLATFORM_DEVID_NONE, NULL, 0);
+	if (IS_ERR(dmic_dev)) {
+		dev_err(dev, "failed to create DMIC device\n");
+		return PTR_ERR(dmic_dev);
+	}
+
+	addr = pci_resource_start(pci, 0);
+	chip->base = devm_ioremap(&pci->dev, addr, pci_resource_len(pci, 0));
+
+	res = devm_kzalloc(&pci->dev, sizeof(struct resource) * num_res, GFP_KERNEL);
+	if (!res) {
+		platform_device_unregister(dmic_dev);
+		return -ENOMEM;
+	}
+
+	for (i = 0; i < num_res; i++, res_acp++) {
+		res[i].name = res_acp->name;
+		res[i].flags = res_acp->flags;
+		res[i].start = addr + res_acp->start;
+		res[i].end = addr + res_acp->end;
+		if (res_acp->flags == IORESOURCE_IRQ) {
+			res[i].start = pci->irq;
+			res[i].end = res[i].start;
+		}
+	}
+
+	memset(&pdevinfo, 0, sizeof(pdevinfo));
+
+	pdevinfo.name = chip->name;
+	pdevinfo.id = 0;
+	pdevinfo.parent = &pci->dev;
+	pdevinfo.num_res = num_res;
+	pdevinfo.res = &res[0];
+	pdevinfo.data = chip;
+	pdevinfo.size_data = sizeof(*chip);
+
+	pdev = platform_device_register_full(&pdevinfo);
+	if (IS_ERR(pdev)) {
+		dev_err(&pci->dev, "cannot register %s device\n", pdevinfo.name);
+		platform_device_unregister(dmic_dev);
+		ret = PTR_ERR(pdev);
+	}
+
+	return ret;
+};
+
+static void acp_pci_remove(struct pci_dev *pci)
+{
+	if (dmic_dev)
+		platform_device_unregister(dmic_dev);
+	if (pdev)
+		platform_device_unregister(pdev);
+}
+
+/* PCI IDs */
+static const struct pci_device_id acp_pci_ids[] = {
+	{ PCI_DEVICE(PCI_VENDOR_ID_AMD, ACP_PCI_DEV_ID)},
+	{ 0, }
+};
+MODULE_DEVICE_TABLE(pci, acp_pci_ids);
+
+/* pci_driver definition */
+static struct pci_driver snd_amd_acp_pci_driver = {
+	.name = KBUILD_MODNAME,
+	.id_table = acp_pci_ids,
+	.probe = acp_pci_probe,
+	.remove = acp_pci_remove,
+};
+module_pci_driver(snd_amd_acp_pci_driver);
+
+MODULE_LICENSE("Dual BSD/GPL");
+MODULE_ALIAS(DRV_NAME);

sound/soc/amd/acp/acp-pdm.c

@@ -0,0 +1,193 @@
+// SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause)
+//
+// This file is provided under a dual BSD/GPLv2 license. When using or
+// redistributing this file, you may do so under either license.
+//
+// Copyright(c) 2022 Advanced Micro Devices, Inc.
+//
+// Authors: Ajit Kumar Pandey <AjitKumar.Pandey@amd.com>
+//	    Vijendar Mukunda <Vijendar.Mukunda@amd.com>
+//
+
+/*
+ * Generic Hardware interface for ACP Audio PDM controller
+ */
+
+#include <linux/err.h>
+#include <linux/io.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+
+#include "amd.h"
+
+#define DRV_NAME "acp-pdm"
+
+#define PDM_DMA_STAT		0x10
+#define PDM_DMA_INTR_MASK	0x10000
+#define PDM_DEC_64		0x2
+#define PDM_CLK_FREQ_MASK	0x07
+#define PDM_MISC_CTRL_MASK	0x10
+#define PDM_ENABLE		0x01
+#define PDM_DISABLE		0x00
+#define DMA_EN_MASK		0x02
+#define DELAY_US		5
+#define PDM_TIMEOUT		1000
+#define ACP_REGION2_OFFSET	0x02000000
+
+static int acp_dmic_prepare(struct snd_pcm_substream *substream,
+			    struct snd_soc_dai *dai)
+{
+	struct acp_stream *stream = substream->runtime->private_data;
+	struct device *dev = dai->component->dev;
+	struct acp_dev_data *adata = dev_get_drvdata(dev);
+	u32 physical_addr, size_dmic, period_bytes;
+	unsigned int dmic_ctrl;
+
+	/* Enable default DMIC clk */
+	writel(PDM_CLK_FREQ_MASK, adata->acp_base + ACP_WOV_CLK_CTRL);
+	dmic_ctrl = readl(adata->acp_base + ACP_WOV_MISC_CTRL);
+	dmic_ctrl |= PDM_MISC_CTRL_MASK;
+	writel(dmic_ctrl, adata->acp_base + ACP_WOV_MISC_CTRL);
+
+	period_bytes = frames_to_bytes(substream->runtime,
+			substream->runtime->period_size);
+	size_dmic = frames_to_bytes(substream->runtime,
+			substream->runtime->buffer_size);
+
+	physical_addr = stream->reg_offset + MEM_WINDOW_START;
+
+	/* Init DMIC Ring buffer */
+	writel(physical_addr, adata->acp_base + ACP_WOV_RX_RINGBUFADDR);
+	writel(size_dmic, adata->acp_base + ACP_WOV_RX_RINGBUFSIZE);
+	writel(period_bytes, adata->acp_base + ACP_WOV_RX_INTR_WATERMARK_SIZE);
+	writel(0x01, adata->acp_base + ACPAXI2AXI_ATU_CTRL);
+
+	return 0;
+}
+
+static int acp_dmic_dai_trigger(struct snd_pcm_substream *substream,
+				int cmd, struct snd_soc_dai *dai)
+{
+	struct device *dev = dai->component->dev;
+	struct acp_dev_data *adata = dev_get_drvdata(dev);
+	unsigned int dma_enable;
+	int ret = 0;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		dma_enable = readl(adata->acp_base + ACP_WOV_PDM_DMA_ENABLE);
+		if (!(dma_enable & DMA_EN_MASK)) {
+			writel(PDM_ENABLE, adata->acp_base + ACP_WOV_PDM_ENABLE);
+			writel(PDM_ENABLE, adata->acp_base + ACP_WOV_PDM_DMA_ENABLE);
+		}
+
+		ret = readl_poll_timeout_atomic(adata->acp_base + ACP_WOV_PDM_DMA_ENABLE,
+						dma_enable, (dma_enable & DMA_EN_MASK),
+						DELAY_US, PDM_TIMEOUT);
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		dma_enable = readl(adata->acp_base + ACP_WOV_PDM_DMA_ENABLE);
+		if ((dma_enable & DMA_EN_MASK)) {
+			writel(PDM_DISABLE, adata->acp_base + ACP_WOV_PDM_ENABLE);
+			writel(PDM_DISABLE, adata->acp_base + ACP_WOV_PDM_DMA_ENABLE);
+
+		}
+
+		ret = readl_poll_timeout_atomic(adata->acp_base + ACP_WOV_PDM_DMA_ENABLE,
+						dma_enable, !(dma_enable & DMA_EN_MASK),
+						DELAY_US, PDM_TIMEOUT);
+		break;
+	default:
+		ret = -EINVAL;
+		break;
+	}
+
+	return ret;
+}
+
+static int acp_dmic_hwparams(struct snd_pcm_substream *substream,
+			     struct snd_pcm_hw_params *hwparams, struct snd_soc_dai *dai)
+{
+	struct device *dev = dai->component->dev;
+	struct acp_dev_data *adata = dev_get_drvdata(dev);
+	unsigned int channels, ch_mask;
+
+	channels = params_channels(hwparams);
+	switch (channels) {
+	case 2:
+		ch_mask = 0;
+		break;
+	case 4:
+		ch_mask = 1;
+		break;
+	case 6:
+		ch_mask = 2;
+		break;
+	default:
+		dev_err(dev, "Invalid channels %d\n", channels);
+		return -EINVAL;
+	}
+
+	if (params_format(hwparams) != SNDRV_PCM_FORMAT_S32_LE) {
+		dev_err(dai->dev, "Invalid format:%d\n", params_format(hwparams));
+		return -EINVAL;
+	}
+
+	writel(ch_mask, adata->acp_base + ACP_WOV_PDM_NO_OF_CHANNELS);
+	writel(PDM_DEC_64, adata->acp_base + ACP_WOV_PDM_DECIMATION_FACTOR);
+
+	return 0;
+}
+
+static int acp_dmic_dai_startup(struct snd_pcm_substream *substream,
+				struct snd_soc_dai *dai)
+{
+	struct acp_stream *stream = substream->runtime->private_data;
+	struct device *dev = dai->component->dev;
+	struct acp_dev_data *adata = dev_get_drvdata(dev);
+	u32 ext_int_ctrl;
+
+	stream->dai_id = DMIC_INSTANCE;
+	stream->irq_bit = BIT(PDM_DMA_STAT);
+	stream->pte_offset = ACP_SRAM_PDM_PTE_OFFSET;
+	stream->reg_offset = ACP_REGION2_OFFSET;
+
+	/* Enable DMIC Interrupts */
+	ext_int_ctrl = readl(adata->acp_base + ACP_EXTERNAL_INTR_CNTL);
+	ext_int_ctrl |= PDM_DMA_INTR_MASK;
+	writel(ext_int_ctrl, adata->acp_base + ACP_EXTERNAL_INTR_CNTL);
+
+	return 0;
+}
+
+static void acp_dmic_dai_shutdown(struct snd_pcm_substream *substream,
+				  struct snd_soc_dai *dai)
+{
+	struct device *dev = dai->component->dev;
+	struct acp_dev_data *adata = dev_get_drvdata(dev);
+	u32 ext_int_ctrl;
+
+	/* Disable DMIC interrrupts */
+	ext_int_ctrl = readl(adata->acp_base + ACP_EXTERNAL_INTR_CNTL);
+	ext_int_ctrl |= ~PDM_DMA_INTR_MASK;
+	writel(ext_int_ctrl, adata->acp_base + ACP_EXTERNAL_INTR_CNTL);
+}
+
+const struct snd_soc_dai_ops acp_dmic_dai_ops = {
+	.prepare	= acp_dmic_prepare,
+	.hw_params	= acp_dmic_hwparams,
+	.trigger	= acp_dmic_dai_trigger,
+	.startup	= acp_dmic_dai_startup,
+	.shutdown	= acp_dmic_dai_shutdown,
+};
+EXPORT_SYMBOL_NS_GPL(acp_dmic_dai_ops, SND_SOC_ACP_COMMON);
+
+MODULE_LICENSE("Dual BSD/GPL");
+MODULE_ALIAS(DRV_NAME);

sound/soc/amd/acp/acp-renoir.c

@@ -25,15 +25,46 @@
 
 #define DRV_NAME "acp_asoc_renoir"
 
+#define ACP_SOFT_RST_DONE_MASK	0x00010001
+
+#define ACP_PWR_ON_MASK		0x01
+#define ACP_PWR_OFF_MASK	0x00
+#define ACP_PGFSM_STAT_MASK	0x03
+#define ACP_POWERED_ON		0x00
+#define ACP_PWR_ON_IN_PROGRESS	0x01
+#define ACP_POWERED_OFF		0x02
+#define DELAY_US		5
+#define ACP_TIMEOUT		500
+
+#define ACP_ERROR_MASK 0x20000000
+#define ACP_EXT_INTR_STAT_CLEAR_MASK 0xFFFFFFFF
+
 static struct snd_soc_acpi_codecs amp_rt1019 = {
 	.num_codecs = 1,
 	.codecs = {"10EC1019"}
 };
 
+static struct snd_soc_acpi_codecs amp_max = {
+	.num_codecs = 1,
+	.codecs = {"MX98360A"}
+};
+
 static struct snd_soc_acpi_mach snd_soc_acpi_amd_acp_machines[] = {
 	{
 		.id = "10EC5682",
-		.drv_name = "rn_rt5682_rt1019",
+		.drv_name = "acp3xalc56821019",
+		.machine_quirk = snd_soc_acpi_codec_list,
+		.quirk_data = &amp_rt1019,
+	},
+	{
+		.id = "RTL5682",
+		.drv_name = "acp3xalc5682sm98360",
+		.machine_quirk = snd_soc_acpi_codec_list,
+		.quirk_data = &amp_max,
+	},
+	{
+		.id = "RTL5682",
+		.drv_name = "acp3xalc5682s1019",
 		.machine_quirk = snd_soc_acpi_codec_list,
 		.quirk_data = &amp_rt1019,
 	},
@@ -97,13 +128,145 @@ static struct snd_soc_dai_driver acp_renoir_dai[] = {
 	.ops = &asoc_acp_cpu_dai_ops,
 	.probe = &asoc_acp_i2s_probe,
 },
+{
+	.name = "acp-pdm-dmic",
+	.id = DMIC_INSTANCE,
+	.capture = {
+		.rates = SNDRV_PCM_RATE_8000_48000,
+		.formats = SNDRV_PCM_FMTBIT_S32_LE,
+		.channels_min = 2,
+		.channels_max = 2,
+		.rate_min = 8000,
+		.rate_max = 48000,
+	},
+	.ops = &acp_dmic_dai_ops,
+},
 };
 
+static int acp3x_power_on(void __iomem *base)
+{
+	u32 val;
+
+	val = readl(base + ACP_PGFSM_STATUS);
+
+	if (val == ACP_POWERED_ON)
+		return 0;
+
+	if ((val & ACP_PGFSM_STAT_MASK) != ACP_PWR_ON_IN_PROGRESS)
+		writel(ACP_PWR_ON_MASK, base + ACP_PGFSM_CONTROL);
+
+	return readl_poll_timeout(base + ACP_PGFSM_STATUS, val, !val, DELAY_US, ACP_TIMEOUT);
+}
+
+static int acp3x_power_off(void __iomem *base)
+{
+	u32 val;
+
+	writel(ACP_PWR_OFF_MASK, base + ACP_PGFSM_CONTROL);
+
+	return readl_poll_timeout(base + ACP_PGFSM_STATUS, val,
+				  (val & ACP_PGFSM_STAT_MASK) == ACP_POWERED_OFF,
+				  DELAY_US, ACP_TIMEOUT);
+}
+
+static int acp3x_reset(void __iomem *base)
+{
+	u32 val;
+	int ret;
+
+	writel(1, base + ACP_SOFT_RESET);
+
+	ret = readl_poll_timeout(base + ACP_SOFT_RESET, val, val & ACP_SOFT_RST_DONE_MASK,
+				 DELAY_US, ACP_TIMEOUT);
+	if (ret)
+		return ret;
+
+	writel(0, base + ACP_SOFT_RESET);
+
+	return readl_poll_timeout(base + ACP_SOFT_RESET, val, !val, DELAY_US, ACP_TIMEOUT);
+}
+
+static void acp3x_enable_interrupts(void __iomem *base)
+{
+	u32 ext_intr_ctrl;
+
+	writel(0x01, base + ACP_EXTERNAL_INTR_ENB);
+	ext_intr_ctrl = readl(base + ACP_EXTERNAL_INTR_CNTL);
+	ext_intr_ctrl |= ACP_ERROR_MASK;
+	writel(ext_intr_ctrl, base + ACP_EXTERNAL_INTR_CNTL);
+}
+
+static void acp3x_disable_interrupts(void __iomem *base)
+{
+	writel(ACP_EXT_INTR_STAT_CLEAR_MASK, base + ACP_EXTERNAL_INTR_STAT);
+	writel(0x00, base + ACP_EXTERNAL_INTR_ENB);
+}
+
+static int rn_acp_init(void __iomem *base)
+{
+	int ret;
+
+	/* power on */
+	ret = acp3x_power_on(base);
+	if (ret)
+		return ret;
+
+	writel(0x01, base + ACP_CONTROL);
+
+	/* Reset */
+	ret = acp3x_reset(base);
+	if (ret)
+		return ret;
+
+	acp3x_enable_interrupts(base);
+
+	return 0;
+}
+
+static int rn_acp_deinit(void __iomem *base)
+{
+	int ret = 0;
+
+	acp3x_disable_interrupts(base);
+
+	/* Reset */
+	ret = acp3x_reset(base);
+	if (ret)
+		return ret;
+
+	writel(0x00, base + ACP_CONTROL);
+
+	/* power off */
+	ret = acp3x_power_off(base);
+	if (ret)
+		return ret;
+
+	return 0;
+}
 static int renoir_audio_probe(struct platform_device *pdev)
 {
 	struct device *dev = &pdev->dev;
+	struct acp_chip_info *chip;
 	struct acp_dev_data *adata;
 	struct resource *res;
+	int ret;
+
+	chip = dev_get_platdata(&pdev->dev);
+	if (!chip || !chip->base) {
+		dev_err(&pdev->dev, "ACP chip data is NULL\n");
+		return -ENODEV;
+	}
+
+	if (chip->acp_rev != ACP3X_DEV) {
+		dev_err(&pdev->dev, "Un-supported ACP Revision %d\n", chip->acp_rev);
+		return -ENODEV;
+	}
+
+	ret = rn_acp_init(chip->base);
+	if (ret) {
+		dev_err(&pdev->dev, "ACP Init failed\n");
+		return -EINVAL;
+	}
 
 	adata = devm_kzalloc(dev, sizeof(struct acp_dev_data), GFP_KERNEL);
 	if (!adata)
@@ -119,13 +282,11 @@ static int renoir_audio_probe(struct platform_device *pdev)
 	if (!adata->acp_base)
 		return -ENOMEM;
 
-	res = platform_get_resource_byname(pdev, IORESOURCE_IRQ, "acp_dai_irq");
+	ret = platform_get_irq_byname(pdev, "acp_dai_irq");
-	if (!res) {
+	if (ret < 0)
-		dev_err(&pdev->dev, "IORESOURCE_IRQ FAILED\n");
+		return ret;
-		return -ENODEV;
+	adata->i2s_irq = ret;
-	}
 
-	adata->i2s_irq = res->start;
 	adata->dev = dev;
 	adata->dai_driver = acp_renoir_dai;
 	adata->num_dai = ARRAY_SIZE(acp_renoir_dai);
@@ -142,6 +303,20 @@ static int renoir_audio_probe(struct platform_device *pdev)
 static int renoir_audio_remove(struct platform_device *pdev)
 {
 	struct device *dev = &pdev->dev;
+	struct acp_chip_info *chip;
+	int ret;
+
+	chip = dev_get_platdata(&pdev->dev);
+	if (!chip || !chip->base) {
+		dev_err(&pdev->dev, "ACP chip data is NULL\n");
+		return -ENODEV;
+	}
+
+	ret = rn_acp_deinit(chip->base);
+	if (ret) {
+		dev_err(&pdev->dev, "ACP de-init Failed\n");
+		return -EINVAL;
+	}
 
 	acp_platform_unregister(dev);
 	return 0;

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

@@ -40,6 +40,15 @@ static struct acp_card_drvdata sof_rt5682_max_data = {
 	.gpio_spkr_en = EN_SPKR_GPIO_NONE,
 };
 
+static struct acp_card_drvdata sof_rt5682s_rt1019_data = {
+	.hs_cpu_id = I2S_SP,
+	.amp_cpu_id = I2S_SP,
+	.dmic_cpu_id = DMIC,
+	.hs_codec_id = RT5682S,
+	.amp_codec_id = RT1019,
+	.dmic_codec_id = DMIC,
+};
+
 static struct acp_card_drvdata sof_rt5682s_max_data = {
 	.hs_cpu_id = I2S_SP,
 	.amp_cpu_id = I2S_SP,
@@ -126,6 +135,10 @@ static const struct platform_device_id board_ids[] = {
 		.name = "rt5682s-max",
 		.driver_data = (kernel_ulong_t)&sof_rt5682s_max_data
 	},
+	{
+		.name = "rt5682s-rt1019",
+		.driver_data = (kernel_ulong_t)&sof_rt5682s_rt1019_data
+	},
 	{ }
 };
 static struct platform_driver acp_asoc_audio = {
@@ -143,4 +156,5 @@ MODULE_DESCRIPTION("ACP chrome SOF audio support");
 MODULE_ALIAS("platform:rt5682-rt1019");
 MODULE_ALIAS("platform:rt5682-max");
 MODULE_ALIAS("platform:rt5682s-max");
+MODULE_ALIAS("platform:rt5682s-rt1019");
 MODULE_LICENSE("GPL v2");

sound/soc/amd/acp/amd.h

@@ -12,13 +12,19 @@
 #define __AMD_ACP_H
 
 #include <sound/pcm.h>
+#include <sound/soc.h>
 #include <sound/soc-acpi.h>
+#include <sound/soc-dai.h>
+
 #include "chip_offset_byte.h"
 
+#define ACP3X_DEV			3
+
 #define I2S_SP_INSTANCE			0x00
 #define I2S_BT_INSTANCE			0x01
+#define DMIC_INSTANCE			0x02
 
-#define MEM_WINDOW_START		0x4000000
+#define MEM_WINDOW_START		0x4080000
 
 #define ACP_I2S_REG_START		0x1242400
 #define ACP_I2S_REG_END			0x1242810
@@ -38,6 +44,7 @@
 #define ACP_SRAM_SP_CP_PTE_OFFSET	0x100
 #define ACP_SRAM_BT_PB_PTE_OFFSET	0x200
 #define ACP_SRAM_BT_CP_PTE_OFFSET	0x300
+#define ACP_SRAM_PDM_PTE_OFFSET		0x400
 #define PAGE_SIZE_4K_ENABLE		0x2
 
 #define I2S_SP_TX_MEM_WINDOW_START	0x4000000
@@ -68,6 +75,12 @@
 
 #define ACP_MAX_STREAM			6
 
+struct acp_chip_info {
+	char *name;		/* Platform name */
+	unsigned int acp_rev;	/* ACP Revision id */
+	void __iomem *base;	/* ACP memory PCI base */
+};
+
 struct acp_stream {
 	struct snd_pcm_substream *substream;
 	int irq_bit;
@@ -96,6 +109,7 @@ struct acp_dev_data {
 };
 
 extern const struct snd_soc_dai_ops asoc_acp_cpu_dai_ops;
+extern const struct snd_soc_dai_ops acp_dmic_dai_ops;
 
 int asoc_acp_i2s_probe(struct snd_soc_dai *dai);
 int acp_platform_register(struct device *dev);
@@ -103,6 +117,9 @@ int acp_platform_unregister(struct device *dev);
 
 int acp_machine_select(struct acp_dev_data *adata);
 
+/* Machine configuration */
+int snd_amd_acp_find_config(struct pci_dev *pci);
+
 static inline u64 acp_get_byte_count(struct acp_dev_data *adata, int dai_id, int direction)
 {
 	u64 byte_count, low = 0, high = 0;
@@ -131,6 +148,10 @@ static inline u64 acp_get_byte_count(struct acp_dev_data *adata, int dai_id, int
 			high = readl(adata->acp_base + ACP_I2S_RX_LINEARPOSITIONCNTR_HIGH);
 			low = readl(adata->acp_base + ACP_I2S_RX_LINEARPOSITIONCNTR_LOW);
 			break;
+		case DMIC_INSTANCE:
+			high = readl(adata->acp_base + ACP_WOV_RX_LINEARPOSITIONCNTR_HIGH);
+			low = readl(adata->acp_base + ACP_WOV_RX_LINEARPOSITIONCNTR_LOW);
+			break;
 		default:
 			dev_err(adata->dev, "Invalid dai id %x\n", dai_id);
 			return -EINVAL;

sound/soc/amd/acp/chip_offset_byte.h

@@ -14,6 +14,12 @@
 #define ACPAXI2AXI_ATU_CTRL                           0xC40
 #define ACPAXI2AXI_ATU_PAGE_SIZE_GRP_5                0xC20
 #define ACPAXI2AXI_ATU_BASE_ADDR_GRP_5                0xC24
+
+#define ACP_PGFSM_CONTROL			0x141C
+#define ACP_PGFSM_STATUS                        0x1420
+#define ACP_SOFT_RESET                          0x1000
+#define ACP_CONTROL                             0x1004
+
 #define ACP_EXTERNAL_INTR_ENB                         0x1800
 #define ACP_EXTERNAL_INTR_CNTL                        0x1804
 #define ACP_EXTERNAL_INTR_STAT                        0x1808
@@ -73,4 +79,24 @@
 #define ACP_BTTDM_ITER                                0x280C
 #define ACP_BTTDM_TXFRMT                              0x2810
 
+/* Registers from ACP_WOV_PDM block */
+
+#define ACP_WOV_PDM_ENABLE                            0x2C04
+#define ACP_WOV_PDM_DMA_ENABLE                        0x2C08
+#define ACP_WOV_RX_RINGBUFADDR                        0x2C0C
+#define ACP_WOV_RX_RINGBUFSIZE                        0x2C10
+#define ACP_WOV_RX_LINKPOSITIONCNTR                   0x2C14
+#define ACP_WOV_RX_LINEARPOSITIONCNTR_HIGH            0x2C18
+#define ACP_WOV_RX_LINEARPOSITIONCNTR_LOW             0x2C1C
+#define ACP_WOV_RX_INTR_WATERMARK_SIZE                0x2C20
+#define ACP_WOV_PDM_FIFO_FLUSH                        0x2C24
+#define ACP_WOV_PDM_NO_OF_CHANNELS                    0x2C28
+#define ACP_WOV_PDM_DECIMATION_FACTOR                 0x2C2C
+#define ACP_WOV_PDM_VAD_CTRL                          0x2C30
+#define ACP_WOV_BUFFER_STATUS                         0x2C58
+#define ACP_WOV_MISC_CTRL                             0x2C5C
+#define ACP_WOV_CLK_CTRL                              0x2C60
+#define ACP_PDM_VAD_DYNAMIC_CLK_GATING_EN             0x2C64
+#define ACP_WOV_ERROR_STATUS_REGISTER                 0x2C68
+
 #endif

sound/soc/amd/mach-config.h

@@ -14,6 +14,7 @@
 
 #define FLAG_AMD_SOF			BIT(1)
 #define FLAG_AMD_SOF_ONLY_DMIC		BIT(2)
+#define FLAG_AMD_LEGACY			BIT(3)
 
 #define ACP_PCI_DEV_ID			0x15E2
 

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

@@ -394,13 +394,10 @@ static int acp3x_audio_probe(struct platform_device *pdev)
 	if (!adata->acp3x_base)
 		return -ENOMEM;
 
-	res = platform_get_resource(pdev, IORESOURCE_IRQ, 0);
+	status = platform_get_irq(pdev, 0);
-	if (!res) {
+	if (status < 0)
-		dev_err(&pdev->dev, "IORESOURCE_IRQ FAILED\n");
+		return status;
-		return -ENODEV;
+	adata->i2s_irq = status;
-	}
-
-	adata->i2s_irq = res->start;
 
 	dev_set_drvdata(&pdev->dev, adata);
 	status = devm_snd_soc_register_component(&pdev->dev,

sound/soc/amd/raven/acp3x.h

@@ -87,7 +87,7 @@ struct acp3x_platform_info {
 
 struct i2s_dev_data {
 	bool tdm_mode;
-	unsigned int i2s_irq;
+	int i2s_irq;
 	u16 i2s_instance;
 	u32 tdm_fmt;
 	u32 substream_type;

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

@@ -399,13 +399,11 @@ static int acp_pdm_audio_probe(struct platform_device *pdev)
 	if (!adata->acp_base)
 		return -ENOMEM;
 
-	res = platform_get_resource(pdev, IORESOURCE_IRQ, 0);
+	status = platform_get_irq(pdev, 0);
-	if (!res) {
+	if (status < 0)
-		dev_err(&pdev->dev, "IORESOURCE_IRQ FAILED\n");
+		return status;
-		return -ENODEV;
+	adata->pdm_irq = status;
-	}
 
-	adata->pdm_irq = res->start;
 	adata->capture_stream = NULL;
 
 	dev_set_drvdata(&pdev->dev, adata);

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

@@ -212,10 +212,15 @@ static int snd_rn_acp_probe(struct pci_dev *pci,
 	acpi_integer dmic_status;
 #endif
 	const struct dmi_system_id *dmi_id;
-	unsigned int irqflags;
+	unsigned int irqflags, flag;
 	int ret, index;
 	u32 addr;
 
+	/* Return if acp config flag is defined */
+	flag = snd_amd_acp_find_config(pci);
+	if (flag)
+		return -ENODEV;
+
 	/* Renoir device check */
 	if (pci->revision != 0x01)
 		return -ENODEV;

sound/soc/amd/renoir/rn_acp3x.h

@@ -88,3 +88,6 @@ static inline void rn_writel(u32 val, void __iomem *base_addr)
 {
 	writel(val, base_addr - ACP_PHY_BASE_ADDRESS);
 }
+
+/* Machine configuration */
+int snd_amd_acp_find_config(struct pci_dev *pci);

sound/soc/amd/vangogh/acp5x-mach.c

@@ -110,7 +110,7 @@ static struct snd_pcm_hw_constraint_list constraints_sample_bits = {
 static int acp5x_8821_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct snd_soc_card *card = rtd->card;
 	struct acp5x_platform_info *machine = snd_soc_card_get_drvdata(card);
 
@@ -131,7 +131,7 @@ static int acp5x_8821_startup(struct snd_pcm_substream *substream)
 static int acp5x_nau8821_hw_params(struct snd_pcm_substream *substream,
 				   struct snd_pcm_hw_params *params)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct snd_soc_card *card = rtd->card;
 	struct snd_soc_dai *codec_dai =
 			snd_soc_card_get_codec_dai(card,
@@ -153,7 +153,7 @@ static int acp5x_nau8821_hw_params(struct snd_pcm_substream *substream,
 static int acp5x_cs35l41_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct snd_soc_card *card = rtd->card;
 	struct acp5x_platform_info *machine = snd_soc_card_get_drvdata(card);
 
@@ -170,13 +170,14 @@ static int acp5x_cs35l41_startup(struct snd_pcm_substream *substream)
 static int acp5x_cs35l41_hw_params(struct snd_pcm_substream *substream,
 				   struct snd_pcm_hw_params *params)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct snd_soc_card *card = rtd->card;
 	struct snd_soc_dai *codec_dai;
 	int ret, i;
 	unsigned int num_codecs = rtd->num_codecs;
 	unsigned int bclk_val;
 
+	ret = 0;
 	for (i = 0; i < num_codecs; i++) {
 		codec_dai = asoc_rtd_to_codec(rtd, i);
 		if ((strcmp(codec_dai->name, "spi-VLV1776:00") == 0) ||
@@ -242,7 +243,7 @@ SND_SOC_DAILINK_DEF(platform,
 
 static struct snd_soc_dai_link acp5x_dai[] = {
 	{
-		.name = "acp5x-8825-play",
+		.name = "acp5x-8821-play",
 		.stream_name = "Playback/Capture",
 		.dai_fmt = SND_SOC_DAIFMT_I2S  | SND_SOC_DAIFMT_NB_NF |
 			   SND_SOC_DAIFMT_CBC_CFC,
@@ -341,7 +342,6 @@ static struct snd_soc_card acp5x_card = {
 	.num_controls = ARRAY_SIZE(acp5x_8821_controls),
 };
 
-
 static int acp5x_vg_quirk_cb(const struct dmi_system_id *id)
 {
 	acp5x_machine_id = VG_JUPITER;
@@ -371,7 +371,7 @@ static int acp5x_probe(struct platform_device *pdev)
 		return -ENOMEM;
 
 	dmi_check_system(acp5x_vg_quirk_table);
-	switch(acp5x_machine_id) {
+	switch (acp5x_machine_id) {
 	case VG_JUPITER:
 		card = &acp5x_card;
 		acp5x_card.dev = &pdev->dev;

sound/soc/amd/vangogh/acp5x-pcm-dma.c

@@ -281,7 +281,7 @@ static int acp5x_dma_hw_params(struct snd_soc_component *component,
 		return -EINVAL;
 	}
 	size = params_buffer_bytes(params);
-	rtd->dma_addr = substream->dma_buffer.addr;
+	rtd->dma_addr = substream->runtime->dma_addr;
 	rtd->num_pages = (PAGE_ALIGN(size) >> PAGE_SHIFT);
 	config_acp5x_dma(rtd, substream->stream);
 	return 0;
@@ -388,13 +388,11 @@ static int acp5x_audio_probe(struct platform_device *pdev)
 	if (!adata->acp5x_base)
 		return -ENOMEM;
 
-	res = platform_get_resource(pdev, IORESOURCE_IRQ, 0);
+	status = platform_get_irq(pdev, 0);
-	if (!res) {
+	if (status < 0)
-		dev_err(&pdev->dev, "IORESOURCE_IRQ FAILED\n");
+		return status;
-		return -ENODEV;
+	adata->i2s_irq = status;
-	}
 
-	adata->i2s_irq = res->start;
 	dev_set_drvdata(&pdev->dev, adata);
 	status = devm_snd_soc_register_component(&pdev->dev,
 						 &acp5x_i2s_component,
@@ -426,51 +424,51 @@ static int acp5x_audio_remove(struct platform_device *pdev)
 static int __maybe_unused acp5x_pcm_resume(struct device *dev)
 {
 	struct i2s_dev_data *adata;
-	u32 val, reg_val, frmt_val;
+	struct i2s_stream_instance *rtd;
+	u32 val;
 
-	reg_val = 0;
-	frmt_val = 0;
 	adata = dev_get_drvdata(dev);
 
 	if (adata->play_stream && adata->play_stream->runtime) {
-		struct i2s_stream_instance *rtd =
+		rtd = adata->play_stream->runtime->private_data;
-			adata->play_stream->runtime->private_data;
 		config_acp5x_dma(rtd, SNDRV_PCM_STREAM_PLAYBACK);
-		switch (rtd->i2s_instance) {
+		acp_writel((rtd->xfer_resolution  << 3), rtd->acp5x_base + ACP_HSTDM_ITER);
-		case I2S_HS_INSTANCE:
+		if (adata->tdm_mode == TDM_ENABLE) {
-			reg_val = ACP_HSTDM_ITER;
+			acp_writel(adata->tdm_fmt, adata->acp5x_base + ACP_HSTDM_TXFRMT);
-			frmt_val = ACP_HSTDM_TXFRMT;
+			val = acp_readl(adata->acp5x_base + ACP_HSTDM_ITER);
-			break;
+			acp_writel(val | 0x2, adata->acp5x_base + ACP_HSTDM_ITER);
-		case I2S_SP_INSTANCE:
+		}
-		default:
+	}
-			reg_val = ACP_I2STDM_ITER;
+	if (adata->i2ssp_play_stream && adata->i2ssp_play_stream->runtime) {
-			frmt_val = ACP_I2STDM_TXFRMT;
+		rtd = adata->i2ssp_play_stream->runtime->private_data;
+		config_acp5x_dma(rtd, SNDRV_PCM_STREAM_PLAYBACK);
+		acp_writel((rtd->xfer_resolution  << 3), rtd->acp5x_base + ACP_I2STDM_ITER);
+		if (adata->tdm_mode == TDM_ENABLE) {
+			acp_writel(adata->tdm_fmt, adata->acp5x_base + ACP_I2STDM_TXFRMT);
+			val = acp_readl(adata->acp5x_base + ACP_I2STDM_ITER);
+			acp_writel(val | 0x2, adata->acp5x_base + ACP_I2STDM_ITER);
 		}
-		acp_writel((rtd->xfer_resolution  << 3),
-			   rtd->acp5x_base + reg_val);
 	}
 
 	if (adata->capture_stream && adata->capture_stream->runtime) {
-		struct i2s_stream_instance *rtd =
+		rtd = adata->capture_stream->runtime->private_data;
-			adata->capture_stream->runtime->private_data;
 		config_acp5x_dma(rtd, SNDRV_PCM_STREAM_CAPTURE);
-		switch (rtd->i2s_instance) {
+		acp_writel((rtd->xfer_resolution  << 3), rtd->acp5x_base + ACP_HSTDM_IRER);
-		case I2S_HS_INSTANCE:
+		if (adata->tdm_mode == TDM_ENABLE) {
-			reg_val = ACP_HSTDM_IRER;
+			acp_writel(adata->tdm_fmt, adata->acp5x_base + ACP_HSTDM_RXFRMT);
-			frmt_val = ACP_HSTDM_RXFRMT;
+			val = acp_readl(adata->acp5x_base + ACP_HSTDM_IRER);
-			break;
+			acp_writel(val | 0x2, adata->acp5x_base + ACP_HSTDM_IRER);
-		case I2S_SP_INSTANCE:
-		default:
-			reg_val = ACP_I2STDM_IRER;
-			frmt_val = ACP_I2STDM_RXFRMT;
 		}
-		acp_writel((rtd->xfer_resolution  << 3),
-			   rtd->acp5x_base + reg_val);
 	}
-	if (adata->tdm_mode == TDM_ENABLE) {
+	if (adata->i2ssp_capture_stream && adata->i2ssp_capture_stream->runtime) {
-		acp_writel(adata->tdm_fmt, adata->acp5x_base + frmt_val);
+		rtd = adata->i2ssp_capture_stream->runtime->private_data;
-		val = acp_readl(adata->acp5x_base + reg_val);
+		config_acp5x_dma(rtd, SNDRV_PCM_STREAM_CAPTURE);
-		acp_writel(val | 0x2, adata->acp5x_base + reg_val);
+		acp_writel((rtd->xfer_resolution  << 3), rtd->acp5x_base + ACP_I2STDM_IRER);
+		if (adata->tdm_mode == TDM_ENABLE) {
+			acp_writel(adata->tdm_fmt, adata->acp5x_base + ACP_I2STDM_RXFRMT);
+			val = acp_readl(adata->acp5x_base + ACP_I2STDM_IRER);
+			acp_writel(val | 0x2, adata->acp5x_base + ACP_I2STDM_IRER);
+		}
 	}
 	acp_writel(1, adata->acp5x_base + ACP_EXTERNAL_INTR_ENB);
 	return 0;

sound/soc/amd/vangogh/acp5x.h

@@ -85,7 +85,7 @@
 struct i2s_dev_data {
 	bool tdm_mode;
 	bool master_mode;
-	unsigned int i2s_irq;
+	int i2s_irq;
 	u16 i2s_instance;
 	u32 tdm_fmt;
 	void __iomem *acp5x_base;

sound/soc/atmel/Kconfig

@@ -160,4 +160,20 @@ config SND_MCHP_SOC_SPDIFRX
 
 	  This S/PDIF RX driver is compliant with IEC-60958 standard and
 	  includes programmable User Data and Channel Status fields.
+
+config SND_MCHP_SOC_PDMC
+	tristate "Microchip ASoC driver for boards using PDMC"
+	depends on OF && (ARCH_AT91 || COMPILE_TEST)
+	select SND_SOC_GENERIC_DMAENGINE_PCM
+	select REGMAP_MMIO
+	help
+	  Say Y or M if you want to add support for Microchip ASoC PDMC driver on the
+	  following Microchip platforms:
+	  - sama7g5
+
+	  The Pulse Density Microphone Controller (PDMC) interfaces up to 4 digital
+	  microphones PDM outputs. It generates a single clock line and samples 1 or
+	  2 data lines. The signal path includes an audio grade programmable
+	  decimation filter and outputs 24-bit audio words.
+
 endif

sound/soc/atmel/Makefile

@@ -7,6 +7,7 @@ snd-soc-atmel-i2s-objs := atmel-i2s.o
 snd-soc-mchp-i2s-mcc-objs := mchp-i2s-mcc.o
 snd-soc-mchp-spdiftx-objs := mchp-spdiftx.o
 snd-soc-mchp-spdifrx-objs := mchp-spdifrx.o
+snd-soc-mchp-pdmc-objs := mchp-pdmc.o
 
 # pdc and dma need to both be built-in if any user of
 # ssc is built-in.
@@ -21,6 +22,7 @@ obj-$(CONFIG_SND_ATMEL_SOC_I2S) += snd-soc-atmel-i2s.o
 obj-$(CONFIG_SND_MCHP_SOC_I2S_MCC) += snd-soc-mchp-i2s-mcc.o
 obj-$(CONFIG_SND_MCHP_SOC_SPDIFTX) += snd-soc-mchp-spdiftx.o
 obj-$(CONFIG_SND_MCHP_SOC_SPDIFRX) += snd-soc-mchp-spdifrx.o
+obj-$(CONFIG_SND_MCHP_SOC_PDMC) += snd-soc-mchp-pdmc.o
 
 # AT91 Machine Support
 snd-soc-sam9g20-wm8731-objs := sam9g20_wm8731.o

sound/soc/atmel/atmel_ssc_dai.c

@@ -280,7 +280,10 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream,
 
 	/* Enable PMC peripheral clock for this SSC */
 	pr_debug("atmel_ssc_dai: Starting clock\n");
-	clk_enable(ssc_p->ssc->clk);
+	ret = clk_enable(ssc_p->ssc->clk);
+	if (ret)
+		return ret;
+
 	ssc_p->mck_rate = clk_get_rate(ssc_p->ssc->clk);
 
 	/* Reset the SSC unless initialized to keep it in a clean state */

sound/soc/atmel/mchp-spdifrx.c

@@ -920,7 +920,7 @@ static int mchp_spdifrx_probe(struct platform_device *pdev)
 
 	err = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0);
 	if (err) {
-		dev_err(&pdev->dev, "failed to register PMC: %d\n", err);
+		dev_err(&pdev->dev, "failed to register PCM: %d\n", err);
 		return err;
 	}
 

sound/soc/atmel/mikroe-proto.c

@@ -115,7 +115,8 @@ static int snd_proto_probe(struct platform_device *pdev)
 	cpu_np = of_parse_phandle(np, "i2s-controller", 0);
 	if (!cpu_np) {
 		dev_err(&pdev->dev, "i2s-controller missing\n");
-		return -EINVAL;
+		ret = -EINVAL;
+		goto put_codec_node;
 	}
 	dai->cpus->of_node = cpu_np;
 	dai->platforms->of_node = cpu_np;
@@ -125,7 +126,8 @@ static int snd_proto_probe(struct platform_device *pdev)
 						       &bitclkmaster, &framemaster);
 	if (bitclkmaster != framemaster) {
 		dev_err(&pdev->dev, "Must be the same bitclock and frame master\n");
-		return -EINVAL;
+		ret = -EINVAL;
+		goto put_cpu_node;
 	}
 	if (bitclkmaster) {
 		if (codec_np == bitclkmaster)
@@ -136,18 +138,20 @@ static int snd_proto_probe(struct platform_device *pdev)
 		dai_fmt |= snd_soc_daifmt_parse_clock_provider_as_flag(np, NULL);
 	}
 
-	of_node_put(bitclkmaster);
+
-	of_node_put(framemaster);
 	dai->dai_fmt = dai_fmt;
-
-	of_node_put(codec_np);
-	of_node_put(cpu_np);
-
 	ret = snd_soc_register_card(&snd_proto);
 	if (ret)
 		dev_err_probe(&pdev->dev, ret,
 			"snd_soc_register_card() failed\n");
 
+
+put_cpu_node:
+	of_node_put(bitclkmaster);
+	of_node_put(framemaster);
+	of_node_put(cpu_np);
+put_codec_node:
+	of_node_put(codec_np);
 	return ret;
 }
 

sound/soc/atmel/sam9g20_wm8731.c

@@ -214,6 +214,7 @@ static int at91sam9g20ek_audio_probe(struct platform_device *pdev)
 	cpu_np = of_parse_phandle(np, "atmel,ssc-controller", 0);
 	if (!cpu_np) {
 		dev_err(&pdev->dev, "dai and pcm info missing\n");
+		of_node_put(codec_np);
 		return -EINVAL;
 	}
 	at91sam9g20ek_dai.cpus->of_node = cpu_np;

sound/soc/atmel/sam9x5_wm8731.c

@@ -142,7 +142,7 @@ static int sam9x5_wm8731_driver_probe(struct platform_device *pdev)
 	if (!cpu_np) {
 		dev_err(&pdev->dev, "atmel,ssc-controller node missing\n");
 		ret = -EINVAL;
-		goto out;
+		goto out_put_codec_np;
 	}
 	dai->cpus->of_node = cpu_np;
 	dai->platforms->of_node = cpu_np;
@@ -153,12 +153,9 @@ static int sam9x5_wm8731_driver_probe(struct platform_device *pdev)
 	if (ret != 0) {
 		dev_err(&pdev->dev, "Failed to set SSC %d for audio: %d\n",
 			ret, priv->ssc_id);
-		goto out;
+		goto out_put_cpu_np;
 	}
 
-	of_node_put(codec_np);
-	of_node_put(cpu_np);
-
 	ret = devm_snd_soc_register_card(&pdev->dev, card);
 	if (ret) {
 		dev_err(&pdev->dev, "Platform device allocation failed\n");
@@ -167,10 +164,14 @@ static int sam9x5_wm8731_driver_probe(struct platform_device *pdev)
 
 	dev_dbg(&pdev->dev, "%s ok\n", __func__);
 
-	return ret;
+	goto out_put_cpu_np;
 
 out_put_audio:
 	atmel_ssc_put_audio(priv->ssc_id);
+out_put_cpu_np:
+	of_node_put(cpu_np);
+out_put_codec_np:
+	of_node_put(codec_np);
 out:
 	return ret;
 }

sound/soc/codecs/Kconfig

@@ -53,6 +53,7 @@ config SND_SOC_ALL_CODECS
 	imply SND_SOC_AK5558
 	imply SND_SOC_ALC5623
 	imply SND_SOC_ALC5632
+	imply SND_SOC_AW8738
 	imply SND_SOC_BT_SCO
 	imply SND_SOC_BD28623
 	imply SND_SOC_CQ0093VC
@@ -244,6 +245,7 @@ config SND_SOC_ALL_CODECS
 	imply SND_SOC_WCD9335
 	imply SND_SOC_WCD934X
 	imply SND_SOC_WCD938X_SDW
+	imply SND_SOC_LPASS_MACRO_COMMON
 	imply SND_SOC_LPASS_RX_MACRO
 	imply SND_SOC_LPASS_TX_MACRO
 	imply SND_SOC_WL1273
@@ -578,6 +580,15 @@ config SND_SOC_ALC5632
 	tristate
 	depends on I2C
 
+config SND_SOC_AW8738
+	tristate "Awinic AW8738 Audio Amplifier"
+	select GPIOLIB
+	help
+	  Enable support for the Awinic AW8738 audio amplifier (or similar).
+	  The driver supports simple audio amplifiers similar to
+	  SND_SOC_SIMPLE_AMPLIFIER, but additionally allows setting the
+	  operation mode using the Awinic-specific one-wire pulse control.
+
 config SND_SOC_BD28623
 	tristate "ROHM BD28623 CODEC"
 	help
@@ -733,6 +744,7 @@ config SND_SOC_CS4349
 
 config SND_SOC_CS47L15
 	tristate
+	depends on MFD_CS47L15
 
 config SND_SOC_CS47L24
 	tristate
@@ -740,15 +752,19 @@ config SND_SOC_CS47L24
 
 config SND_SOC_CS47L35
 	tristate
+	depends on MFD_CS47L35
 
 config SND_SOC_CS47L85
 	tristate
+	depends on MFD_CS47L85
 
 config SND_SOC_CS47L90
 	tristate
+	depends on MFD_CS47L90
 
 config SND_SOC_CS47L92
 	tristate
+	depends on MFD_CS47L92
 
 # Cirrus Logic Quad-Channel ADC
 config SND_SOC_CS53L30
@@ -842,7 +858,6 @@ config SND_SOC_DA9055
 
 config SND_SOC_DMIC
 	tristate "Generic Digital Microphone CODEC"
-	depends on GPIOLIB
 	help
 	  Enable support for the Generic Digital Microphone CODEC.
 	  Select this if your sound card has DMICs.
@@ -1349,7 +1364,6 @@ config SND_SOC_RT9120
 	tristate "Richtek RT9120 Stereo Class-D Amplifier"
 	depends on I2C
 	select REGMAP_I2C
-	select GPIOLIB
 	help
 	  Enable support for Richtek RT9120 20W, stereo, inductor-less,
 	  high-efficiency Class-D audio amplifier.
@@ -1393,11 +1407,10 @@ config SND_SOC_SIGMADSP_REGMAP
 
 config SND_SOC_SIMPLE_AMPLIFIER
 	tristate "Simple Audio Amplifier"
-	select GPIOLIB
 
 config SND_SOC_SIMPLE_MUX
 	tristate "Simple Audio Mux"
-	select GPIOLIB
+	depends on GPIOLIB
 
 config SND_SOC_SPDIF
 	tristate "S/PDIF CODEC"
@@ -1485,6 +1498,15 @@ config SND_SOC_TAS5720
 	  Enable support for Texas Instruments TAS5720L/M high-efficiency mono
 	  Class-D audio power amplifiers.
 
+config SND_SOC_TAS5805M
+	tristate "Texas Instruments TAS5805M speaker amplifier"
+	depends on I2C
+	help
+	  Enable support for Texas Instruments TAS5805M Class-D
+	  amplifiers. This is a speaker amplifier with an integrated
+	  DSP. DSP configuration for each instance needs to be supplied
+	  via a device-tree attribute.
+
 config SND_SOC_TAS6424
 	tristate "Texas Instruments TAS6424 Quad-Channel Audio amplifier"
 	depends on I2C
@@ -1901,7 +1923,6 @@ config SND_SOC_WSA881X
 config SND_SOC_ZL38060
 	tristate "Microsemi ZL38060 Connected Home Audio Processor"
 	depends on SPI_MASTER
-	select GPIOLIB
 	select REGMAP
 	help
 	  Support for ZL38060 Connected Home Audio Processor from Microsemi,
@@ -1915,7 +1936,7 @@ config SND_SOC_LM4857
 
 config SND_SOC_MAX9759
 	tristate "Maxim MAX9759 speaker Amplifier"
-	select GPIOLIB
+	depends on GPIOLIB
 
 config SND_SOC_MAX9768
 	tristate
@@ -1998,6 +2019,9 @@ config SND_SOC_TPA6130A2
 	tristate "Texas Instruments TPA6130A2 headphone amplifier"
 	depends on I2C
 
+config SND_SOC_LPASS_MACRO_COMMON
+        tristate
+
 config SND_SOC_LPASS_WSA_MACRO
 	depends on COMMON_CLK
 	select REGMAP_MMIO
@@ -2006,16 +2030,19 @@ config SND_SOC_LPASS_WSA_MACRO
 config SND_SOC_LPASS_VA_MACRO
 	depends on COMMON_CLK
 	select REGMAP_MMIO
+	select SND_SOC_LPASS_MACRO_COMMON
 	tristate "Qualcomm VA Macro in LPASS(Low Power Audio SubSystem)"
 
 config SND_SOC_LPASS_RX_MACRO
 	depends on COMMON_CLK
 	select REGMAP_MMIO
+	select SND_SOC_LPASS_MACRO_COMMON
 	tristate "Qualcomm RX Macro in LPASS(Low Power Audio SubSystem)"
 
 config SND_SOC_LPASS_TX_MACRO
 	depends on COMMON_CLK
 	select REGMAP_MMIO
+	select SND_SOC_LPASS_MACRO_COMMON
 	tristate "Qualcomm TX Macro in LPASS(Low Power Audio SubSystem)"
 
 endmenu

sound/soc/codecs/Makefile

@@ -45,6 +45,7 @@ snd-soc-ak4671-objs := ak4671.o
 snd-soc-ak5386-objs := ak5386.o
 snd-soc-ak5558-objs := ak5558.o
 snd-soc-arizona-objs := arizona.o arizona-jack.o
+snd-soc-aw8738-objs := aw8738.o
 snd-soc-bd28623-objs := bd28623.o
 snd-soc-bt-sco-objs := bt-sco.o
 snd-soc-cpcap-objs := cpcap.o
@@ -112,6 +113,7 @@ snd-soc-l3-objs := l3.o
 snd-soc-lm4857-objs := lm4857.o
 snd-soc-lm49453-objs := lm49453.o
 snd-soc-lochnagar-sc-objs := lochnagar-sc.o
+snd-soc-lpass-macro-common-objs := lpass-macro-common.o
 snd-soc-lpass-rx-macro-objs := lpass-rx-macro.o
 snd-soc-lpass-tx-macro-objs := lpass-tx-macro.o
 snd-soc-lpass-wsa-macro-objs := lpass-wsa-macro.o
@@ -236,6 +238,7 @@ snd-soc-sti-sas-objs := sti-sas.o
 snd-soc-tas5086-objs := tas5086.o
 snd-soc-tas571x-objs := tas571x.o
 snd-soc-tas5720-objs := tas5720.o
+snd-soc-tas5805m-objs := tas5805m.o
 snd-soc-tas6424-objs := tas6424.o
 snd-soc-tda7419-objs := tda7419.o
 snd-soc-tas2770-objs := tas2770.o
@@ -386,6 +389,7 @@ obj-$(CONFIG_SND_SOC_AK5558)	+= snd-soc-ak5558.o
 obj-$(CONFIG_SND_SOC_ALC5623)    += snd-soc-alc5623.o
 obj-$(CONFIG_SND_SOC_ALC5632)	+= snd-soc-alc5632.o
 obj-$(CONFIG_SND_SOC_ARIZONA)	+= snd-soc-arizona.o
+obj-$(CONFIG_SND_SOC_AW8738)	+= snd-soc-aw8738.o
 obj-$(CONFIG_SND_SOC_BD28623)	+= snd-soc-bd28623.o
 obj-$(CONFIG_SND_SOC_BT_SCO)	+= snd-soc-bt-sco.o
 obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o
@@ -574,6 +578,7 @@ obj-$(CONFIG_SND_SOC_TAS2764)	+= snd-soc-tas2764.o
 obj-$(CONFIG_SND_SOC_TAS5086)	+= snd-soc-tas5086.o
 obj-$(CONFIG_SND_SOC_TAS571X)	+= snd-soc-tas571x.o
 obj-$(CONFIG_SND_SOC_TAS5720)	+= snd-soc-tas5720.o
+obj-$(CONFIG_SND_SOC_TAS5805M)	+= snd-soc-tas5805m.o
 obj-$(CONFIG_SND_SOC_TAS6424)	+= snd-soc-tas6424.o
 obj-$(CONFIG_SND_SOC_TDA7419)	+= snd-soc-tda7419.o
 obj-$(CONFIG_SND_SOC_TAS2770) += snd-soc-tas2770.o
@@ -674,6 +679,7 @@ obj-$(CONFIG_SND_SOC_MAX9877)	+= snd-soc-max9877.o
 obj-$(CONFIG_SND_SOC_MAX98504)	+= snd-soc-max98504.o
 obj-$(CONFIG_SND_SOC_SIMPLE_AMPLIFIER)	+= snd-soc-simple-amplifier.o
 obj-$(CONFIG_SND_SOC_TPA6130A2)	+= snd-soc-tpa6130a2.o
+obj-$(CONFIG_SND_SOC_LPASS_MACRO_COMMON)	+= snd-soc-lpass-macro-common.o
 obj-$(CONFIG_SND_SOC_LPASS_WSA_MACRO)	+= snd-soc-lpass-wsa-macro.o
 obj-$(CONFIG_SND_SOC_LPASS_VA_MACRO)	+= snd-soc-lpass-va-macro.o
 obj-$(CONFIG_SND_SOC_LPASS_RX_MACRO)	+= snd-soc-lpass-rx-macro.o

sound/soc/codecs/ak4613.c

@@ -653,15 +653,10 @@ static int ak4613_i2c_probe(struct i2c_client *i2c,
 	struct ak4613_priv *priv;
 
 	regmap_cfg = NULL;
-	if (np) {
+	if (np)
-		const struct of_device_id *of_id;
+		regmap_cfg = of_device_get_match_data(dev);
-
+	else
-		of_id = of_match_device(ak4613_of_match, dev);
-		if (of_id)
-			regmap_cfg = of_id->data;
-	} else {
 		regmap_cfg = (const struct regmap_config *)id->driver_data;
-	}
 
 	if (!regmap_cfg)
 		return -EINVAL;

sound/soc/codecs/aw8738.c

@@ -0,0 +1,104 @@
+// SPDX-License-Identifier: GPL-2.0-only
+
+#include <linux/gpio/consumer.h>
+#include <linux/module.h>
+#include <linux/regulator/consumer.h>
+#include <sound/soc.h>
+
+struct aw8738_priv {
+	struct gpio_desc *gpiod_mode;
+	unsigned int mode;
+};
+
+static int aw8738_drv_event(struct snd_soc_dapm_widget *w,
+			    struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_component *c = snd_soc_dapm_to_component(w->dapm);
+	struct aw8738_priv *aw = snd_soc_component_get_drvdata(c);
+	int i;
+
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		for (i = 0; i < aw->mode; i++) {
+			gpiod_set_value_cansleep(aw->gpiod_mode, 0);
+			udelay(2);
+			gpiod_set_value_cansleep(aw->gpiod_mode, 1);
+			udelay(2);
+		}
+		msleep(40);
+		break;
+	case SND_SOC_DAPM_PRE_PMD:
+		gpiod_set_value_cansleep(aw->gpiod_mode, 0);
+		usleep_range(1000, 2000);
+		break;
+	default:
+		WARN(1, "Unexpected event");
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static const struct snd_soc_dapm_widget aw8738_dapm_widgets[] = {
+	SND_SOC_DAPM_INPUT("IN"),
+	SND_SOC_DAPM_OUT_DRV_E("DRV", SND_SOC_NOPM, 0, 0, NULL, 0, aw8738_drv_event,
+			       SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+	SND_SOC_DAPM_OUTPUT("OUT"),
+};
+
+static const struct snd_soc_dapm_route aw8738_dapm_routes[] = {
+	{ "DRV", NULL, "IN" },
+	{ "OUT", NULL, "DRV" },
+};
+
+static const struct snd_soc_component_driver aw8738_component_driver = {
+	.dapm_widgets = aw8738_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(aw8738_dapm_widgets),
+	.dapm_routes = aw8738_dapm_routes,
+	.num_dapm_routes = ARRAY_SIZE(aw8738_dapm_routes),
+};
+
+static int aw8738_probe(struct platform_device *pdev)
+{
+	struct device *dev = &pdev->dev;
+	struct aw8738_priv *aw;
+	int ret;
+
+	aw = devm_kzalloc(dev, sizeof(*aw), GFP_KERNEL);
+	if (!aw)
+		return -ENOMEM;
+	platform_set_drvdata(pdev, aw);
+
+	aw->gpiod_mode = devm_gpiod_get(dev, "mode", GPIOD_OUT_LOW);
+	if (IS_ERR(aw->gpiod_mode))
+		return dev_err_probe(dev, PTR_ERR(aw->gpiod_mode),
+				     "Failed to get 'mode' gpio");
+
+	ret = device_property_read_u32(dev, "awinic,mode", &aw->mode);
+	if (ret)
+		return -EINVAL;
+
+	return devm_snd_soc_register_component(&pdev->dev,
+					       &aw8738_component_driver,
+					       NULL, 0);
+}
+
+#ifdef CONFIG_OF
+static const struct of_device_id aw8738_of_match[] = {
+	{ .compatible = "awinic,aw8738" },
+	{ }
+};
+MODULE_DEVICE_TABLE(of, aw8738_of_match);
+#endif
+
+static struct platform_driver aw8738_driver = {
+	.probe	= aw8738_probe,
+	.driver = {
+		.name = "aw8738",
+		.of_match_table = of_match_ptr(aw8738_of_match),
+	},
+};
+module_platform_driver(aw8738_driver);
+
+MODULE_DESCRIPTION("Awinic AW8738 Amplifier Driver");
+MODULE_LICENSE("GPL v2");

sound/soc/codecs/bt-sco.c

@@ -13,11 +13,15 @@
 static const struct snd_soc_dapm_widget bt_sco_widgets[] = {
 	SND_SOC_DAPM_INPUT("RX"),
 	SND_SOC_DAPM_OUTPUT("TX"),
+	SND_SOC_DAPM_AIF_IN("BT_SCO_RX", "Playback", 0,
+			    SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("BT_SCO_TX", "Capture", 0,
+			     SND_SOC_NOPM, 0, 0),
 };
 
 static const struct snd_soc_dapm_route bt_sco_routes[] = {
-	{ "Capture", NULL, "RX" },
+	{ "BT_SCO_TX", NULL, "RX" },
-	{ "TX", NULL, "Playback" },
+	{ "TX", NULL, "BT_SCO_RX" },
 };
 
 static struct snd_soc_dai_driver bt_sco_dai[] = {

sound/soc/codecs/cs35l41.c

@@ -1115,9 +1115,7 @@ static const struct snd_soc_component_driver soc_component_dev_cs35l41 = {
 	.set_sysclk = cs35l41_component_set_sysclk,
 };
 
-static int cs35l41_handle_pdata(struct device *dev,
+static int cs35l41_handle_pdata(struct device *dev, struct cs35l41_platform_data *pdata)
-				struct cs35l41_platform_data *pdata,
-				struct cs35l41_private *cs35l41)
 {
 	struct cs35l41_irq_cfg *irq_gpio1_config = &pdata->irq_config1;
 	struct cs35l41_irq_cfg *irq_gpio2_config = &pdata->irq_config2;
@@ -1260,7 +1258,7 @@ int cs35l41_probe(struct cs35l41_private *cs35l41,
 	if (pdata) {
 		cs35l41->pdata = *pdata;
 	} else {
-		ret = cs35l41_handle_pdata(cs35l41->dev, &cs35l41->pdata, cs35l41);
+		ret = cs35l41_handle_pdata(cs35l41->dev, &cs35l41->pdata);
 		if (ret != 0)
 			return ret;
 	}