Merge tag 'asoc-v6.1' of...

Merge tag 'asoc-v6.1' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: Updates for v6.1

This has been a very quiet release for the core but quite a busy one for
drivers with a big crop of new drivers and lots of feature additions and
fixes to existing ones:

 - A new string helper parse_int_array_user().
 - Improvements to the SOF IPC4 code, especially around trace.
 - Support for AMD Rembrant DSPs, AMD Pink Sardine ACP 6.2, Apple Silcon
   systems, Everest ES8326, Intel Sky Lake and Kaby Lake, MediaTek
   MT8186 support, NXP i.MX8ULP DSPs, Qualcomm SC8280XP, SM8250 and SM8450
   and Texas Instruments SRC4392

There is a conflict with the conversion of I2C remove functions to void
in the cs42l42 driver which is fairly straightforward to resolve but
should be highlighted to Linus.

Documentation/devicetree/bindings/dsp/fsl,dsp.yaml

@@ -20,6 +20,7 @@ properties:
       - fsl,imx8qxp-dsp
       - fsl,imx8qm-dsp
       - fsl,imx8mp-dsp
+      - fsl,imx8ulp-dsp
       - fsl,imx8qxp-hifi4
       - fsl,imx8qm-hifi4
       - fsl,imx8mp-hifi4

Documentation/devicetree/bindings/soc/qcom/qcom,apr.yaml

@@ -20,6 +20,9 @@ properties:
       - qcom,apr-v2
       - qcom,gpr
 
+  power-domains:
+    maxItems: 1
+
   qcom,apr-domain:
     $ref: /schemas/types.yaml#/definitions/uint32
     enum: [1, 2, 3, 4, 5, 6, 7]
@@ -52,6 +55,26 @@ properties:
         2 = Audio DSP Domain
         3 = Application Processor Domain
 
+  qcom,glink-channels:
+    $ref: /schemas/types.yaml#/definitions/string-array
+    description: Channel name used for the communication
+    items:
+      - const: apr_audio_svc
+
+  qcom,intents:
+    $ref: /schemas/types.yaml#/definitions/uint32-array
+    description:
+      List of (size, amount) pairs describing what intents should be
+      preallocated for this virtual channel. This can be used to tweak the
+      default intents available for the channel to meet expectations of the
+      remote.
+
+  qcom,smd-channels:
+    $ref: /schemas/types.yaml#/definitions/string-array
+    description: Channel name used for the communication
+    items:
+      - const: apr_audio_svc
+
   '#address-cells':
     const: 1
 
@@ -97,6 +120,26 @@ patternProperties:
             3 = AMDB Service.
             4 = Voice processing manager.
 
+      clock-controller:
+        $ref: /schemas/sound/qcom,q6dsp-lpass-clocks.yaml#
+        description: Qualcomm DSP LPASS clock controller
+        unevaluatedProperties: false
+
+      dais:
+        type: object
+        oneOf:
+          - $ref: /schemas/sound/qcom,q6apm-dai.yaml#
+          - $ref: /schemas/sound/qcom,q6dsp-lpass-ports.yaml#
+          - $ref: /schemas/sound/qcom,q6asm-dais.yaml#
+        unevaluatedProperties: false
+        description: Qualcomm DSP audio ports
+
+      routing:
+        type: object
+        $ref: /schemas/sound/qcom,q6adm-routing.yaml#
+        unevaluatedProperties: false
+        description: Qualcomm DSP LPASS audio routing
+
       qcom,protection-domain:
         $ref: /schemas/types.yaml#/definitions/string-array
         description: protection domain service name and path for apr service
@@ -107,17 +150,44 @@ patternProperties:
           "tms/servreg", "msm/modem/wlan_pd".
           "tms/servreg", "msm/slpi/sensor_pd".
 
-      '#address-cells':
-        const: 1
+    allOf:
+      - if:
+          properties:
+            compatible:
+              enum:
+                - qcom,q6afe
+        then:
+          properties:
+            dais:
+              properties:
+                compatible:
+                  const: qcom,q6afe-dais
 
-      '#size-cells':
-        const: 0
+      - if:
+          properties:
+            compatible:
+              enum:
+                - qcom,q6apm
+        then:
+          properties:
+            dais:
+              properties:
+                compatible:
+                  enum:
+                    - qcom,q6apm-dais
+                    - qcom,q6apm-lpass-dais
 
-    patternProperties:
-      "^.*@[0-9a-f]+$":
-        type: object
-        description:
-          Service based devices like clock controllers or digital audio interfaces.
+      - if:
+          properties:
+            compatible:
+              enum:
+                - qcom,q6asm
+        then:
+          properties:
+            dais:
+              properties:
+                compatible:
+                  const: qcom,q6asm-dais
 
     additionalProperties: false
 
@@ -125,6 +195,30 @@ required:
   - compatible
   - qcom,domain
 
+allOf:
+  - if:
+      properties:
+        compatible:
+          enum:
+            - qcom,gpr
+    then:
+      properties:
+        power-domains: false
+
+  - if:
+      required:
+        - qcom,glink-channels
+    then:
+      properties:
+        qcom,smd-channels: false
+
+  - if:
+      required:
+        - qcom,smd-channels
+    then:
+      properties:
+        qcom,glink-channels: false
+
 additionalProperties: false
 
 examples:

Documentation/devicetree/bindings/sound/adi,max98396.yaml

@@ -78,6 +78,40 @@ properties:
       interleaved on a single output channel.
     type: boolean
 
+  adi,dmon-stuck-enable:
+    description:
+      Enables the "data monitor stuck" feature. Once the data monitor is
+      enabled, it actively monitors the selected input data (from DIN) to the
+      speaker amplifier. Once a data error is detected, the data monitor
+      automatically places the device into software shutdown.
+    type: boolean
+
+  adi,dmon-stuck-threshold-bits:
+    description:
+      Sets the threshold for the "data monitor stuck" feature, in bits.
+    enum: [9, 11, 13, 15]
+    default: 15
+
+  adi,dmon-magnitude-enable:
+    description:
+      Enables the "data monitor magnitude" feature. Once the data monitor is
+      enabled, it actively monitors the selected input data (from DIN) to the
+      speaker amplifier. Once a data error is detected, the data monitor
+      automatically places the device into software shutdown.
+    type: boolean
+
+  adi,dmon-magnitude-threshold-bits:
+    description:
+      Sets the threshold for the "data monitor magnitude" feature, in bits.
+    enum: [2, 3, 4, 5]
+    default: 5
+
+  adi,dmon-duration-ms:
+    description:
+      Sets the duration for the "data monitor" feature, in milliseconds.
+    enum: [64, 256, 1024, 4096]
+    default: 64
+
   reset-gpios:
     maxItems: 1
 

Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml

+# SPDX-License-Identifier: (GPL-2.0+ OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/allwinner,sun50i-h6-dmic.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Allwinner H6 DMIC
+
+maintainers:
+  - Ban Tao <fengzheng923@gmail.com>
+
+properties:
+  compatible:
+    const: allwinner,sun50i-h6-dmic
+
+  "#sound-dai-cells":
+    const: 0
+
+  reg:
+    maxItems: 1
+
+  interrupts:
+    maxItems: 1
+
+  clocks:
+    items:
+      - description: Bus Clock
+      - description: Module Clock
+
+  clock-names:
+    items:
+      - const: bus
+      - const: mod
+
+  dmas:
+    items:
+      - description: RX DMA Channel
+
+  dma-names:
+    items:
+      - const: rx
+
+  resets:
+    maxItems: 1
+
+required:
+  - "#sound-dai-cells"
+  - compatible
+  - reg
+  - interrupts
+  - clocks
+  - clock-names
+  - dmas
+  - dma-names
+  - resets
+
+additionalProperties: false
+
+examples:
+  - |
+    #include <dt-bindings/interrupt-controller/arm-gic.h>
+    #include <dt-bindings/interrupt-controller/irq.h>
+
+    #include <dt-bindings/clock/sun50i-h6-ccu.h>
+    #include <dt-bindings/reset/sun50i-h6-ccu.h>
+
+    dmic: dmic@5095000 {
+      #sound-dai-cells = <0>;
+      compatible = "allwinner,sun50i-h6-dmic";
+      reg = <0x05095000 0x400>;
+      interrupts = <GIC_SPI 22 IRQ_TYPE_LEVEL_HIGH>;
+      clocks = <&ccu CLK_BUS_DMIC>, <&ccu CLK_DMIC>;
+      clock-names = "bus", "mod";
+      dmas = <&dma 7>;
+      dma-names = "rx";
+      resets = <&ccu RST_BUS_DMIC>;
+    };
+
+...

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

@@ -40,6 +40,7 @@ properties:
 patternProperties:
   "^dai-link-[0-9]+$":
     type: object
+    additionalProperties: false
     description: |-
       dai-link child nodes:
         Container for dai-link level properties and the CODEC sub-nodes.
@@ -63,6 +64,7 @@ patternProperties:
     patternProperties:
       "^codec-[0-9]+$":
         type: object
+        additionalProperties: false
         description: |-
           Codecs:
           dai-link representing backend links should have at least one subnode.

Documentation/devicetree/bindings/sound/apple,mca.yaml

+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/apple,mca.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Apple MCA I2S transceiver
+
+description: |
+  MCA is an I2S transceiver peripheral found on M1 and other Apple chips. It is
+  composed of a number of identical clusters which can operate independently
+  or in an interlinked fashion. Up to 6 clusters have been seen on an MCA.
+
+maintainers:
+  - Martin Povišer <povik+lin@cutebit.org>
+
+properties:
+  compatible:
+    items:
+      - enum:
+          - apple,t6000-mca
+          - apple,t8103-mca
+      - const: apple,mca
+
+  reg:
+    items:
+      - description: Register region of the MCA clusters proper
+      - description: Register region of the DMA glue and its FIFOs
+
+  interrupts:
+    minItems: 4
+    maxItems: 6
+    description:
+      One interrupt per each cluster
+
+  '#address-cells':
+    const: 1
+
+  '#size-cells':
+    const: 0
+
+  dmas:
+    minItems: 16
+    maxItems: 24
+    description:
+      DMA channels corresponding to the SERDES units in the peripheral. They are
+      listed in groups of four per cluster, and within the group they are given
+      as associated to the TXA, RXA, TXB, RXB units.
+
+  dma-names:
+    minItems: 16
+    items:
+      - const: tx0a
+      - const: rx0a
+      - const: tx0b
+      - const: rx0b
+      - const: tx1a
+      - const: rx1a
+      - const: tx1b
+      - const: rx1b
+      - const: tx2a
+      - const: rx2a
+      - const: tx2b
+      - const: rx2b
+      - const: tx3a
+      - const: rx3a
+      - const: tx3b
+      - const: rx3b
+      - const: tx4a
+      - const: rx4a
+      - const: tx4b
+      - const: rx4b
+      - const: tx5a
+      - const: rx5a
+      - const: tx5b
+      - const: rx5b
+    description: |
+      Names for the DMA channels: 'tx'/'rx', then cluster number, then 'a'/'b'
+      based on the associated SERDES unit.
+
+  clocks:
+    minItems: 4
+    maxItems: 6
+    description:
+      Clusters' input reference clock.
+
+  resets:
+    maxItems: 1
+
+  power-domains:
+    minItems: 5
+    maxItems: 7
+    description:
+      First a general power domain for register access, then the power
+      domains of individual clusters for their operation.
+
+  '#sound-dai-cells':
+    const: 1
+
+required:
+  - compatible
+  - reg
+  - dmas
+  - dma-names
+  - clocks
+  - power-domains
+  - '#sound-dai-cells'
+
+additionalProperties: false
+
+examples:
+  - |
+    mca: i2s@9b600000 {
+      compatible = "apple,t6000-mca", "apple,mca";
+      reg = <0x9b600000 0x10000>,
+            <0x9b200000 0x20000>;
+
+      clocks = <&nco 0>, <&nco 1>, <&nco 2>, <&nco 3>;
+      power-domains = <&ps_audio_p>, <&ps_mca0>, <&ps_mca1>,
+                      <&ps_mca2>, <&ps_mca3>;
+      dmas = <&admac 0>, <&admac 1>, <&admac 2>, <&admac 3>,
+             <&admac 4>, <&admac 5>, <&admac 6>, <&admac 7>,
+             <&admac 8>, <&admac 9>, <&admac 10>, <&admac 11>,
+             <&admac 12>, <&admac 13>, <&admac 14>, <&admac 15>;
+      dma-names = "tx0a", "rx0a", "tx0b", "rx0b",
+                  "tx1a", "rx1a", "tx1b", "rx1b",
+                  "tx2a", "rx2a", "tx2b", "rx2b",
+                  "tx3a", "rx3a", "tx3b", "rx3b";
+
+      #sound-dai-cells = <1>;
+    };

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

@@ -19,14 +19,17 @@ properties:
     description: "device name prefix"
     $ref: /schemas/types.yaml#/definitions/string
   convert-rate:
-    description: CPU to Codec rate convert.
-    $ref: /schemas/types.yaml#/definitions/uint32
+    $ref: "/schemas/sound/dai-params.yaml#/$defs/dai-sample-rate"
   convert-channels:
-    description: CPU to Codec rate channels.
-    $ref: /schemas/types.yaml#/definitions/uint32
+    $ref: "/schemas/sound/dai-params.yaml#/$defs/dai-channels"
+  convert-sample-format:
+    $ref: "/schemas/sound/dai-params.yaml#/$defs/dai-sample-format"
+
 patternProperties:
   "^endpoint(@[0-9a-f]+)?":
     $ref: /schemas/graph.yaml#/$defs/endpoint-base
+    unevaluatedProperties: false
+
     properties:
       mclk-fs:
         description: |
@@ -65,12 +68,18 @@ patternProperties:
             - msb
             - lsb
       convert-rate:
-        description: CPU to Codec rate convert.
-        $ref: /schemas/types.yaml#/definitions/uint32
+        $ref: "/schemas/sound/dai-params.yaml#/$defs/dai-sample-rate"
       convert-channels:
-        description: CPU to Codec rate channels.
-        $ref: /schemas/types.yaml#/definitions/uint32
+        $ref: "/schemas/sound/dai-params.yaml#/$defs/dai-channels"
+      convert-sample-format:
+        $ref: "/schemas/sound/dai-params.yaml#/$defs/dai-sample-format"
 
+      dai-tdm-slot-num:
+        description: Number of slots in use.
+        $ref: /schemas/types.yaml#/definitions/uint32
+      dai-tdm-slot-width:
+        description: Width in bits for each slot.
+        $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

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

@@ -27,11 +27,12 @@ properties:
     description: User specified audio sound widgets.
     $ref: /schemas/types.yaml#/definitions/non-unique-string-array
   convert-rate:
-    description: CPU to Codec rate convert.
-    $ref: /schemas/types.yaml#/definitions/uint32
+    $ref: "/schemas/sound/dai-params.yaml#/$defs/dai-sample-rate"
   convert-channels:
-    description: CPU to Codec rate channels.
-    $ref: /schemas/types.yaml#/definitions/uint32
+    $ref: "/schemas/sound/dai-params.yaml#/$defs/dai-channels"
+  convert-sample-format:
+    $ref: "/schemas/sound/dai-params.yaml#/$defs/dai-sample-format"
+
   pa-gpios:
     maxItems: 1
   hp-det-gpio:

Documentation/devicetree/bindings/sound/cirrus,cs42l42.yaml

@@ -19,6 +19,7 @@ properties:
   compatible:
     enum:
       - cirrus,cs42l42
+      - cirrus,cs42l83
 
   reg:
     description:

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

+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/dai-params.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Digital Audio Interface (DAI) Stream Parameters
+
+maintainers:
+  - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+
+select: false
+
+$defs:
+
+  dai-channels:
+    description: Number of audio channels used by DAI
+    $ref: /schemas/types.yaml#/definitions/uint32
+    minimum: 1
+    maximum: 32
+
+  dai-sample-format:
+    description: Audio sample format used by DAI
+    $ref: /schemas/types.yaml#/definitions/string
+    enum:
+      - s8
+      - s16_le
+      - s24_le
+      - s24_3le
+      - s32_le
+
+  dai-sample-rate:
+    description: Audio sample rate used by DAI
+    $ref: /schemas/types.yaml#/definitions/uint32
+    minimum: 8000
+    maximum: 192000
+
+properties: {}
+
+additionalProperties: true

Documentation/devicetree/bindings/sound/everest,es8326.yaml

+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/everest,es8326.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Everest ES8326 audio CODEC
+
+maintainers:
+  - David Yang <yangxiaohua@everest-semi.com>
+
+properties:
+  compatible:
+    const: everest,es8326
+
+  reg:
+    maxItems: 1
+
+  clocks:
+    items:
+      - description: clock for master clock (MCLK)
+
+  clock-names:
+    items:
+      - const: mclk
+
+  "#sound-dai-cells":
+    const: 0
+
+  everest,jack-pol:
+    $ref: /schemas/types.yaml#/definitions/uint8
+    description: |
+      just the value of reg 57. Bit(3) decides whether the jack polarity is inverted.
+      Bit(2) decides whether the button on the headset is inverted.
+      Bit(1)/(0) decides the mic properity to be OMTP/CTIA or auto.
+    minimum: 0x00
+    maximum: 0x0f
+    default: 0x0f
+
+  everest,mic1-src:
+    $ref: /schemas/types.yaml#/definitions/uint8
+    description:
+      the value of reg 2A when headset plugged.
+    minimum: 0x00
+    maximum: 0x77
+    default: 0x22
+
+  everest,mic2-src:
+    $ref: /schemas/types.yaml#/definitions/uint8
+    description:
+      the value of reg 2A when headset unplugged.
+    minimum: 0x00
+    maximum: 0x77
+    default: 0x44
+
+  everest,jack-detect-inverted:
+    $ref: /schemas/types.yaml#/definitions/flag
+    description:
+      Defined to invert the jack detection.
+
+  everest,interrupt-src:
+    $ref: /schemas/types.yaml#/definitions/uint8
+    description: |
+      value of reg 0x58, Defines the interrupt source.
+      Bit(2) 1 means button press triggers irq, 0 means not.
+      Bit(3) 1 means PIN9 is the irq source for jack detection. When set to 0,
+      bias change on PIN9 do not triggers irq.
+      Bit(4) 1 means PIN27 is the irq source for jack detection.
+      Bit(5) 1 means PIN9 is the irq source after MIC detect.
+      Bit(6) 1 means PIN27 is the irq source after MIC detect.
+    minimum: 0
+    maximum: 0x3c
+    default: 0x08
+
+  everest,interrupt-clk:
+    $ref: /schemas/types.yaml#/definitions/uint8
+    description: |
+     value of reg 0x59, Defines the interrupt output behavior.
+     Bit(0-3) 0 means irq pulse equals 512*internal clock
+       1 means irq pulse equals 1024*internal clock
+       2 means ...
+       7 means irq pulse equals 65536*internal clock
+       8 means irq mutes PA
+       9 means irq mutes PA and DAC output
+     Bit(4) 1 means we invert the interrupt output.
+     Bit(6) 1 means the chip do not detect jack type after button released.
+       0 means the chip detect jack type again after button released.
+    minimum: 0
+    maximum: 0x7f
+    default: 0x45
+
+required:
+  - compatible
+  - reg
+  - "#sound-dai-cells"
+
+additionalProperties: false
+
+examples:
+  - |
+    i2c {
+      #address-cells = <1>;
+      #size-cells = <0>;
+      es8326: codec@19 {
+        compatible = "everest,es8326";
+        reg = <0x19>;
+        clocks = <&clks 10>;
+        clock-names = "mclk";
+        #sound-dai-cells = <0>;
+        everest,mic1-src = [22];
+        everest,mic2-src = [44];
+        everest,jack-pol = [0e];
+        everest,interrupt-src = [08];
+        everest,interrupt-clk = [45];
+      };
+    };

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

+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/fsl,sai.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Freescale Synchronous Audio Interface (SAI).
+
+maintainers:
+  - Shengjiu Wang <shengjiu.wang@nxp.com>
+
+description: |
+  The SAI is based on I2S module that used communicating with audio codecs,
+  which provides a synchronous audio interface that supports fullduplex
+  serial interfaces with frame synchronization such as I2S, AC97, TDM, and
+  codec/DSP interfaces.
+
+properties:
+  compatible:
+    oneOf:
+      - enum:
+          - fsl,vf610-sai
+          - fsl,imx6sx-sai
+          - fsl,imx6ul-sai
+          - fsl,imx7ulp-sai
+          - fsl,imx8mq-sai
+          - fsl,imx8qm-sai
+          - fsl,imx8ulp-sai
+      - items:
+          - enum:
+              - fsl,imx8mm-sai
+              - fsl,imx8mn-sai
+              - fsl,imx8mp-sai
+          - const: fsl,imx8mq-sai
+
+  reg:
+    maxItems: 1
+
+  interrupts:
+    items:
+      - description: receive and transmit interrupt
+
+  dmas:
+    maxItems: 2
+
+  dma-names:
+    maxItems: 2
+
+  clocks:
+    items:
+      - description: The ipg clock for register access
+      - description: master clock source 0 (obsoleted)
+      - description: master clock source 1
+      - description: master clock source 2
+      - description: master clock source 3
+      - description: PLL clock source for 8kHz series
+      - description: PLL clock source for 11kHz series
+    minItems: 4
+
+  clock-names:
+    oneOf:
+      - items:
+          - const: bus
+          - const: mclk0
+          - const: mclk1
+          - const: mclk2
+          - const: mclk3
+          - const: pll8k
+          - const: pll11k
+        minItems: 4
+      - items:
+          - const: bus
+          - const: mclk1
+          - const: mclk2
+          - const: mclk3
+          - const: pll8k
+          - const: pll11k
+        minItems: 4
+
+  lsb-first:
+    description: |
+      Configures whether the LSB or the MSB is transmitted
+      first for the fifo data. If this property is absent,
+      the MSB is transmitted first as default, or the LSB
+      is transmitted first.
+    type: boolean
+
+  big-endian:
+    description: |
+      required if all the SAI registers are big-endian rather than little-endian.
+    type: boolean
+
+  fsl,sai-synchronous-rx:
+    description: |
+      SAI will work in the synchronous mode (sync Tx with Rx) which means
+      both the transmitter and the receiver will send and receive data by
+      following receiver's bit clocks and frame sync clocks.
+    type: boolean
+
+  fsl,sai-asynchronous:
+    description: |
+      SAI will work in the asynchronous mode, which means both transmitter
+      and receiver will send and receive data by following their own bit clocks
+      and frame sync clocks separately.
+      If both fsl,sai-asynchronous and fsl,sai-synchronous-rx are absent, the
+      default synchronous mode (sync Rx with Tx) will be used, which means both
+      transmitter and receiver will send and receive data by following clocks
+      of transmitter.
+    type: boolean
+
+  fsl,dataline:
+    $ref: /schemas/types.yaml#/definitions/uint32-matrix
+    description: |
+      Configure the dataline. It has 3 value for each configuration
+    maxItems: 16
+    items:
+      items:
+        - description: format Default(0), I2S(1) or PDM(2)
+          enum: [0, 1, 2]
+        - description: dataline mask for 'rx'
+        - description: dataline mask for 'tx'
+
+  fsl,sai-mclk-direction-output:
+    description: SAI will output the SAI MCLK clock.
+    type: boolean
+
+  fsl,shared-interrupt:
+    description: Interrupt is shared with other modules.
+    type: boolean
+
+  "#sound-dai-cells":
+    const: 0
+    description: optional, some dts node didn't add it.
+
+allOf:
+  - if:
+      properties:
+        compatible:
+          contains:
+            const: fsl,vf610-sai
+    then:
+      properties:
+        dmas:
+          items:
+            - description: DMA controller phandle and request line for TX
+            - description: DMA controller phandle and request line for RX
+        dma-names:
+          items:
+            - const: tx
+            - const: rx
+    else:
+      properties:
+        dmas:
+          items:
+            - description: DMA controller phandle and request line for RX
+            - description: DMA controller phandle and request line for TX
+        dma-names:
+          items:
+            - const: rx
+            - const: tx
+  - if:
+      required:
+        - fsl,sai-asynchronous
+    then:
+      properties:
+        fsl,sai-synchronous-rx: false
+
+required:
+  - compatible
+  - reg
+  - interrupts
+  - dmas
+  - dma-names
+  - clocks
+  - clock-names
+
+additionalProperties: false
+
+examples:
+  - |
+    #include <dt-bindings/interrupt-controller/arm-gic.h>
+    #include <dt-bindings/clock/vf610-clock.h>
+    sai2: sai@40031000 {
+        compatible = "fsl,vf610-sai";
+        reg = <0x40031000 0x1000>;
+        interrupts = <86 IRQ_TYPE_LEVEL_HIGH>;
+        pinctrl-names = "default";
+        pinctrl-0 = <&pinctrl_sai2_1>;
+        clocks = <&clks VF610_CLK_PLATFORM_BUS>,
+                 <&clks VF610_CLK_SAI2>,
+                 <&clks 0>, <&clks 0>;
+        clock-names = "bus", "mclk1", "mclk2", "mclk3";
+        dma-names = "tx", "rx";
+        dmas = <&edma0 0 21>,
+               <&edma0 0 20>;
+        big-endian;
+        lsb-first;
+    };
+
+  - |
+    #include <dt-bindings/interrupt-controller/arm-gic.h>
+    #include <dt-bindings/clock/imx8mm-clock.h>
+    sai1: sai@30010000 {
+        compatible = "fsl,imx8mm-sai", "fsl,imx8mq-sai";
+        reg = <0x30010000 0x10000>;
+        interrupts = <GIC_SPI 95 IRQ_TYPE_LEVEL_HIGH>;
+        clocks = <&clk IMX8MM_CLK_SAI1_IPG>,
+                 <&clk IMX8MM_CLK_DUMMY>,
+                 <&clk IMX8MM_CLK_SAI1_ROOT>,
+                 <&clk IMX8MM_CLK_DUMMY>, <&clk IMX8MM_CLK_DUMMY>;
+        clock-names = "bus", "mclk0", "mclk1", "mclk2", "mclk3";
+        dmas = <&sdma2 0 2 0>, <&sdma2 1 2 0>;
+        dma-names = "rx", "tx";
+        fsl,dataline = <1 0xff 0xff 2 0xff 0x11>;
+        #sound-dai-cells = <0>;
+    };

Documentation/devicetree/bindings/sound/fsl-sai.txt

-Freescale Synchronous Audio Interface (SAI).
-
-The SAI is based on I2S module that used communicating with audio codecs,
-which provides a synchronous audio interface that supports fullduplex
-serial interfaces with frame synchronization such as I2S, AC97, TDM, and
-codec/DSP interfaces.
-
-Required properties:
-
-  - compatible		: Compatible list, contains "fsl,vf610-sai",
-			  "fsl,imx6sx-sai", "fsl,imx6ul-sai",
-			  "fsl,imx7ulp-sai", "fsl,imx8mq-sai",
-			  "fsl,imx8qm-sai", "fsl,imx8mm-sai",
-			  "fsl,imx8mn-sai", "fsl,imx8mp-sai", or
-			  "fsl,imx8ulp-sai".
-
-  - reg			: Offset and length of the register set for the device.
-
-  - clocks		: Must contain an entry for each entry in clock-names.
-
-  - clock-names		: Must include the "bus" for register access and
-			  "mclk1", "mclk2", "mclk3" for bit clock and frame
-			  clock providing.
-                          "pll8k", "pll11k" are optional, they are the clock
-                          source for root clock, one is for 8kHz series rates
-                          another one is for 11kHz series rates.
-  - dmas		: Generic dma devicetree binding as described in
-			  Documentation/devicetree/bindings/dma/dma.txt.
-
-  - dma-names		: Two dmas have to be defined, "tx" and "rx".
-
-  - pinctrl-names	: Must contain a "default" entry.
-
-  - pinctrl-NNN		: One property must exist for each entry in
-			  pinctrl-names. See ../pinctrl/pinctrl-bindings.txt
-			  for details of the property values.
-
-  - lsb-first		: Configures whether the LSB or the MSB is transmitted
-			  first for the fifo data. If this property is absent,
-			  the MSB is transmitted first as default, or the LSB
-			  is transmitted first.
-
-  - fsl,sai-synchronous-rx: This is a boolean property. If present, indicating
-			  that SAI will work in the synchronous mode (sync Tx
-			  with Rx) which means both the transmitter and the
-			  receiver will send and receive data by following
-			  receiver's bit clocks and frame sync clocks.
-
-  - fsl,sai-asynchronous: This is a boolean property. If present, indicating
-			  that SAI will work in the asynchronous mode, which
-			  means both transmitter and receiver will send and
-			  receive data by following their own bit clocks and
-			  frame sync clocks separately.
-
-  - fsl,dataline        : configure the dataline. it has 3 value for each configuration
-                          first one means the type: I2S(1) or PDM(2)
-                          second one is dataline mask for 'rx'
-                          third one is dataline mask for 'tx'.
-                          for example: fsl,dataline = <1 0xff 0xff 2 0xff 0x11>;
-                          it means I2S type rx mask is 0xff, tx mask is 0xff, PDM type
-                          rx mask is 0xff, tx mask is 0x11 (dataline 1 and 4 enabled).
-
-Optional properties:
-
-  - big-endian		: Boolean property, required if all the SAI
-			  registers are big-endian rather than little-endian.
-
-Optional properties (for mx6ul):
-
-  - fsl,sai-mclk-direction-output: This is a boolean property. If present,
-			 indicates that SAI will output the SAI MCLK clock.
-
-Note:
-- If both fsl,sai-asynchronous and fsl,sai-synchronous-rx are absent, the
-  default synchronous mode (sync Rx with Tx) will be used, which means both
-  transmitter and receiver will send and receive data by following clocks
-  of transmitter.
-- fsl,sai-asynchronous and fsl,sai-synchronous-rx are exclusive.
-
-Example:
-sai2: sai@40031000 {
-	      compatible = "fsl,vf610-sai";
-	      reg = <0x40031000 0x1000>;
-	      pinctrl-names = "default";
-	      pinctrl-0 = <&pinctrl_sai2_1>;
-	      clocks = <&clks VF610_CLK_PLATFORM_BUS>,
-		     <&clks VF610_CLK_SAI2>,
-		     <&clks 0>, <&clks 0>;
-	      clock-names = "bus", "mclk1", "mclk2", "mclk3";
-	      dma-names = "tx", "rx";
-	      dmas = <&edma0 0 VF610_EDMA_MUXID0_SAI2_TX>,
-		   <&edma0 0 VF610_EDMA_MUXID0_SAI2_RX>;
-	      big-endian;
-	      lsb-first;
-};

Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml

@@ -61,6 +61,8 @@ patternProperties:
       cpu:
         description: Holds subnode which indicates cpu dai.
         type: object
+        additionalProperties: false
+
         properties:
           sound-dai:
             maxItems: 1
@@ -68,6 +70,8 @@ patternProperties:
       codec:
         description: Holds subnode which indicates codec dai.
         type: object
+        additionalProperties: false
+
         properties:
           sound-dai:
             maxItems: 1

Documentation/devicetree/bindings/sound/imx-audio-card.yaml

@@ -58,6 +58,7 @@ patternProperties:
       cpu:
         description: Holds subnode which indicates cpu dai.
         type: object
+        additionalProperties: false
         properties:
           sound-dai:
             maxItems: 1
@@ -65,6 +66,7 @@ patternProperties:
       codec:
         description: Holds subnode which indicates codec dai.
         type: object
+        additionalProperties: false
         properties:
           sound-dai:
             minItems: 1

Documentation/devicetree/bindings/sound/mt8186-mt6366-da7219-max98357.yaml

@@ -43,6 +43,16 @@ properties:
     required:
       - sound-dai
 
+  mediatek,adsp:
+    $ref: /schemas/types.yaml#/definitions/phandle
+    description: The phandle of MT8186 ADSP platform.
+
+  mediatek,dai-link:
+    $ref: /schemas/types.yaml#/definitions/string-array
+    description:
+      A list of the desired dai-links in the sound card. Each entry is a
+      name defined in the machine driver.
+
 additionalProperties: false
 
 required:

Documentation/devicetree/bindings/sound/mt8186-mt6366-rt1019-rt5682s.yaml

@@ -43,6 +43,16 @@ properties:
     required:
       - sound-dai
 
+  mediatek,adsp:
+    $ref: /schemas/types.yaml#/definitions/phandle
+    description: The phandle of MT8186 ADSP platform.
+
+  mediatek,dai-link:
+    $ref: /schemas/types.yaml#/definitions/string-array
+    description:
+      A list of the desired dai-links in the sound card. Each entry is a
+      name defined in the machine driver.
+
 additionalProperties: false
 
 required:

Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml

@@ -30,6 +30,8 @@ properties:
 
   headset-codec:
     type: object
+    additionalProperties: false
+
     properties:
       sound-dai:
         $ref: /schemas/types.yaml#/definitions/phandle
@@ -38,6 +40,8 @@ properties:
 
   speaker-codecs:
     type: object
+    additionalProperties: false
+
     properties:
       sound-dai:
         minItems: 1

Documentation/devicetree/bindings/sound/nau8825.txt

@@ -71,6 +71,9 @@ Optional properties:
 
   - nuvoton,crosstalk-enable: make crosstalk function enable if set.
 
+  - nuvoton,adcout-drive-strong: make the drive strength of ADCOUT IO PIN strong if set.
+      Otherwise, the drive keeps normal strength.
+
   - clocks: list of phandle and clock specifier pairs according to common clock bindings for the
       clocks described in clock-names
   - clock-names: should include "mclk" for the MCLK master clock

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

@@ -14,6 +14,8 @@ properties:
     enum:
       - qcom,sc7280-lpass-rx-macro
       - qcom,sm8250-lpass-rx-macro
+      - qcom,sm8450-lpass-rx-macro
+      - qcom,sc8280xp-lpass-rx-macro
 
   reg:
     maxItems: 1

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

@@ -14,6 +14,8 @@ properties:
     enum:
       - qcom,sc7280-lpass-tx-macro
       - qcom,sm8250-lpass-tx-macro
+      - qcom,sm8450-lpass-tx-macro
+      - qcom,sc8280xp-lpass-tx-macro
 
   reg:
     maxItems: 1

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

@@ -14,6 +14,8 @@ properties:
     enum:
       - qcom,sc7280-lpass-va-macro
       - qcom,sm8250-lpass-va-macro
+      - qcom,sm8450-lpass-va-macro
+      - qcom,sc8280xp-lpass-va-macro
 
   reg:
     maxItems: 1

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

@@ -14,6 +14,8 @@ properties:
     enum:
       - qcom,sc7280-lpass-wsa-macro
       - qcom,sm8250-lpass-wsa-macro
+      - qcom,sm8450-lpass-wsa-macro
+      - qcom,sc8280xp-lpass-wsa-macro
 
   reg:
     maxItems: 1

Documentation/devicetree/bindings/sound/qcom,q6adm-routing.yaml

+# SPDX-License-Identifier: GPL-2.0 OR BSD-2-Clause
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/qcom,q6adm-routing.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Qualcomm Audio Device Manager (Q6ADM) routing
+
+maintainers:
+  - Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
+  - Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
+
+description:
+  Qualcomm Audio Device Manager (Q6ADM) routing node represents routing
+  specific configuration.
+
+properties:
+  compatible:
+    enum:
+      - qcom,q6adm-routing
+
+  "#sound-dai-cells":
+    const: 0
+
+required:
+  - compatible
+  - "#sound-dai-cells"
+
+additionalProperties: false
+
+examples:
+  - |
+    #include <dt-bindings/soc/qcom,apr.h>
+    #include <dt-bindings/sound/qcom,q6asm.h>
+
+    apr {
+        compatible = "qcom,apr-v2";
+        qcom,domain = <APR_DOMAIN_ADSP>;
+        #address-cells = <1>;
+        #size-cells = <0>;
+
+        service@8 {
+            compatible = "qcom,q6adm";
+            reg = <APR_SVC_ADM>;
+            qcom,protection-domain = "avs/audio", "msm/adsp/audio_pd";
+
+            routing {
+                compatible = "qcom,q6adm-routing";
+                #sound-dai-cells = <0>;
+            };
+        };
+    };

Documentation/devicetree/bindings/sound/qcom,q6adm.txt

-Qualcomm Audio Device Manager (Q6ADM) binding
-
-Q6ADM is one of the APR audio service on Q6DSP.
-Please refer to qcom,apr.txt for details of the coommon apr service bindings
-used by the apr service device.
-
-- but must contain the following property:
-
-- compatible:
-	Usage: required
-	Value type: <stringlist>
-	Definition: must be "qcom,q6adm-v<MAJOR-NUMBER>.<MINOR-NUMBER>".
-		   Or "qcom,q6adm" where the version number can be queried
-		   from DSP.
-		   example "qcom,q6adm-v2.0"
-
-
-= ADM routing
-"routing" subnode of the ADM node represents adm routing specific configuration
-
-- compatible:
-	Usage: required
-	Value type: <stringlist>
-	Definition: must be "qcom,q6adm-routing".
-
-- #sound-dai-cells
-	Usage: required
-	Value type: <u32>
-	Definition: Must be 0
-
-= EXAMPLE
-apr-service@8 {
-	compatible = "qcom,q6adm";
-	reg = <APR_SVC_ADM>;
-	q6routing: routing {
-		compatible = "qcom,q6adm-routing";
-		#sound-dai-cells = <0>;
-	};
-};

Documentation/devicetree/bindings/sound/qcom,q6afe.txt

-Qualcomm Audio Front End (Q6AFE) binding
-
-AFE is one of the APR audio service on Q6DSP
-Please refer to qcom,apr.txt for details of the common apr service bindings
-used by all apr services. Must contain the following properties.
-
-- compatible:
-	Usage: required
-	Value type: <stringlist>
-	Definition: must be "qcom,q6afe-v<MAJOR-NUMBER>.<MINOR-NUMBER>"
-		  Or "qcom,q6afe" where the version number can be queried
-		  from DSP.
-		  example "qcom,q6afe"
-
-= EXAMPLE
-
-apr-service@4 {
-	compatible = "qcom,q6afe";
-	reg = <APR_SVC_AFE>;
-};

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

@@ -16,16 +16,12 @@ properties:
   compatible:
     const: qcom,q6apm-dais
 
-  reg:
-    maxItems: 1
-
   iommus:
     maxItems: 1
 
 required:
   - compatible
   - iommus
-  - reg
 
 additionalProperties: false
 
@@ -37,17 +33,14 @@ examples:
         #address-cells = <1>;
         #size-cells = <0>;
         qcom,domain = <GPR_DOMAIN_ID_ADSP>;
+
         service@1 {
             compatible = "qcom,q6apm";
             reg = <1>;
 
-          #address-cells = <1>;
-          #size-cells = <0>;
-
-          apm-dai@1 {
+            dais {
                 compatible = "qcom,q6apm-dais";
                 iommus = <&apps_smmu 0x1801 0x0>;
-            reg = <1>;
             };
         };
     };

Documentation/devicetree/bindings/sound/qcom,q6asm-dais.yaml

+# SPDX-License-Identifier: GPL-2.0 OR BSD-2-Clause
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/qcom,q6asm-dais.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Qualcomm Audio Stream Manager (Q6ASM)
+
+maintainers:
+  - Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
+  - Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
+
+description:
+  Q6ASM is one of the APR audio services on Q6DSP. Each of its subnodes
+  represent a dai with board specific configuration.
+
+properties:
+  compatible:
+    enum:
+      - qcom,q6asm-dais
+
+  iommus:
+    maxItems: 1
+
+  "#sound-dai-cells":
+    const: 1
+
+  "#address-cells":
+    const: 1
+
+  "#size-cells":
+    const: 0
+
+patternProperties:
+  "^dai@[0-9]+$":
+    type: object
+    description:
+      Q6ASM Digital Audio Interface
+
+    properties:
+      reg:
+        maxItems: 1
+
+      direction:
+        $ref: /schemas/types.yaml#/definitions/uint32
+        enum: [0, 1, 2]
+        description: |
+          The direction of the dai stream::
+           - Q6ASM_DAI_TX_RX (0) for both tx and rx
+           - Q6ASM_DAI_TX (1) for only tx (Capture/Encode)
+           - Q6ASM_DAI_RX (2) for only rx (Playback/Decode)
+
+      is-compress-dai:
+        type: boolean
+        description:
+          Compress offload dai.
+
+    dependencies:
+      is-compress-dai: ["direction"]
+
+    required:
+      - reg
+
+    additionalProperties: false
+
+required:
+  - compatible
+  - "#sound-dai-cells"
+  - "#address-cells"
+  - "#size-cells"
+
+additionalProperties: false
+
+examples:
+  - |
+    #include <dt-bindings/soc/qcom,apr.h>
+    #include <dt-bindings/sound/qcom,q6asm.h>
+
+    apr {
+        compatible = "qcom,apr-v2";
+        qcom,domain = <APR_DOMAIN_ADSP>;
+        #address-cells = <1>;
+        #size-cells = <0>;
+
+        service@7 {
+            compatible = "qcom,q6asm";
+            reg = <APR_SVC_ASM>;
+            qcom,protection-domain = "avs/audio", "msm/adsp/audio_pd";
+
+            dais {
+                compatible = "qcom,q6asm-dais";
+                iommus = <&apps_smmu 0x1821 0x0>;
+                #address-cells = <1>;
+                #size-cells = <0>;
+                #sound-dai-cells = <1>;
+
+                dai@0 {
+                    reg = <0>;
+                };
+
+                dai@1 {
+                    reg = <1>;
+                };
+
+                dai@2 {
+                    reg = <2>;
+                    is-compress-dai;
+                    direction = <1>;
+                };
+            };
+        };
+    };

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

-Qualcomm Audio Stream Manager (Q6ASM) binding
-
-Q6ASM is one of the APR audio service on Q6DSP.
-Please refer to qcom,apr.txt for details of the common apr service bindings
-used by the apr service device.
-
-- but must contain the following property:
-
-- compatible:
-	Usage: required
-	Value type: <stringlist>
-	Definition: must be "qcom,q6asm-v<MAJOR-NUMBER>.<MINOR-NUMBER>".
-		    Or "qcom,q6asm" where the version number can be queried
-		    from DSP.
-		    example "qcom,q6asm-v2.0"
-
-= ASM DAIs (Digital Audio Interface)
-"dais" subnode of the ASM node represents dai specific configuration
-
-- compatible:
-	Usage: required
-	Value type: <stringlist>
-	Definition: must be "qcom,q6asm-dais".
-
-- #sound-dai-cells
-	Usage: required
-	Value type: <u32>
-	Definition: Must be 1
-
-== ASM DAI is subnode of "dais" and represent a dai, it includes board specific
-configuration of each dai. Must contain the following properties.
-
-- reg
-	Usage: required
-	Value type: <u32>
-	Definition: Must be dai id
-
-- direction:
-	Usage: Required for Compress offload dais
-	Value type: <u32>
-	Definition: Specifies the direction of the dai stream
-			Q6ASM_DAI_TX_RX (0) for both tx and rx
-			Q6ASM_DAI_TX (1) for only tx (Capture/Encode)
-			Q6ASM_DAI_RX (2) for only rx (Playback/Decode)
-
-- is-compress-dai:
-	Usage: Required for Compress offload dais
-	Value type: <boolean>
-	Definition: present for Compress offload dais
-
-
-= EXAMPLE
-#include <dt-bindings/sound/qcom,q6asm.h>
-
-apr-service@7 {
-	compatible = "qcom,q6asm";
-	reg = <APR_SVC_ASM>;
-	q6asmdai: dais {
-		compatible = "qcom,q6asm-dais";
-		#address-cells = <1>;
-		#size-cells = <0>;
-		#sound-dai-cells = <1>;
-
-		dai@0 {
-			reg = <0>;
-			direction = <Q6ASM_DAI_RX>;
-			is-compress-dai;
-		};
-	};
-};

Documentation/devicetree/bindings/sound/qcom,q6core.txt

-Qualcomm ADSP Core service binding
-
-Q6CORE is one of the APR audio service on Q6DSP.
-Please refer to qcom,apr.txt for details of the common apr service bindings
-used by the apr service device.
-
-- but must contain the following property:
-
-- compatible:
-	Usage: required
-	Value type: <stringlist>
-	Definition: must be "qcom,q6core-v<MAJOR-NUMBER>.<MINOR-NUMBER>".
-		   Or "qcom,q6core" where the version number can be queried
-		   from DSP.
-		   example "qcom,q6core-v2.0"
-
-= EXAMPLE
-apr-service@3 {
-	compatible = "qcom,q6core";
-	reg = <APR_SVC_ADSP_CORE>;
-};

Documentation/devicetree/bindings/sound/qcom,q6dsp-lpass-clocks.yaml

@@ -18,9 +18,6 @@ properties:
       - qcom,q6afe-clocks
       - qcom,q6prm-lpass-clocks
 
-  reg:
-    maxItems: 1
-
   '#clock-cells':
     const: 2
     description:
@@ -32,7 +29,6 @@ properties:
 
 required:
   - compatible
-  - reg
   - "#clock-cells"
 
 additionalProperties: false
@@ -42,15 +38,18 @@ examples:
     #include <dt-bindings/soc/qcom,apr.h>
     #include <dt-bindings/sound/qcom,q6afe.h>
     apr {
+        compatible = "qcom,apr-v2";
+        qcom,domain = <APR_DOMAIN_ADSP>;
         #address-cells = <1>;
         #size-cells = <0>;
-        apr-service@4 {
+
+        service@4 {
+            compatible = "qcom,q6afe";
             reg = <APR_SVC_AFE>;
-            #address-cells = <1>;
-            #size-cells = <0>;
-            clock-controller@2 {
+            qcom,protection-domain = "avs/audio", "msm/adsp/audio_pd";
+
+            clock-controller {
                 compatible = "qcom,q6afe-clocks";
-              reg = <2>;
                 #clock-cells = <2>;
             };
         };
@@ -63,14 +62,13 @@ examples:
         qcom,domain = <GPR_DOMAIN_ID_ADSP>;
         #address-cells = <1>;
         #size-cells = <0>;
+
         service@2 {
             reg = <GPR_PRM_MODULE_IID>;
             compatible = "qcom,q6prm";
-            #address-cells = <1>;
-            #size-cells = <0>;
-            clock-controller@2 {
+
+            clock-controller {
                 compatible = "qcom,q6prm-lpass-clocks";
-              reg = <2>;
                 #clock-cells = <2>;
             };
         };

Documentation/devicetree/bindings/sound/qcom,q6dsp-lpass-ports.yaml

@@ -18,9 +18,6 @@ properties:
       - qcom,q6afe-dais
       - qcom,q6apm-lpass-dais
 
-  reg:
-    maxItems: 1
-
   '#sound-dai-cells':
     const: 1
 
@@ -145,7 +142,6 @@ patternProperties:
 
 required:
   - compatible
-  - reg
   - "#sound-dai-cells"
   - "#address-cells"
   - "#size-cells"
@@ -157,15 +153,18 @@ examples:
     #include <dt-bindings/soc/qcom,apr.h>
     #include <dt-bindings/sound/qcom,q6afe.h>
     apr {
+        compatible = "qcom,apr-v2";
         #address-cells = <1>;
         #size-cells = <0>;
-        apr-service@4 {
+        qcom,domain = <APR_DOMAIN_ADSP>;
+
+        service@4 {
+            compatible = "qcom,q6afe";
             reg = <APR_SVC_AFE>;
-            #address-cells = <1>;
-            #size-cells = <0>;
-            q6afedai@1 {
+            qcom,protection-domain = "avs/audio", "msm/adsp/audio_pd";
+
+            dais {
                 compatible = "qcom,q6afe-dais";
-              reg = <1>;
                 #address-cells = <1>;
                 #size-cells = <0>;
                 #sound-dai-cells = <1>;
@@ -184,14 +183,13 @@ examples:
         #address-cells = <1>;
         #size-cells = <0>;
         qcom,domain = <GPR_DOMAIN_ID_ADSP>;
+
         service@1 {
             compatible = "qcom,q6apm";
             reg = <GPR_APM_MODULE_IID>;
-            #address-cells = <1>;
-            #size-cells = <0>;
-            q6apmdai@1 {
+
+            dais {
                 compatible = "qcom,q6apm-lpass-dais";
-              reg = <1>;
                 #address-cells = <1>;
                 #size-cells = <0>;
                 #sound-dai-cells = <1>;

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

@@ -20,9 +20,11 @@ properties:
       - qcom,apq8016-sbc-sndcard
       - qcom,db845c-sndcard
       - qcom,msm8916-qdsp6-sndcard
+      - qcom,qrb5165-rb5-sndcard
+      - qcom,sc8280xp-sndcard
       - qcom,sdm845-sndcard
       - qcom,sm8250-sndcard
-      - qcom,qrb5165-rb5-sndcard
+      - qcom,sm8450-sndcard
 
   audio-routing:
     $ref: /schemas/types.yaml#/definitions/non-unique-string-array
@@ -71,6 +73,8 @@ patternProperties:
       cpu:
         description: Holds subnode which indicates cpu dai.
         type: object
+        additionalProperties: false
+
         properties:
           sound-dai:
             maxItems: 1
@@ -78,6 +82,8 @@ patternProperties:
       platform:
         description: Holds subnode which indicates platform dai.
         type: object
+        additionalProperties: false
+
         properties:
           sound-dai:
             maxItems: 1
@@ -85,6 +91,8 @@ patternProperties:
       codec:
         description: Holds subnode which indicates codec dai.
         type: object
+        additionalProperties: false
+
         properties:
           sound-dai:
             minItems: 1

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

@@ -129,6 +129,8 @@ properties:
     patternProperties:
       "^dvc-[0-1]$":
         type: object
+        additionalProperties: false
+
         properties:
           dmas:
             maxItems: 1
@@ -145,7 +147,7 @@ properties:
     patternProperties:
       "^mix-[0-1]$":
         type: object
-        # no properties
+        additionalProperties: false
     additionalProperties: false
 
   rcar_sound,ctu:
@@ -154,7 +156,7 @@ properties:
     patternProperties:
       "^ctu-[0-7]$":
         type: object
-        # no properties
+        additionalProperties: false
     additionalProperties: false
 
   rcar_sound,src:
@@ -163,6 +165,8 @@ properties:
     patternProperties:
       "^src-[0-9]$":
         type: object
+        additionalProperties: false
+
         properties:
           interrupts:
             maxItems: 1
@@ -186,6 +190,8 @@ properties:
     patternProperties:
       "^ssiu-[0-9]+$":
         type: object
+        additionalProperties: false
+
         properties:
           dmas:
             maxItems: 2
@@ -206,6 +212,8 @@ properties:
     patternProperties:
       "^ssi-[0-9]$":
         type: object
+        additionalProperties: false
+
         properties:
           interrupts:
             maxItems: 1
@@ -243,6 +251,8 @@ properties:
     patternProperties:
       "^dai([0-9]+)?$":
         type: object
+        additionalProperties: false
+
         properties:
           playback:
             $ref: /schemas/types.yaml#/definitions/phandle-array

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

@@ -23,6 +23,7 @@ properties:
 
   cpu:
     type: object
+    additionalProperties: false
     properties:
       sound-dai:
         minItems: 2
@@ -34,6 +35,7 @@ properties:
       - sound-dai
 
   codec:
+    additionalProperties: false
     type: object
     properties:
       sound-dai:

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

@@ -19,6 +19,7 @@ properties:
 
   cpu:
     type: object
+    additionalProperties: false
     properties:
       sound-dai:
         maxItems: 1
@@ -28,6 +29,7 @@ properties:
 
   codec:
     type: object
+    additionalProperties: false
     properties:
       sound-dai:
         maxItems: 1

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

@@ -19,6 +19,7 @@ properties:
 
   codec:
     type: object
+    additionalProperties: false
     properties:
       sound-dai:
         description: List of phandles to the CODEC and HDMI IP nodes.
@@ -30,6 +31,7 @@ properties:
 
   cpu:
     type: object
+    additionalProperties: false
     properties:
       sound-dai:
         description: Phandle to the Samsung I2S controller.

Documentation/devicetree/bindings/sound/st,stm32-sai.yaml

@@ -60,6 +60,7 @@ required:
 patternProperties:
   "^audio-controller@[0-9a-f]+$":
     type: object
+    additionalProperties: false
     description:
       Two subnodes corresponding to SAI sub-block instances A et B
       can be defined. Subnode can be omitted for unsused sub-block.
@@ -121,6 +122,10 @@ patternProperties:
         description: Configure the SAI device as master clock provider.
         const: 0
 
+      port:
+        $ref: audio-graph-port.yaml#
+        unevaluatedProperties: false
+
     required:
       - compatible
       - "#sound-dai-cells"

Documentation/devicetree/bindings/sound/ti,src4xxx.yaml

+# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/ti,src4xxx.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Texas Instruments SRC4392 Device Tree Bindings
+
+description: |
+  The SRC4392 is a digital audio codec that can be connected via
+  I2C or SPI. Currently, only I2C bus is supported.
+
+maintainers:
+  - Matt Flax <flatmax@flatmax.com>
+
+allOf:
+  - $ref: name-prefix.yaml#
+
+properties:
+  compatible:
+    const: ti,src4392
+
+  "#sound-dai-cells":
+    const: 0
+
+  reg:
+    maxItems: 1
+
+required:
+  - "#sound-dai-cells"
+  - compatible
+  - reg
+
+additionalProperties: false
+
+examples:
+  - |
+    i2c {
+        #address-cells = <1>;
+        #size-cells = <0>;
+
+        audio-codec@70 {
+            #sound-dai-cells = <0>;
+            compatible = "ti,src4392";
+            reg = <0x70>;
+        };
+    };
+...

Documentation/devicetree/bindings/sound/ti,ts3a227e.yaml

+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/ti,ts3a227e.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Texas Instruments TS3A227E
+  Autonomous Audio Accessory Detection and Configuration Switch
+
+maintainers:
+  - Dylan Reid <dgreid@chromium.org>
+
+description: |
+  The TS3A227E detect headsets of 3-ring and 4-ring standards and
+  switches automatically to route the microphone correctly. It also
+  handles key press detection in accordance with the Android audio
+  headset specification v1.0.
+
+properties:
+  compatible:
+    enum:
+      - ti,ts3a227e
+
+  reg:
+    const: 0x3b
+
+  interrupts:
+    maxItems: 1
+
+  ti,micbias:
+    $ref: /schemas/types.yaml#/definitions/uint32
+    description: Intended MICBIAS voltage (datasheet section 9.6.7).
+    enum:
+      - 0 # 2.1 V
+      - 1 # 2.2 V
+      - 2 # 2.3 V
+      - 3 # 2.4 V
+      - 4 # 2.5 V
+      - 5 # 2.6 V
+      - 6 # 2.7 V
+      - 7 # 2.8 V
+    default: 1
+
+  ti,debounce-release-ms:
+    description: key release debounce time in ms (datasheet section 9.6.7).
+    enum:
+      - 0
+      - 20
+    default: 20
+
+  ti,debounce-press-ms:
+    description: key press debounce time in ms (datasheet section 9.6.7).
+    enum:
+      - 2
+      - 40
+      - 80
+      - 120
+    default: 80
+
+  ti,debounce-insertion-ms:
+    description: headset insertion debounce time in ms (datasheet section 9.6.5).
+    enum:
+      - 2
+      - 30
+      - 60
+      - 90
+      - 120
+      - 150
+      - 1000
+      - 2000
+    default: 90
+
+required:
+  - compatible
+  - reg
+  - interrupts
+
+additionalProperties: false
+
+examples:
+  - |
+    #include <dt-bindings/interrupt-controller/irq.h>
+    i2c {
+        #address-cells = <1>;
+        #size-cells = <0>;
+        codec: audio-controller@3b {
+            compatible = "ti,ts3a227e";
+            reg = <0x3b>;
+            interrupt-parent = <&gpio1>;
+            interrupts = <3 IRQ_TYPE_LEVEL_LOW>;
+        };
+    };
+
+...

Documentation/devicetree/bindings/sound/ts3a227e.txt

-Texas Instruments TS3A227E
-Autonomous Audio Accessory Detection and Configuration Switch
-
-The TS3A227E detect headsets of 3-ring and 4-ring standards and
-switches automatically to route the microphone correctly.  It also
-handles key press detection in accordance with the Android audio
-headset specification v1.0.
-
-Required properties:
-
- - compatible:		Should contain "ti,ts3a227e".
- - reg:			The i2c address. Should contain <0x3b>.
- - interrupts:		Interrupt number for /INT pin from the 227e
-
-Optional properies:
- - ti,micbias:   Intended MICBIAS voltage (datasheet section 9.6.7).
-      Select 0/1/2/3/4/5/6/7 to specify MICBIAS voltage
-      2.1V/2.2V/2.3V/2.4V/2.5V/2.6V/2.7V/2.8V
-      Default value is "1" (2.2V).
-
-Examples:
-
-	i2c {
-		ts3a227e@3b {
-			compatible = "ti,ts3a227e";
-			reg = <0x3b>;
-			interrupt-parent = <&gpio>;
-			interrupts = <3 IRQ_TYPE_LEVEL_LOW>;
-		};
-	};

Documentation/devicetree/bindings/spi/mediatek,spi-mtk-nor.yaml

@@ -85,8 +85,9 @@ examples:
         compatible = "mediatek,mt8173-nor";
         reg = <0 0x1100d000 0 0xe0>;
         interrupts = <1>;
-        clocks = <&pericfg CLK_PERI_SPI>, <&topckgen CLK_TOP_SPINFI_IFR_SEL>;
-        clock-names = "spi", "sf";
+        clocks = <&pericfg CLK_PERI_SPI>, <&topckgen CLK_TOP_SPINFI_IFR_SEL>,
+                 <&pericfg CLK_PERI_NFI>;
+        clock-names = "spi", "sf", "axi";
         #address-cells = <1>;
         #size-cells = <0>;
 

MAINTAINERS

@@ -820,6 +820,13 @@ L:	linux-media@vger.kernel.org
 S:	Maintained
 F:	drivers/staging/media/sunxi/cedrus/
 
+ALLWINNER DMIC DRIVERS
+M:	Ban Tao <fengzheng923@gmail.com>
+L:	alsa-devel@alsa-project.org (moderated for non-subscribers)
+S:	Maintained
+F:	Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml
+F:	sound/soc/sunxi/sun50i-dmic.c
+
 ALPHA PORT
 M:	Richard Henderson <richard.henderson@linaro.org>
 M:	Ivan Kokshaysky <ink@jurassic.park.msu.ru>
@@ -1899,6 +1906,15 @@ F:	include/dt-bindings/pinctrl/apple.h
 F:	include/linux/apple-mailbox.h
 F:	include/linux/soc/apple/*
 
+ARM/APPLE MACHINE SOUND DRIVERS
+M:	Martin Povišer <povik+lin@cutebit.org>
+L:	asahi@lists.linux.dev
+L:	alsa-devel@alsa-project.org (moderated for non-subscribers)
+S:	Maintained
+F:	Documentation/devicetree/bindings/sound/apple,*
+F:	sound/soc/apple/*
+F:	sound/soc/codecs/cs42l83-i2c.c
+
 ARM/ARTPEC MACHINE SUPPORT
 M:	Jesper Nilsson <jesper.nilsson@axis.com>
 M:	Lars Persson <lars.persson@axis.com>
@@ -16614,6 +16630,9 @@ M:	Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
 M:	Banajit Goswami <bgoswami@quicinc.com>
 L:	alsa-devel@alsa-project.org (moderated for non-subscribers)
 S:	Supported
+F:	Documentation/devicetree/bindings/soc/qcom/qcom,apr.yaml
+F:	Documentation/devicetree/bindings/sound/qcom,*
+F:	drivers/soc/qcom/apr.c
 F:	include/dt-bindings/sound/qcom,wcd9335.h
 F:	sound/soc/codecs/lpass-rx-macro.*
 F:	sound/soc/codecs/lpass-tx-macro.*

drivers/soundwire/bus.c

@@ -297,6 +297,38 @@ int sdw_transfer(struct sdw_bus *bus, struct sdw_msg *msg)
 	return ret;
 }
 
+/**
+ * sdw_show_ping_status() - Direct report of PING status, to be used by Peripheral drivers
+ * @bus: SDW bus
+ * @sync_delay: Delay before reading status
+ */
+void sdw_show_ping_status(struct sdw_bus *bus, bool sync_delay)
+{
+	u32 status;
+
+	if (!bus->ops->read_ping_status)
+		return;
+
+	/*
+	 * wait for peripheral to sync if desired. 10-15ms should be more than
+	 * enough in most cases.
+	 */
+	if (sync_delay)
+		usleep_range(10000, 15000);
+
+	mutex_lock(&bus->msg_lock);
+
+	status = bus->ops->read_ping_status(bus);
+
+	mutex_unlock(&bus->msg_lock);
+
+	if (!status)
+		dev_warn(bus->dev, "%s: no peripherals attached\n", __func__);
+	else
+		dev_dbg(bus->dev, "PING status: %#x\n", status);
+}
+EXPORT_SYMBOL(sdw_show_ping_status);
+
 /**
  * sdw_transfer_defer() - Asynchronously transfer message to a SDW Slave device
  * @bus: SDW bus

drivers/soundwire/cadence_master.c

@@ -756,6 +756,14 @@ cdns_reset_page_addr(struct sdw_bus *bus, unsigned int dev_num)
 }
 EXPORT_SYMBOL(cdns_reset_page_addr);
 
+u32 cdns_read_ping_status(struct sdw_bus *bus)
+{
+	struct sdw_cdns *cdns = bus_to_cdns(bus);
+
+	return cdns_readl(cdns, CDNS_MCP_SLAVE_STAT);
+}
+EXPORT_SYMBOL(cdns_read_ping_status);
+
 /*
  * IRQ handling
  */

drivers/soundwire/cadence_master.h

@@ -177,6 +177,8 @@ enum sdw_command_response
 cdns_xfer_msg_defer(struct sdw_bus *bus,
 		    struct sdw_msg *msg, struct sdw_defer *defer);
 
+u32 cdns_read_ping_status(struct sdw_bus *bus);
+
 int cdns_bus_conf(struct sdw_bus *bus, struct sdw_bus_params *params);
 
 int cdns_set_sdw_stream(struct snd_soc_dai *dai,

drivers/soundwire/intel.c

@@ -1262,6 +1262,7 @@ static struct sdw_master_ops sdw_intel_ops = {
 	.set_bus_conf = cdns_bus_conf,
 	.pre_bank_switch = intel_pre_bank_switch,
 	.post_bank_switch = intel_post_bank_switch,
+	.read_ping_status = cdns_read_ping_status,
 };
 
 static int intel_init(struct sdw_intel *sdw)

include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h

@@ -193,6 +193,24 @@
 #define LPASS_CLK_ID_RX_CORE_MCLK	59
 #define LPASS_CLK_ID_RX_CORE_NPL_MCLK	60
 #define LPASS_CLK_ID_VA_CORE_2X_MCLK	61
+/* Clock ID for MCLK for WSA2 core */
+#define LPASS_CLK_ID_WSA2_CORE_MCLK	62
+/* Clock ID for NPL MCLK for WSA2 core */
+#define LPASS_CLK_ID_WSA2_CORE_2X_MCLK	63
+/* Clock ID for RX Core TX MCLK */
+#define LPASS_CLK_ID_RX_CORE_TX_MCLK	64
+/* Clock ID for RX CORE TX 2X MCLK */
+#define LPASS_CLK_ID_RX_CORE_TX_2X_MCLK	65
+/* Clock ID for WSA core TX MCLK */
+#define LPASS_CLK_ID_WSA_CORE_TX_MCLK	66
+/* Clock ID for WSA core TX 2X MCLK */
+#define LPASS_CLK_ID_WSA_CORE_TX_2X_MCLK	67
+/* Clock ID for WSA2 core TX MCLK */
+#define LPASS_CLK_ID_WSA2_CORE_TX_MCLK	68
+/* Clock ID for WSA2 core TX 2X MCLK */
+#define LPASS_CLK_ID_WSA2_CORE_TX_2X_MCLK	69
+/* Clock ID for RX CORE MCLK2 2X  MCLK */
+#define LPASS_CLK_ID_RX_CORE_MCLK2_2X_MCLK	70
 
 #define LPASS_HW_AVTIMER_VOTE		101
 #define LPASS_HW_MACRO_VOTE		102

include/linux/platform_data/ssm2518.h

-/* SPDX-License-Identifier: GPL-2.0-only */
-/*
- * SSM2518 amplifier audio driver
- *
- * Copyright 2013 Analog Devices Inc.
- *  Author: Lars-Peter Clausen <lars@metafoo.de>
- */
-
-#ifndef __LINUX_PLATFORM_DATA_SSM2518_H__
-#define __LINUX_PLATFORM_DATA_SSM2518_H__
-
-/**
- * struct ssm2518_platform_data - Platform data for the ssm2518 driver
- * @enable_gpio: GPIO connected to the nSD pin. Set to -1 if the nSD pin is
- *            hardwired.
- */
-struct ssm2518_platform_data {
-	int enable_gpio;
-};
-
-#endif

include/linux/soundwire/sdw.h

@@ -839,6 +839,8 @@ struct sdw_defer {
  * @set_bus_conf: Set the bus configuration
  * @pre_bank_switch: Callback for pre bank switch
  * @post_bank_switch: Callback for post bank switch
+ * @read_ping_status: Read status from PING frames, reported with two bits per Device.
+ * Bits 31:24 are reserved.
  */
 struct sdw_master_ops {
 	int (*read_prop)(struct sdw_bus *bus);
@@ -855,6 +857,7 @@ struct sdw_master_ops {
 			struct sdw_bus_params *params);
 	int (*pre_bank_switch)(struct sdw_bus *bus);
 	int (*post_bank_switch)(struct sdw_bus *bus);
+	u32 (*read_ping_status)(struct sdw_bus *bus);
 
 };
 
@@ -919,6 +922,8 @@ int sdw_bus_master_add(struct sdw_bus *bus, struct device *parent,
 		       struct fwnode_handle *fwnode);
 void sdw_bus_master_delete(struct sdw_bus *bus);
 
+void sdw_show_ping_status(struct sdw_bus *bus, bool sync_delay);
+
 /**
  * sdw_port_config: Master or Slave Port configuration
  *

include/linux/string_helpers.h

@@ -21,6 +21,8 @@ enum string_size_units {
 void string_get_size(u64 size, u64 blk_size, enum string_size_units units,
 		     char *buf, int len);
 
+int parse_int_array_user(const char __user *from, size_t count, int **array);
+
 #define UNESCAPE_SPACE		BIT(0)
 #define UNESCAPE_OCTAL		BIT(1)
 #define UNESCAPE_HEX		BIT(2)

include/sound/cs42l42.h

@@ -40,6 +40,7 @@
 #define CS42L42_PAGE_30		0x3000
 
 #define CS42L42_CHIP_ID		0x42A42
+#define CS42L83_CHIP_ID		0x42A83
 
 /* Page 0x10 Global Registers */
 #define CS42L42_DEVID_AB		(CS42L42_PAGE_10 + 0x01)

include/sound/intel-nhlt.h

@@ -136,6 +136,8 @@ 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);
 
+int intel_nhlt_ssp_mclk_mask(struct nhlt_acpi_table *nhlt, int ssp_num);
+
 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,
@@ -169,6 +171,11 @@ static inline int intel_nhlt_ssp_endpoint_mask(struct nhlt_acpi_table *nhlt, u8
 	return 0;
 }
 
+static inline int intel_nhlt_ssp_mclk_mask(struct nhlt_acpi_table *nhlt, int ssp_num)
+{
+	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

@@ -39,6 +39,7 @@ struct asoc_simple_dai {
 struct asoc_simple_data {
 	u32 convert_rate;
 	u32 convert_channels;
+	const char *convert_sample_format;
 };
 
 struct asoc_simple_jack {

include/sound/soc-acpi-intel-match.h

@@ -14,7 +14,6 @@
  * these tables are not constants, some fields can be used for
  * pdata or machine ops
  */
-extern struct snd_soc_acpi_mach snd_soc_acpi_intel_haswell_machines[];
 extern struct snd_soc_acpi_mach snd_soc_acpi_intel_broadwell_machines[];
 extern struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_machines[];
 extern struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[];
@@ -30,6 +29,7 @@ extern struct snd_soc_acpi_mach snd_soc_acpi_intel_tgl_machines[];
 extern struct snd_soc_acpi_mach snd_soc_acpi_intel_ehl_machines[];
 extern struct snd_soc_acpi_mach snd_soc_acpi_intel_jsl_machines[];
 extern struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[];
+extern struct snd_soc_acpi_mach snd_soc_acpi_intel_rpl_machines[];
 extern struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_machines[];
 
 extern struct snd_soc_acpi_mach snd_soc_acpi_intel_cnl_sdw_machines[];
@@ -38,6 +38,7 @@ extern struct snd_soc_acpi_mach snd_soc_acpi_intel_cml_sdw_machines[];
 extern struct snd_soc_acpi_mach snd_soc_acpi_intel_icl_sdw_machines[];
 extern struct snd_soc_acpi_mach snd_soc_acpi_intel_tgl_sdw_machines[];
 extern struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_sdw_machines[];
+extern struct snd_soc_acpi_mach snd_soc_acpi_intel_rpl_sdw_machines[];
 extern struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_sdw_machines[];
 
 /*

include/sound/soc.h

@@ -31,31 +31,31 @@
 #define SOC_DOUBLE_VALUE(xreg, shift_left, shift_right, xmax, xinvert, xautodisable) \
 	((unsigned long)&(struct soc_mixer_control) \
 	{.reg = xreg, .rreg = xreg, .shift = shift_left, \
-	.rshift = shift_right, .max = xmax, .platform_max = xmax, \
+	.rshift = shift_right, .max = xmax, \
 	.invert = xinvert, .autodisable = xautodisable})
 #define SOC_DOUBLE_S_VALUE(xreg, shift_left, shift_right, xmin, xmax, xsign_bit, xinvert, xautodisable) \
 	((unsigned long)&(struct soc_mixer_control) \
 	{.reg = xreg, .rreg = xreg, .shift = shift_left, \
-	.rshift = shift_right, .min = xmin, .max = xmax, .platform_max = xmax, \
+	.rshift = shift_right, .min = xmin, .max = xmax, \
 	.sign_bit = xsign_bit, .invert = xinvert, .autodisable = xautodisable})
 #define SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert, xautodisable) \
 	SOC_DOUBLE_VALUE(xreg, xshift, xshift, xmax, xinvert, xautodisable)
 #define SOC_SINGLE_VALUE_EXT(xreg, xmax, xinvert) \
 	((unsigned long)&(struct soc_mixer_control) \
-	{.reg = xreg, .max = xmax, .platform_max = xmax, .invert = xinvert})
+	{.reg = xreg, .max = xmax, .invert = xinvert})
 #define SOC_DOUBLE_R_VALUE(xlreg, xrreg, xshift, xmax, xinvert) \
 	((unsigned long)&(struct soc_mixer_control) \
 	{.reg = xlreg, .rreg = xrreg, .shift = xshift, .rshift = xshift, \
-	.max = xmax, .platform_max = xmax, .invert = xinvert})
+	.max = xmax, .invert = xinvert})
 #define SOC_DOUBLE_R_S_VALUE(xlreg, xrreg, xshift, xmin, xmax, xsign_bit, xinvert) \
 	((unsigned long)&(struct soc_mixer_control) \
 	{.reg = xlreg, .rreg = xrreg, .shift = xshift, .rshift = xshift, \
-	.max = xmax, .min = xmin, .platform_max = xmax, .sign_bit = xsign_bit, \
+	.max = xmax, .min = xmin, .sign_bit = xsign_bit, \
 	.invert = xinvert})
 #define SOC_DOUBLE_R_RANGE_VALUE(xlreg, xrreg, xshift, xmin, xmax, xinvert) \
 	((unsigned long)&(struct soc_mixer_control) \
 	{.reg = xlreg, .rreg = xrreg, .shift = xshift, .rshift = xshift, \
-	.min = xmin, .max = xmax, .platform_max = xmax, .invert = xinvert})
+	.min = xmin, .max = xmax, .invert = xinvert})
 #define SOC_SINGLE(xname, reg, shift, max, invert) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
 	.info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\
@@ -68,7 +68,7 @@
 	.private_value = (unsigned long)&(struct soc_mixer_control) \
 		{.reg = xreg, .rreg = xreg, .shift = xshift, \
 		 .rshift = xshift,  .min = xmin, .max = xmax, \
-		 .platform_max = xmax, .invert = xinvert} }
+		 .invert = xinvert} }
 #define SOC_SINGLE_TLV(xname, reg, shift, max, invert, tlv_array) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
 	.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
@@ -99,7 +99,7 @@
 	.private_value = (unsigned long)&(struct soc_mixer_control) \
 		{.reg = xreg, .rreg = xreg, .shift = xshift, \
 		 .rshift = xshift, .min = xmin, .max = xmax, \
-		 .platform_max = xmax, .invert = xinvert} }
+		 .invert = xinvert} }
 #define SOC_DOUBLE(xname, reg, shift_left, shift_right, max, invert) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
 	.info = snd_soc_info_volsw, .get = snd_soc_get_volsw, \
@@ -199,7 +199,7 @@
 	.put = snd_soc_put_volsw, \
 	.private_value = (unsigned long)&(struct soc_mixer_control) \
 	{.reg = xreg, .rreg = xreg,  \
-	 .min = xmin, .max = xmax, .platform_max = xmax, \
+	 .min = xmin, .max = xmax, \
 	.sign_bit = 7,} }
 #define SOC_DOUBLE_S8_TLV(xname, xreg, xmin, xmax, tlv_array) \
 {	.iface  = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
@@ -273,7 +273,7 @@
 	.private_value = (unsigned long)&(struct soc_mixer_control) \
 		{.reg = xreg, .rreg = xreg, .shift = xshift, \
 		 .rshift = xshift, .min = xmin, .max = xmax, \
-		 .platform_max = xmax, .invert = xinvert} }
+		 .invert = xinvert} }
 #define SOC_DOUBLE_EXT_TLV(xname, xreg, shift_left, shift_right, xmax, xinvert,\
 	 xhandler_get, xhandler_put, tlv_array) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
@@ -1062,7 +1062,8 @@ struct snd_soc_pcm_runtime {
 	unsigned int params_select; /* currently selected param for dai link */
 
 	/* Dynamic PCM BE runtime data */
-	struct snd_soc_dpcm_runtime dpcm[2];
+	struct snd_soc_dpcm_runtime dpcm[SNDRV_PCM_STREAM_LAST + 1];
+	struct snd_soc_dapm_widget *c2c_widget[SNDRV_PCM_STREAM_LAST + 1];
 
 	long pmdown_time;
 
@@ -1078,11 +1079,6 @@ struct snd_soc_pcm_runtime {
 	 *	asoc_rtd_to_codec()
 	 */
 	struct snd_soc_dai **dais;
-	unsigned int num_codecs;
-	unsigned int num_cpus;
-
-	struct snd_soc_dapm_widget *playback_widget;
-	struct snd_soc_dapm_widget *capture_widget;
 
 	struct delayed_work delayed_work;
 	void (*close_delayed_work_func)(struct snd_soc_pcm_runtime *rtd);
@@ -1108,7 +1104,7 @@ struct snd_soc_pcm_runtime {
 };
 /* see soc_new_pcm_runtime()  */
 #define asoc_rtd_to_cpu(rtd, n)   (rtd)->dais[n]
-#define asoc_rtd_to_codec(rtd, n) (rtd)->dais[n + (rtd)->num_cpus]
+#define asoc_rtd_to_codec(rtd, n) (rtd)->dais[n + (rtd)->dai_link->num_cpus]
 #define asoc_substream_to_rtd(substream) \
 	(struct snd_soc_pcm_runtime *)snd_pcm_substream_chip(substream)
 
@@ -1118,15 +1114,15 @@ struct snd_soc_pcm_runtime {
 	     (i)++)
 #define for_each_rtd_cpu_dais(rtd, i, dai)				\
 	for ((i) = 0;							\
-	     ((i) < rtd->num_cpus) && ((dai) = asoc_rtd_to_cpu(rtd, i)); \
+	     ((i) < rtd->dai_link->num_cpus) && ((dai) = asoc_rtd_to_cpu(rtd, i)); \
 	     (i)++)
 #define for_each_rtd_codec_dais(rtd, i, dai)				\
 	for ((i) = 0;							\
-	     ((i) < rtd->num_codecs) && ((dai) = asoc_rtd_to_codec(rtd, i)); \
+	     ((i) < rtd->dai_link->num_codecs) && ((dai) = asoc_rtd_to_codec(rtd, i)); \
 	     (i)++)
 #define for_each_rtd_dais(rtd, i, dai)					\
 	for ((i) = 0;							\
-	     ((i) < (rtd)->num_cpus + (rtd)->num_codecs) &&		\
+	     ((i) < (rtd)->dai_link->num_cpus + (rtd)->dai_link->num_codecs) &&	\
 		     ((dai) = (rtd)->dais[i]);				\
 	     (i)++)
 

include/sound/sof.h

@@ -89,6 +89,7 @@ struct snd_sof_pdata {
 	/* machine */
 	struct platform_device *pdev_mach;
 	const struct snd_soc_acpi_mach *machine;
+	const struct snd_sof_of_mach *of_machine;
 
 	void *hw_pdata;
 
@@ -102,6 +103,7 @@ struct snd_sof_pdata {
 struct sof_dev_desc {
 	/* list of machines using this configuration */
 	struct snd_soc_acpi_mach *machines;
+	struct snd_sof_of_mach *of_machines;
 
 	/* alternate list of machines using this configuration */
 	struct snd_soc_acpi_mach *alt_machines;

include/sound/sof/control.h

@@ -117,11 +117,11 @@ struct sof_ipc_ctrl_data {
 	/* control data - add new types if needed */
 	union {
 		/* channel values can be used by volume type controls */
-		struct sof_ipc_ctrl_value_chan chanv[0];
+		DECLARE_FLEX_ARRAY(struct sof_ipc_ctrl_value_chan, chanv);
 		/* component values used by routing controls like mux, mixer */
-		struct sof_ipc_ctrl_value_comp compv[0];
+		DECLARE_FLEX_ARRAY(struct sof_ipc_ctrl_value_comp, compv);
 		/* data can be used by binary controls */
-		struct sof_abi_hdr data[0];
+		DECLARE_FLEX_ARRAY(struct sof_abi_hdr, data);
 	};
 } __packed;
 

include/sound/sof/dai.h

@@ -84,6 +84,7 @@ enum sof_ipc_dai_type {
 	SOF_DAI_AMD_BT,			/**< AMD ACP BT*/
 	SOF_DAI_AMD_SP,			/**< AMD ACP SP */
 	SOF_DAI_AMD_DMIC,		/**< AMD ACP DMIC */
+	SOF_DAI_AMD_HS,			/**< Amd HS */
 	SOF_DAI_MEDIATEK_AFE,		/**< Mediatek AFE */
 };
 
@@ -112,6 +113,7 @@ struct sof_ipc_dai_config {
 		struct sof_ipc_dai_acp_params acpbt;
 		struct sof_ipc_dai_acp_params acpsp;
 		struct sof_ipc_dai_acpdmic_params acpdmic;
+		struct sof_ipc_dai_acp_params acphs;
 		struct sof_ipc_dai_mtk_afe_params afe;
 	};
 } __packed;

include/sound/sof/ipc4/header.h

@@ -427,6 +427,11 @@ struct sof_ipc4_dx_state_info {
 #define SOF_IPC4_NOTIFICATION_TYPE_GET(x)	(((x) & SOF_IPC4_NOTIFICATION_TYPE_MASK) >> \
 						 SOF_IPC4_NOTIFICATION_TYPE_SHIFT)
 
+#define SOF_IPC4_LOG_CORE_SHIFT			12
+#define SOF_IPC4_LOG_CORE_MASK			GENMASK(15, 12)
+#define SOF_IPC4_LOG_CORE_GET(x)		(((x) & SOF_IPC4_LOG_CORE_MASK) >> \
+						 SOF_IPC4_LOG_CORE_SHIFT)
+
 /* Value of notification type field - must fit into 8 bits */
 enum sof_ipc4_notification_type {
 	/* Phrase detected (notification from WoV module) */

include/trace/events/sof.h

+/* SPDX-License-Identifier: GPL-2.0-only */
+/*
+ * Copyright(c) 2022 Intel Corporation. All rights reserved.
+ *
+ * Author: Noah Klayman <noah.klayman@intel.com>
+ */
+
+#undef TRACE_SYSTEM
+#define TRACE_SYSTEM sof
+
+#if !defined(_TRACE_SOF_H) || defined(TRACE_HEADER_MULTI_READ)
+#define _TRACE_SOF_H
+#include <linux/tracepoint.h>
+#include <linux/types.h>
+#include <sound/sof/stream.h>
+#include "../../../sound/soc/sof/sof-audio.h"
+
+DECLARE_EVENT_CLASS(sof_widget_template,
+	TP_PROTO(struct snd_sof_widget *swidget),
+	TP_ARGS(swidget),
+	TP_STRUCT__entry(
+		__string(name, swidget->widget->name)
+		__field(int, use_count)
+	),
+	TP_fast_assign(
+		__assign_str(name, swidget->widget->name);
+		__entry->use_count = swidget->use_count;
+	),
+	TP_printk("name=%s use_count=%d", __get_str(name), __entry->use_count)
+);
+
+DEFINE_EVENT(sof_widget_template, sof_widget_setup,
+	TP_PROTO(struct snd_sof_widget *swidget),
+	TP_ARGS(swidget)
+);
+
+DEFINE_EVENT(sof_widget_template, sof_widget_free,
+	TP_PROTO(struct snd_sof_widget *swidget),
+	TP_ARGS(swidget)
+);
+
+TRACE_EVENT(sof_ipc3_period_elapsed_position,
+	TP_PROTO(struct snd_sof_dev *sdev, struct sof_ipc_stream_posn *posn),
+	TP_ARGS(sdev, posn),
+	TP_STRUCT__entry(
+		__string(device_name, dev_name(sdev->dev))
+		__field(u64, host_posn)
+		__field(u64, dai_posn)
+		__field(u64, wallclock)
+	),
+	TP_fast_assign(
+		__assign_str(device_name, dev_name(sdev->dev));
+		__entry->host_posn = posn->host_posn;
+		__entry->dai_posn = posn->dai_posn;
+		__entry->wallclock = posn->wallclock;
+	),
+	TP_printk("device_name=%s host_posn=%#llx dai_posn=%#llx wallclock=%#llx",
+		  __get_str(device_name), __entry->host_posn, __entry->dai_posn,
+		  __entry->wallclock)
+);
+
+TRACE_EVENT(sof_pcm_pointer_position,
+	TP_PROTO(struct snd_sof_dev *sdev,
+		struct snd_sof_pcm *spcm,
+		struct snd_pcm_substream *substream,
+		snd_pcm_uframes_t dma_posn,
+		snd_pcm_uframes_t dai_posn
+	),
+	TP_ARGS(sdev, spcm, substream, dma_posn, dai_posn),
+	TP_STRUCT__entry(
+		__string(device_name, dev_name(sdev->dev))
+		__field(u32, pcm_id)
+		__field(int, stream)
+		__field(unsigned long, dma_posn)
+		__field(unsigned long, dai_posn)
+	),
+	TP_fast_assign(
+		__assign_str(device_name, dev_name(sdev->dev));
+		__entry->pcm_id = le32_to_cpu(spcm->pcm.pcm_id);
+		__entry->stream = substream->stream;
+		__entry->dma_posn = dma_posn;
+		__entry->dai_posn = dai_posn;
+	),
+	TP_printk("device_name=%s pcm_id=%d stream=%d dma_posn=%lu dai_posn=%lu",
+		  __get_str(device_name), __entry->pcm_id, __entry->stream,
+		  __entry->dma_posn, __entry->dai_posn)
+);
+
+TRACE_EVENT(sof_stream_position_ipc_rx,
+	TP_PROTO(struct device *dev),
+	TP_ARGS(dev),
+	TP_STRUCT__entry(
+		__string(device_name, dev_name(dev))
+	),
+	TP_fast_assign(
+		__assign_str(device_name, dev_name(dev));
+	),
+	TP_printk("device_name=%s", __get_str(device_name))
+);
+
+TRACE_EVENT(sof_ipc4_fw_config,
+	TP_PROTO(struct snd_sof_dev *sdev, char *key, u32 value),
+	TP_ARGS(sdev, key, value),
+	TP_STRUCT__entry(
+		__string(device_name, dev_name(sdev->dev))
+		__string(key, key)
+		__field(u32, value)
+	),
+	TP_fast_assign(
+		__assign_str(device_name, dev_name(sdev->dev));
+		__assign_str(key, key);
+		__entry->value = value;
+	),
+	TP_printk("device_name=%s key=%s value=%d",
+		  __get_str(device_name), __get_str(key), __entry->value)
+);
+
+#endif /* _TRACE_SOF_H */
+
+/* This part must be outside protection */
+#include <trace/define_trace.h>

include/trace/events/sof_intel.h

+/* SPDX-License-Identifier: GPL-2.0-only */
+/*
+ * Copyright(c) 2022 Intel Corporation. All rights reserved.
+ *
+ * Author: Noah Klayman <noah.klayman@intel.com>
+ */
+
+#undef TRACE_SYSTEM
+#define TRACE_SYSTEM sof_intel
+
+#if !defined(_TRACE_SOF_INTEL_H) || defined(TRACE_HEADER_MULTI_READ)
+#define _TRACE_SOF_INTEL_H
+#include <linux/tracepoint.h>
+#include <sound/hdaudio.h>
+#include "../../../sound/soc/sof/sof-audio.h"
+
+TRACE_EVENT(sof_intel_hda_irq,
+	TP_PROTO(struct snd_sof_dev *sdev, char *source),
+	TP_ARGS(sdev, source),
+	TP_STRUCT__entry(
+		__string(device_name, dev_name(sdev->dev))
+		__string(source, source)
+	),
+	TP_fast_assign(
+		__assign_str(device_name, dev_name(sdev->dev));
+		__assign_str(source, source);
+	),
+	TP_printk("device_name=%s source=%s",
+		  __get_str(device_name), __get_str(source))
+);
+
+DECLARE_EVENT_CLASS(sof_intel_ipc_firmware_template,
+	TP_ARGS(struct snd_sof_dev *sdev, u32 msg, u32 msg_ext),
+	TP_PROTO(sdev, msg, msg_ext),
+	TP_STRUCT__entry(
+		__string(device_name, dev_name(sdev->dev))
+		__field(u32, msg)
+		__field(u32, msg_ext)
+	),
+	TP_fast_assign(
+		__assign_str(device_name, dev_name(sdev->dev));
+		__entry->msg = msg;
+		__entry->msg_ext = msg_ext;
+	),
+	TP_printk("device_name=%s msg=%#x msg_ext=%#x",
+		  __get_str(device_name), __entry->msg, __entry->msg_ext)
+);
+
+DEFINE_EVENT(sof_intel_ipc_firmware_template, sof_intel_ipc_firmware_response,
+	TP_PROTO(struct snd_sof_dev *sdev, u32 msg, u32 msg_ext),
+	TP_ARGS(sdev, msg, msg_ext)
+);
+
+DEFINE_EVENT(sof_intel_ipc_firmware_template, sof_intel_ipc_firmware_initiated,
+	TP_PROTO(struct snd_sof_dev *sdev, u32 msg, u32 msg_ext),
+	TP_ARGS(sdev, msg, msg_ext)
+);
+
+TRACE_EVENT(sof_intel_D0I3C_updated,
+	TP_PROTO(struct snd_sof_dev *sdev, u8 reg),
+	TP_ARGS(sdev, reg),
+	TP_STRUCT__entry(
+		__string(device_name, dev_name(sdev->dev))
+		__field(u8, reg)
+	),
+	TP_fast_assign(
+		__assign_str(device_name, dev_name(sdev->dev));
+		__entry->reg = reg;
+	),
+	TP_printk("device_name=%s register=%#x",
+		  __get_str(device_name), __entry->reg)
+);
+
+TRACE_EVENT(sof_intel_hda_irq_ipc_check,
+	TP_PROTO(struct snd_sof_dev *sdev, u32 irq_status),
+	TP_ARGS(sdev, irq_status),
+	TP_STRUCT__entry(
+		__string(device_name, dev_name(sdev->dev))
+		__field(u32, irq_status)
+	),
+	TP_fast_assign(
+		__assign_str(device_name, dev_name(sdev->dev));
+		__entry->irq_status = irq_status;
+	),
+	TP_printk("device_name=%s irq_status=%#x",
+		  __get_str(device_name), __entry->irq_status)
+);
+
+TRACE_EVENT(sof_intel_hda_dsp_pcm,
+	TP_PROTO(struct snd_sof_dev *sdev,
+		struct hdac_stream *hstream,
+		struct snd_pcm_substream *substream,
+		snd_pcm_uframes_t pos
+	),
+	TP_ARGS(sdev, hstream, substream, pos),
+	TP_STRUCT__entry(
+		__string(device_name, dev_name(sdev->dev))
+		__field(u32, hstream_index)
+		__field(u32, substream)
+		__field(unsigned long, pos)
+	),
+	TP_fast_assign(
+		__assign_str(device_name, dev_name(sdev->dev));
+		__entry->hstream_index = hstream->index;
+		__entry->substream = substream->stream;
+		__entry->pos = pos;
+	),
+	TP_printk("device_name=%s hstream_index=%d substream=%d pos=%lu",
+		  __get_str(device_name), __entry->hstream_index,
+		  __entry->substream, __entry->pos)
+);
+
+TRACE_EVENT(sof_intel_hda_dsp_stream_status,
+	TP_PROTO(struct device *dev, struct hdac_stream *s, u32 status),
+	TP_ARGS(dev, s, status),
+	TP_STRUCT__entry(
+		__string(device_name, dev_name(dev))
+		__field(u32, stream)
+		__field(u32, status)
+	),
+	TP_fast_assign(
+		__assign_str(device_name, dev_name(dev));
+		__entry->stream = s->index;
+		__entry->status = status;
+	),
+	TP_printk("device_name=%s stream=%d status=%#x",
+		  __get_str(device_name), __entry->stream, __entry->status)
+);
+
+TRACE_EVENT(sof_intel_hda_dsp_check_stream_irq,
+	TP_PROTO(struct snd_sof_dev *sdev, u32 status),
+	TP_ARGS(sdev, status),
+	TP_STRUCT__entry(
+		__string(device_name, dev_name(sdev->dev))
+		__field(u32, status)
+	),
+	TP_fast_assign(
+		__assign_str(device_name, dev_name(sdev->dev));
+		__entry->status = status;
+	),
+	TP_printk("device_name=%s status=%#x",
+		  __get_str(device_name), __entry->status)
+);
+
+#endif /* _TRACE_SOF_INTEL_H */
+
+/* This part must be outside protection */
+#include <trace/define_trace.h>

lib/string_helpers.c

@@ -131,6 +131,50 @@ void string_get_size(u64 size, u64 blk_size, const enum string_size_units units,
 }
 EXPORT_SYMBOL(string_get_size);
 
+/**
+ * parse_int_array_user - Split string into a sequence of integers
+ * @from:	The user space buffer to read from
+ * @count:	The maximum number of bytes to read
+ * @array:	Returned pointer to sequence of integers
+ *
+ * On success @array is allocated and initialized with a sequence of
+ * integers extracted from the @from plus an additional element that
+ * begins the sequence and specifies the integers count.
+ *
+ * Caller takes responsibility for freeing @array when it is no longer
+ * needed.
+ */
+int parse_int_array_user(const char __user *from, size_t count, int **array)
+{
+	int *ints, nints;
+	char *buf;
+	int ret = 0;
+
+	buf = memdup_user_nul(from, count);
+	if (IS_ERR(buf))
+		return PTR_ERR(buf);
+
+	get_options(buf, 0, &nints);
+	if (!nints) {
+		ret = -ENOENT;
+		goto free_buf;
+	}
+
+	ints = kcalloc(nints + 1, sizeof(*ints), GFP_KERNEL);
+	if (!ints) {
+		ret = -ENOMEM;
+		goto free_buf;
+	}
+
+	get_options(buf, nints + 1, ints);
+	*array = ints;
+
+free_buf:
+	kfree(buf);
+	return ret;
+}
+EXPORT_SYMBOL(parse_int_array_user);
+
 static bool unescape_space(char **src, char **dst)
 {
 	char *p = *dst, *q = *src;

sound/hda/intel-dsp-config.c

@@ -427,6 +427,11 @@ static const struct config_entry config_table[] = {
 		.device = 0x51cd,
 	},
 	/* Alderlake-PS */
+	{
+		.flags = FLAG_SOF,
+		.device = 0x51c9,
+		.codec_hid =  &essx_83x6,
+	},
 	{
 		.flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE,
 		.device = 0x51c9,

sound/hda/intel-nhlt.c

@@ -157,6 +157,85 @@ int intel_nhlt_ssp_endpoint_mask(struct nhlt_acpi_table *nhlt, u8 device_type)
 }
 EXPORT_SYMBOL(intel_nhlt_ssp_endpoint_mask);
 
+#define SSP_BLOB_V1_0_SIZE		84
+#define SSP_BLOB_V1_0_MDIVC_OFFSET	19 /* offset in u32 */
+
+#define SSP_BLOB_V1_5_SIZE		96
+#define SSP_BLOB_V1_5_MDIVC_OFFSET	21 /* offset in u32 */
+#define SSP_BLOB_VER_1_5		0xEE000105
+
+#define SSP_BLOB_V2_0_SIZE		88
+#define SSP_BLOB_V2_0_MDIVC_OFFSET	20 /* offset in u32 */
+#define SSP_BLOB_VER_2_0		0xEE000200
+
+int intel_nhlt_ssp_mclk_mask(struct nhlt_acpi_table *nhlt, int ssp_num)
+{
+	struct nhlt_endpoint *epnt;
+	struct nhlt_fmt *fmt;
+	struct nhlt_fmt_cfg *cfg;
+	int mclk_mask = 0;
+	int i, j;
+
+	if (!nhlt)
+		return 0;
+
+	epnt = (struct nhlt_endpoint *)nhlt->desc;
+	for (i = 0; i < nhlt->endpoint_count; i++) {
+
+		/* we only care about endpoints connected to an audio codec over SSP */
+		if (epnt->linktype == NHLT_LINK_SSP &&
+		    epnt->device_type == NHLT_DEVICE_I2S &&
+		    epnt->virtual_bus_id == ssp_num) {
+
+			fmt = (struct nhlt_fmt *)(epnt->config.caps + epnt->config.size);
+			cfg = fmt->fmt_config;
+
+			/*
+			 * In theory all formats should use the same MCLK but it doesn't hurt to
+			 * double-check that the configuration is consistent
+			 */
+			for (j = 0; j < fmt->fmt_count; j++) {
+				u32 *blob;
+				int mdivc_offset;
+				int size;
+
+				/* first check we have enough data to read the blob type */
+				if (cfg->config.size < 8)
+					return -EINVAL;
+
+				blob = (u32 *)cfg->config.caps;
+
+				if (blob[1] == SSP_BLOB_VER_2_0) {
+					mdivc_offset = SSP_BLOB_V2_0_MDIVC_OFFSET;
+					size = SSP_BLOB_V2_0_SIZE;
+				} else if (blob[1] == SSP_BLOB_VER_1_5) {
+					mdivc_offset = SSP_BLOB_V1_5_MDIVC_OFFSET;
+					size = SSP_BLOB_V1_5_SIZE;
+				} else {
+					mdivc_offset = SSP_BLOB_V1_0_MDIVC_OFFSET;
+					size = SSP_BLOB_V1_0_SIZE;
+				}
+
+				/* make sure we have enough data for the fixed part of the blob */
+				if (cfg->config.size < size)
+					return -EINVAL;
+
+				mclk_mask |=  blob[mdivc_offset] & GENMASK(1, 0);
+
+				cfg = (struct nhlt_fmt_cfg *)(cfg->config.caps + cfg->config.size);
+			}
+		}
+		epnt = (struct nhlt_endpoint *)((u8 *)epnt + epnt->length);
+	}
+
+	/* make sure only one MCLK is used */
+	if (hweight_long(mclk_mask) != 1)
+		return -EINVAL;
+
+	return mclk_mask;
+}
+EXPORT_SYMBOL(intel_nhlt_ssp_mclk_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/Kconfig

@@ -68,6 +68,7 @@ config SND_SOC_ACPI
 # All the supported SoCs
 source "sound/soc/adi/Kconfig"
 source "sound/soc/amd/Kconfig"
+source "sound/soc/apple/Kconfig"
 source "sound/soc/atmel/Kconfig"
 source "sound/soc/au1x/Kconfig"
 source "sound/soc/bcm/Kconfig"

sound/soc/Makefile

@@ -34,6 +34,7 @@ obj-$(CONFIG_SND_SOC_ACPI) += snd-soc-acpi.o
 obj-$(CONFIG_SND_SOC)	+= snd-soc-core.o
 obj-$(CONFIG_SND_SOC)	+= codecs/
 obj-$(CONFIG_SND_SOC)	+= generic/
+obj-$(CONFIG_SND_SOC)	+= apple/
 obj-$(CONFIG_SND_SOC)	+= adi/
 obj-$(CONFIG_SND_SOC)	+= amd/
 obj-$(CONFIG_SND_SOC)	+= atmel/

sound/soc/amd/Kconfig

@@ -122,8 +122,29 @@ config SND_SOC_AMD_RPL_ACP6x
         tristate "AMD Audio Coprocessor-v6.2 RPL support"
         depends on X86 && PCI
         help
-          This option enables Audio Coprocessor i.e ACP v6.2 support on
+          This option enables Audio Coprocessor i.e. ACP v6.2 support on
           AMD RPL platform. By enabling this flag build will be
           triggered for ACP PCI driver.
           Say m if you have such a device.
           If unsure select "N".
+
+config SND_SOC_AMD_PS
+        tristate "AMD Audio Coprocessor-v6.2 Pink Sardine support"
+        depends on X86 && PCI && ACPI
+        help
+          This option enables Audio Coprocessor i.e ACP v6.2 support on
+          AMD Pink sardine platform. By enabling this flag build will be
+          triggered for ACP PCI driver, ACP PDM DMA driver.
+          Say m if you have such a device.
+          If unsure select "N".
+
+config SND_SOC_AMD_PS_MACH
+        tristate "AMD PINK SARDINE support for DMIC"
+        select SND_SOC_DMIC
+        depends on SND_SOC_AMD_PS
+        help
+          This option enables machine driver for Pink Sardine platform
+          using dmic. ACP IP has PDM Decoder block with DMA controller.
+          DMIC can be connected directly to ACP IP.
+          Say m if you have such a device.
+          If unsure select "N".

sound/soc/amd/Makefile

@@ -18,3 +18,4 @@ obj-$(CONFIG_SND_SOC_AMD_ACP6x) += yc/
 obj-$(CONFIG_SND_SOC_AMD_ACP_COMMON) += acp/
 obj-$(CONFIG_SND_AMD_ACP_CONFIG) += snd-acp-config.o
 obj-$(CONFIG_SND_SOC_AMD_RPL_ACP6x) += rpl/
+obj-$(CONFIG_SND_SOC_AMD_PS) += ps/

sound/soc/amd/acp/acp-i2s.c

@@ -25,6 +25,65 @@
 
 #define DRV_NAME "acp_i2s_playcap"
 
+static int acp_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
+			   unsigned int fmt)
+{
+	struct acp_dev_data *adata = snd_soc_dai_get_drvdata(cpu_dai);
+	int mode;
+
+	mode = fmt & SND_SOC_DAIFMT_FORMAT_MASK;
+	switch (mode) {
+	case SND_SOC_DAIFMT_I2S:
+		adata->tdm_mode = TDM_DISABLE;
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		adata->tdm_mode = TDM_ENABLE;
+		break;
+	default:
+		return -EINVAL;
+	}
+	return 0;
+}
+
+static int acp_i2s_set_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask, u32 rx_mask,
+				int slots, int slot_width)
+{
+	struct device *dev = dai->component->dev;
+	struct acp_dev_data *adata = snd_soc_dai_get_drvdata(dai);
+	struct acp_stream *stream;
+	int slot_len;
+
+	switch (slot_width) {
+	case SLOT_WIDTH_8:
+		slot_len = 8;
+		break;
+	case SLOT_WIDTH_16:
+		slot_len = 16;
+		break;
+	case SLOT_WIDTH_24:
+		slot_len = 24;
+		break;
+	case SLOT_WIDTH_32:
+		slot_len = 0;
+		break;
+	default:
+		dev_err(dev, "Unsupported bitdepth %d\n", slot_width);
+		return -EINVAL;
+	}
+
+	spin_lock_irq(&adata->acp_lock);
+	list_for_each_entry(stream, &adata->stream_list, list) {
+		if (tx_mask && stream->dir == SNDRV_PCM_STREAM_PLAYBACK)
+			adata->tdm_tx_fmt[stream->dai_id - 1] =
+					FRM_LEN | (slots << 15) | (slot_len << 18);
+		else if (rx_mask && stream->dir == SNDRV_PCM_STREAM_CAPTURE)
+			adata->tdm_rx_fmt[stream->dai_id - 1] =
+					FRM_LEN | (slots << 15) | (slot_len << 18);
+	}
+	spin_unlock_irq(&adata->acp_lock);
+	return 0;
+}
+
 static int acp_i2s_hwparams(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params,
 			    struct snd_soc_dai *dai)
 {
@@ -33,7 +92,7 @@ static int acp_i2s_hwparams(struct snd_pcm_substream *substream, struct snd_pcm_
 	struct acp_resource *rsrc;
 	u32 val;
 	u32 xfer_resolution;
-	u32 reg_val;
+	u32 reg_val, fmt_reg, tdm_fmt;
 	u32 lrclk_div_val, bclk_div_val;
 
 	adata = snd_soc_dai_get_drvdata(dai);
@@ -62,12 +121,15 @@ static int acp_i2s_hwparams(struct snd_pcm_substream *substream, struct snd_pcm_
 		switch (dai->driver->id) {
 		case I2S_BT_INSTANCE:
 			reg_val = ACP_BTTDM_ITER;
+			fmt_reg = ACP_BTTDM_TXFRMT;
 			break;
 		case I2S_SP_INSTANCE:
 			reg_val = ACP_I2STDM_ITER;
+			fmt_reg = ACP_I2STDM_TXFRMT;
 			break;
 		case I2S_HS_INSTANCE:
 			reg_val = ACP_HSTDM_ITER;
+			fmt_reg = ACP_HSTDM_TXFRMT;
 			break;
 		default:
 			dev_err(dev, "Invalid dai id %x\n", dai->driver->id);
@@ -77,12 +139,15 @@ static int acp_i2s_hwparams(struct snd_pcm_substream *substream, struct snd_pcm_
 		switch (dai->driver->id) {
 		case I2S_BT_INSTANCE:
 			reg_val = ACP_BTTDM_IRER;
+			fmt_reg = ACP_BTTDM_RXFRMT;
 			break;
 		case I2S_SP_INSTANCE:
 			reg_val = ACP_I2STDM_IRER;
+			fmt_reg = ACP_I2STDM_RXFRMT;
 			break;
 		case I2S_HS_INSTANCE:
 			reg_val = ACP_HSTDM_IRER;
+			fmt_reg = ACP_HSTDM_RXFRMT;
 			break;
 		default:
 			dev_err(dev, "Invalid dai id %x\n", dai->driver->id);
@@ -95,6 +160,16 @@ static int acp_i2s_hwparams(struct snd_pcm_substream *substream, struct snd_pcm_
 	val = val | (xfer_resolution  << 3);
 	writel(val, adata->acp_base + reg_val);
 
+	if (adata->tdm_mode) {
+		val = readl(adata->acp_base + reg_val);
+		writel(val | BIT(1), adata->acp_base + reg_val);
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+			tdm_fmt = adata->tdm_tx_fmt[dai->driver->id - 1];
+		else
+			tdm_fmt = adata->tdm_rx_fmt[dai->driver->id - 1];
+		writel(tdm_fmt, adata->acp_base + fmt_reg);
+	}
+
 	if (rsrc->soc_mclk) {
 		switch (params_format(params)) {
 		case SNDRV_PCM_FORMAT_S16_LE:
@@ -443,6 +518,7 @@ static int acp_i2s_startup(struct snd_pcm_substream *substream, struct snd_soc_d
 	stream->id = dai->driver->id + dir;
 	stream->dai_id = dai->driver->id;
 	stream->irq_bit = irq_bit;
+	stream->dir = substream->stream;
 
 	return 0;
 }
@@ -452,6 +528,8 @@ const struct snd_soc_dai_ops asoc_acp_cpu_dai_ops = {
 	.hw_params = acp_i2s_hwparams,
 	.prepare = acp_i2s_prepare,
 	.trigger = acp_i2s_trigger,
+	.set_fmt = acp_i2s_set_fmt,
+	.set_tdm_slot = acp_i2s_set_tdm_slot,
 };
 EXPORT_SYMBOL_NS_GPL(asoc_acp_cpu_dai_ops, SND_SOC_ACP_COMMON);
 

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

@@ -584,7 +584,7 @@ int acp_sofdsp_dai_links_create(struct snd_soc_card *card)
 	if (drv_data->dmic_cpu_id)
 		num_links++;
 
-	links = devm_kzalloc(dev, sizeof(struct snd_soc_dai_link) * num_links, GFP_KERNEL);
+	links = devm_kcalloc(dev, num_links, sizeof(struct snd_soc_dai_link), GFP_KERNEL);
 	if (!links)
 		return -ENOMEM;
 
@@ -749,7 +749,7 @@ int acp_legacy_dai_links_create(struct snd_soc_card *card)
 	if (drv_data->dmic_cpu_id)
 		num_links++;
 
-	links = devm_kzalloc(dev, sizeof(struct snd_soc_dai_link) * num_links, GFP_KERNEL);
+	links = devm_kcalloc(dev, num_links, sizeof(struct snd_soc_dai_link), GFP_KERNEL);
 	if (!links)
 		return -ENOMEM;
 

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

@@ -62,10 +62,9 @@ static int acp_pci_probe(struct pci_dev *pci, const struct pci_device_id *pci_id
 	if (!chip)
 		return -ENOMEM;
 
-	if (pci_enable_device(pci)) {
-		dev_err(&pci->dev, "pci_enable_device failed\n");
-		return -ENODEV;
-	}
+	if (pci_enable_device(pci))
+		return dev_err_probe(&pci->dev, -ENODEV,
+				     "pci_enable_device failed\n");
 
 	ret = pci_request_regions(pci, "AMD ACP3x audio");
 	if (ret < 0) {
@@ -105,14 +104,13 @@ static int acp_pci_probe(struct pci_dev *pci, const struct pci_device_id *pci_id
 	chip->base = devm_ioremap(&pci->dev, addr, pci_resource_len(pci, 0));
 	if (!chip->base) {
 		ret = -ENOMEM;
-		goto release_regions;
+		goto unregister_dmic_dev;
 	}
 
-	res = devm_kzalloc(&pci->dev, sizeof(struct resource) * num_res, GFP_KERNEL);
+	res = devm_kcalloc(&pci->dev, num_res, sizeof(struct resource), GFP_KERNEL);
 	if (!res) {
-		platform_device_unregister(dmic_dev);
 		ret = -ENOMEM;
-		goto release_regions;
+		goto unregister_dmic_dev;
 	}
 
 	for (i = 0; i < num_res; i++, res_acp++) {
@@ -139,13 +137,14 @@ static int acp_pci_probe(struct pci_dev *pci, const struct pci_device_id *pci_id
 	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);
-		goto release_regions;
+		goto unregister_dmic_dev;
 	}
 
 	return ret;
 
+unregister_dmic_dev:
+	platform_device_unregister(dmic_dev);
 release_regions:
 	pci_release_regions(pci);
 disable_pci:

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

@@ -94,7 +94,7 @@ static irqreturn_t i2s_irq_handler(int irq, void *data)
 	struct acp_resource *rsrc = adata->rsrc;
 	struct acp_stream *stream;
 	u16 i2s_flag = 0;
-	u32 ext_intr_stat, ext_intr_stat1, i;
+	u32 ext_intr_stat, ext_intr_stat1;
 
 	if (!adata)
 		return IRQ_NONE;
@@ -104,25 +104,24 @@ static irqreturn_t i2s_irq_handler(int irq, void *data)
 
 	ext_intr_stat = readl(ACP_EXTERNAL_INTR_STAT(adata, rsrc->irqp_used));
 
-	for (i = 0; i < ACP_MAX_STREAM; i++) {
-		stream = adata->stream[i];
-		if (stream && (ext_intr_stat & stream->irq_bit)) {
+	spin_lock(&adata->acp_lock);
+	list_for_each_entry(stream, &adata->stream_list, list) {
+		if (ext_intr_stat & stream->irq_bit) {
 			writel(stream->irq_bit,
 			       ACP_EXTERNAL_INTR_STAT(adata, rsrc->irqp_used));
 			snd_pcm_period_elapsed(stream->substream);
 			i2s_flag = 1;
-			break;
 		}
 		if (adata->rsrc->no_of_ctrls == 2) {
-			if (stream && (ext_intr_stat1 & stream->irq_bit)) {
+			if (ext_intr_stat1 & stream->irq_bit) {
 				writel(stream->irq_bit, ACP_EXTERNAL_INTR_STAT(adata,
 				       (rsrc->irqp_used - 1)));
 				snd_pcm_period_elapsed(stream->substream);
 				i2s_flag = 1;
-				break;
 			}
 		}
 	}
+	spin_unlock(&adata->acp_lock);
 	if (i2s_flag)
 		return IRQ_HANDLED;
 
@@ -146,9 +145,8 @@ static void config_pte_for_stream(struct acp_dev_data *adata, struct acp_stream
 	writel(0x01, adata->acp_base + ACPAXI2AXI_ATU_CTRL);
 }
 
-static void config_acp_dma(struct acp_dev_data *adata, int cpu_id, int size)
+static void config_acp_dma(struct acp_dev_data *adata, struct acp_stream *stream, int size)
 {
-	struct acp_stream *stream = adata->stream[cpu_id];
 	struct snd_pcm_substream *substream = stream->substream;
 	struct acp_resource *rsrc = adata->rsrc;
 	dma_addr_t addr = substream->dma_buffer.addr;
@@ -174,13 +172,10 @@ static void config_acp_dma(struct acp_dev_data *adata, int cpu_id, int size)
 
 static int acp_dma_open(struct snd_soc_component *component, struct snd_pcm_substream *substream)
 {
-	struct snd_soc_pcm_runtime *soc_runtime = asoc_substream_to_rtd(substream);
-	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(soc_runtime, 0);
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct device *dev = component->dev;
 	struct acp_dev_data *adata = dev_get_drvdata(dev);
 	struct acp_stream *stream;
-	int stream_id = cpu_dai->driver->id * 2 + substream->stream;
 	int ret;
 
 	stream = kzalloc(sizeof(*stream), GFP_KERNEL);
@@ -188,7 +183,10 @@ static int acp_dma_open(struct snd_soc_component *component, struct snd_pcm_subs
 		return -ENOMEM;
 
 	stream->substream = substream;
-	adata->stream[stream_id] = stream;
+
+	spin_lock_irq(&adata->acp_lock);
+	list_add_tail(&stream->list, &adata->stream_list);
+	spin_unlock_irq(&adata->acp_lock);
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 		runtime->hw = acp_pcm_hardware_playback;
@@ -212,16 +210,13 @@ static int acp_dma_hw_params(struct snd_soc_component *component,
 			     struct snd_pcm_substream *substream,
 			     struct snd_pcm_hw_params *params)
 {
-	struct snd_soc_pcm_runtime *soc_runtime = asoc_substream_to_rtd(substream);
 	struct acp_dev_data *adata = snd_soc_component_get_drvdata(component);
-	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(soc_runtime, 0);
 	struct acp_stream *stream = substream->runtime->private_data;
-	int stream_id = cpu_dai->driver->id * 2 + substream->stream;
 	u64 size = params_buffer_bytes(params);
 
 	/* Configure ACP DMA block with params */
 	config_pte_for_stream(adata, stream);
-	config_acp_dma(adata, stream_id, size);
+	config_acp_dma(adata, stream, size);
 
 	return 0;
 }
@@ -261,16 +256,15 @@ static int acp_dma_new(struct snd_soc_component *component,
 static int acp_dma_close(struct snd_soc_component *component,
 			 struct snd_pcm_substream *substream)
 {
-	struct snd_soc_pcm_runtime *soc_runtime = asoc_substream_to_rtd(substream);
-	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(soc_runtime, 0);
 	struct device *dev = component->dev;
 	struct acp_dev_data *adata = dev_get_drvdata(dev);
-	struct acp_stream *stream;
-	int stream_id = cpu_dai->driver->id * 2 + substream->stream;
+	struct acp_stream *stream = substream->runtime->private_data;
 
-	stream = adata->stream[stream_id];
+	/* Remove entry from list */
+	spin_lock_irq(&adata->acp_lock);
+	list_del(&stream->list);
+	spin_unlock_irq(&adata->acp_lock);
 	kfree(stream);
-	adata->stream[stream_id] = NULL;
 
 	return 0;
 }
@@ -305,6 +299,10 @@ int acp_platform_register(struct device *dev)
 		dev_err(dev, "Fail to register acp i2s component\n");
 		return status;
 	}
+
+	INIT_LIST_HEAD(&adata->stream_list);
+	spin_lock_init(&adata->acp_lock);
+
 	return 0;
 }
 EXPORT_SYMBOL_NS_GPL(acp_platform_register, SND_SOC_ACP_COMMON);

sound/soc/amd/acp/amd.h

@@ -21,9 +21,9 @@
 #define ACP3X_DEV			3
 #define ACP6X_DEV			6
 
-#define I2S_SP_INSTANCE			0x00
-#define I2S_BT_INSTANCE			0x01
-#define DMIC_INSTANCE			0x02
+#define DMIC_INSTANCE			0x00
+#define I2S_SP_INSTANCE			0x01
+#define I2S_BT_INSTANCE			0x02
 #define I2S_HS_INSTANCE			0x03
 
 #define MEM_WINDOW_START		0x4080000
@@ -84,6 +84,14 @@
 
 #define ACP_MAX_STREAM			8
 
+#define TDM_ENABLE	1
+#define TDM_DISABLE	0
+
+#define SLOT_WIDTH_8	0x8
+#define SLOT_WIDTH_16	0x10
+#define SLOT_WIDTH_24	0x18
+#define SLOT_WIDTH_32	0x20
+
 struct acp_chip_info {
 	char *name;		/* Platform name */
 	unsigned int acp_rev;	/* ACP Revision id */
@@ -91,10 +99,12 @@ struct acp_chip_info {
 };
 
 struct acp_stream {
+	struct list_head list;
 	struct snd_pcm_substream *substream;
 	int irq_bit;
 	int dai_id;
 	int id;
+	int dir;
 	u64 bytescount;
 	u32 reg_offset;
 	u32 pte_offset;
@@ -119,11 +129,13 @@ struct acp_dev_data {
 	void __iomem *acp_base;
 	unsigned int i2s_irq;
 
+	bool tdm_mode;
 	/* SOC specific dais */
 	struct snd_soc_dai_driver *dai_driver;
 	int num_dai;
 
-	struct acp_stream *stream[ACP_MAX_STREAM];
+	struct list_head stream_list;
+	spinlock_t acp_lock;
 
 	struct snd_soc_acpi_mach *machines;
 	struct platform_device *mach_dev;
@@ -132,6 +144,8 @@ struct acp_dev_data {
 	u32 lrclk_div;
 
 	struct acp_resource *rsrc;
+	u32 tdm_tx_fmt[3];
+	u32 tdm_rx_fmt[3];
 };
 
 union acp_i2stdm_mstrclkgen {

sound/soc/amd/ps/Makefile

+# SPDX-License-Identifier: GPL-2.0+
+# Pink Sardine platform Support
+snd-pci-ps-objs := pci-ps.o
+snd-ps-pdm-dma-objs := ps-pdm-dma.o
+snd-soc-ps-mach-objs := ps-mach.o
+
+obj-$(CONFIG_SND_SOC_AMD_PS) += snd-pci-ps.o
+obj-$(CONFIG_SND_SOC_AMD_PS) += snd-ps-pdm-dma.o
+obj-$(CONFIG_SND_SOC_AMD_PS_MACH)   += snd-soc-ps-mach.o

sound/soc/amd/ps/acp62.h

+/* SPDX-License-Identifier: GPL-2.0+ */
+/*
+ * AMD ALSA SoC PDM Driver
+ *
+ * Copyright (C) 2022 Advanced Micro Devices, Inc. All rights reserved.
+ */
+
+#include <sound/acp62_chip_offset_byte.h>
+
+#define ACP_DEVICE_ID 0x15E2
+#define ACP6x_REG_START		0x1240000
+#define ACP6x_REG_END		0x1250200
+#define ACP6x_DEVS		3
+#define ACP6x_PDM_MODE		1
+
+#define ACP_SOFT_RESET_SOFTRESET_AUDDONE_MASK	0x00010001
+#define ACP_PGFSM_CNTL_POWER_ON_MASK	1
+#define ACP_PGFSM_CNTL_POWER_OFF_MASK	0
+#define ACP_PGFSM_STATUS_MASK		3
+#define ACP_POWERED_ON			0
+#define ACP_POWER_ON_IN_PROGRESS	1
+#define ACP_POWERED_OFF		2
+#define ACP_POWER_OFF_IN_PROGRESS	3
+
+#define ACP_ERROR_MASK 0x20000000
+#define ACP_EXT_INTR_STAT_CLEAR_MASK 0xFFFFFFFF
+#define PDM_DMA_STAT 0x10
+
+#define PDM_DMA_INTR_MASK	0x10000
+#define ACP_ERROR_STAT	29
+#define PDM_DECIMATION_FACTOR	2
+#define ACP_PDM_CLK_FREQ_MASK	7
+#define ACP_WOV_MISC_CTRL_MASK	0x10
+#define ACP_PDM_ENABLE		1
+#define ACP_PDM_DISABLE		0
+#define ACP_PDM_DMA_EN_STATUS	2
+#define TWO_CH		2
+#define DELAY_US	5
+#define ACP_COUNTER	20000
+
+#define ACP_SRAM_PTE_OFFSET	0x03800000
+#define PAGE_SIZE_4K_ENABLE	2
+#define PDM_PTE_OFFSET		0
+#define PDM_MEM_WINDOW_START	0x4000000
+
+#define CAPTURE_MIN_NUM_PERIODS     4
+#define CAPTURE_MAX_NUM_PERIODS     4
+#define CAPTURE_MAX_PERIOD_SIZE     8192
+#define CAPTURE_MIN_PERIOD_SIZE     4096
+
+#define MAX_BUFFER (CAPTURE_MAX_PERIOD_SIZE * CAPTURE_MAX_NUM_PERIODS)
+#define MIN_BUFFER MAX_BUFFER
+
+/* time in ms for runtime suspend delay */
+#define ACP_SUSPEND_DELAY_MS	2000
+
+enum acp_config {
+	ACP_CONFIG_0 = 0,
+	ACP_CONFIG_1,
+	ACP_CONFIG_2,
+	ACP_CONFIG_3,
+	ACP_CONFIG_4,
+	ACP_CONFIG_5,
+	ACP_CONFIG_6,
+	ACP_CONFIG_7,
+	ACP_CONFIG_8,
+	ACP_CONFIG_9,
+	ACP_CONFIG_10,
+	ACP_CONFIG_11,
+	ACP_CONFIG_12,
+	ACP_CONFIG_13,
+	ACP_CONFIG_14,
+	ACP_CONFIG_15,
+};
+
+struct pdm_stream_instance {
+	u16 num_pages;
+	u16 channels;
+	dma_addr_t dma_addr;
+	u64 bytescount;
+	void __iomem *acp62_base;
+};
+
+struct pdm_dev_data {
+	u32 pdm_irq;
+	void __iomem *acp62_base;
+	struct snd_pcm_substream *capture_stream;
+};
+
+static inline u32 acp62_readl(void __iomem *base_addr)
+{
+	return readl(base_addr);
+}
+
+static inline void acp62_writel(u32 val, void __iomem *base_addr)
+{
+	writel(val, base_addr);
+}

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

+// SPDX-License-Identifier: GPL-2.0+
+/*
+ * AMD Pink Sardine ACP PCI Driver
+ *
+ * Copyright 2022 Advanced Micro Devices, Inc.
+ */
+
+#include <linux/pci.h>
+#include <linux/module.h>
+#include <linux/io.h>
+#include <linux/delay.h>
+#include <linux/platform_device.h>
+#include <linux/acpi.h>
+#include <linux/interrupt.h>
+#include <sound/pcm_params.h>
+#include <linux/pm_runtime.h>
+
+#include "acp62.h"
+
+struct acp62_dev_data {
+	void __iomem *acp62_base;
+	struct resource *res;
+	bool acp62_audio_mode;
+	struct platform_device *pdev[ACP6x_DEVS];
+};
+
+static int acp62_power_on(void __iomem *acp_base)
+{
+	u32 val;
+	int timeout;
+
+	val = acp62_readl(acp_base + ACP_PGFSM_STATUS);
+
+	if (!val)
+		return val;
+
+	if ((val & ACP_PGFSM_STATUS_MASK) != ACP_POWER_ON_IN_PROGRESS)
+		acp62_writel(ACP_PGFSM_CNTL_POWER_ON_MASK, acp_base + ACP_PGFSM_CONTROL);
+	timeout = 0;
+	while (++timeout < 500) {
+		val = acp62_readl(acp_base + ACP_PGFSM_STATUS);
+		if (!val)
+			return 0;
+		udelay(1);
+	}
+	return -ETIMEDOUT;
+}
+
+static int acp62_reset(void __iomem *acp_base)
+{
+	u32 val;
+	int timeout;
+
+	acp62_writel(1, acp_base + ACP_SOFT_RESET);
+	timeout = 0;
+	while (++timeout < 500) {
+		val = acp62_readl(acp_base + ACP_SOFT_RESET);
+		if (val & ACP_SOFT_RESET_SOFTRESET_AUDDONE_MASK)
+			break;
+		cpu_relax();
+	}
+	acp62_writel(0, acp_base + ACP_SOFT_RESET);
+	timeout = 0;
+	while (++timeout < 500) {
+		val = acp62_readl(acp_base + ACP_SOFT_RESET);
+		if (!val)
+			return 0;
+		cpu_relax();
+	}
+	return -ETIMEDOUT;
+}
+
+static void acp62_enable_interrupts(void __iomem *acp_base)
+{
+	acp62_writel(1, acp_base + ACP_EXTERNAL_INTR_ENB);
+}
+
+static void acp62_disable_interrupts(void __iomem *acp_base)
+{
+	acp62_writel(ACP_EXT_INTR_STAT_CLEAR_MASK, acp_base +
+		     ACP_EXTERNAL_INTR_STAT);
+	acp62_writel(0, acp_base + ACP_EXTERNAL_INTR_CNTL);
+	acp62_writel(0, acp_base + ACP_EXTERNAL_INTR_ENB);
+}
+
+static int acp62_init(void __iomem *acp_base, struct device *dev)
+{
+	int ret;
+
+	ret = acp62_power_on(acp_base);
+	if (ret) {
+		dev_err(dev, "ACP power on failed\n");
+		return ret;
+	}
+	acp62_writel(0x01, acp_base + ACP_CONTROL);
+	ret = acp62_reset(acp_base);
+	if (ret) {
+		dev_err(dev, "ACP reset failed\n");
+		return ret;
+	}
+	acp62_writel(0x03, acp_base + ACP_CLKMUX_SEL);
+	acp62_enable_interrupts(acp_base);
+	return 0;
+}
+
+static int acp62_deinit(void __iomem *acp_base, struct device *dev)
+{
+	int ret;
+
+	acp62_disable_interrupts(acp_base);
+	ret = acp62_reset(acp_base);
+	if (ret) {
+		dev_err(dev, "ACP reset failed\n");
+		return ret;
+	}
+	acp62_writel(0, acp_base + ACP_CLKMUX_SEL);
+	acp62_writel(0, acp_base + ACP_CONTROL);
+	return 0;
+}
+
+static irqreturn_t acp62_irq_handler(int irq, void *dev_id)
+{
+	struct acp62_dev_data *adata;
+	struct pdm_dev_data *ps_pdm_data;
+	u32 val;
+
+	adata = dev_id;
+	if (!adata)
+		return IRQ_NONE;
+
+	val = acp62_readl(adata->acp62_base + ACP_EXTERNAL_INTR_STAT);
+	if (val & BIT(PDM_DMA_STAT)) {
+		ps_pdm_data = dev_get_drvdata(&adata->pdev[0]->dev);
+		acp62_writel(BIT(PDM_DMA_STAT), adata->acp62_base + ACP_EXTERNAL_INTR_STAT);
+		if (ps_pdm_data->capture_stream)
+			snd_pcm_period_elapsed(ps_pdm_data->capture_stream);
+		return IRQ_HANDLED;
+	}
+	return IRQ_NONE;
+}
+
+static int snd_acp62_probe(struct pci_dev *pci,
+			   const struct pci_device_id *pci_id)
+{
+	struct acp62_dev_data *adata;
+	struct platform_device_info pdevinfo[ACP6x_DEVS];
+	int index, ret;
+	int val = 0x00;
+	struct acpi_device *adev;
+	const union acpi_object *obj;
+	u32 addr;
+	unsigned int irqflags;
+
+	irqflags = IRQF_SHARED;
+	/* Pink Sardine device check */
+	switch (pci->revision) {
+	case 0x63:
+		break;
+	default:
+		dev_dbg(&pci->dev, "acp62 pci device not found\n");
+		return -ENODEV;
+	}
+	if (pci_enable_device(pci)) {
+		dev_err(&pci->dev, "pci_enable_device failed\n");
+		return -ENODEV;
+	}
+
+	ret = pci_request_regions(pci, "AMD ACP6.2 audio");
+	if (ret < 0) {
+		dev_err(&pci->dev, "pci_request_regions failed\n");
+		goto disable_pci;
+	}
+	adata = devm_kzalloc(&pci->dev, sizeof(struct acp62_dev_data),
+			     GFP_KERNEL);
+	if (!adata) {
+		ret = -ENOMEM;
+		goto release_regions;
+	}
+
+	addr = pci_resource_start(pci, 0);
+	adata->acp62_base = devm_ioremap(&pci->dev, addr,
+					 pci_resource_len(pci, 0));
+	if (!adata->acp62_base) {
+		ret = -ENOMEM;
+		goto release_regions;
+	}
+	pci_set_master(pci);
+	pci_set_drvdata(pci, adata);
+	ret = acp62_init(adata->acp62_base, &pci->dev);
+	if (ret)
+		goto release_regions;
+	val = acp62_readl(adata->acp62_base + ACP_PIN_CONFIG);
+	switch (val) {
+	case ACP_CONFIG_0:
+	case ACP_CONFIG_1:
+	case ACP_CONFIG_2:
+	case ACP_CONFIG_3:
+	case ACP_CONFIG_9:
+	case ACP_CONFIG_15:
+		dev_info(&pci->dev, "Audio Mode %d\n", val);
+		break;
+	default:
+
+		/* Checking DMIC hardware*/
+		adev = acpi_find_child_device(ACPI_COMPANION(&pci->dev), 0x02, 0);
+
+		if (!adev)
+			break;
+
+		if (!acpi_dev_get_property(adev, "acp-audio-device-type",
+					   ACPI_TYPE_INTEGER, &obj) &&
+					   obj->integer.value == 2) {
+			adata->res = devm_kzalloc(&pci->dev, sizeof(struct resource), GFP_KERNEL);
+			if (!adata->res) {
+				ret = -ENOMEM;
+				goto de_init;
+			}
+
+			adata->res->name = "acp_iomem";
+			adata->res->flags = IORESOURCE_MEM;
+			adata->res->start = addr;
+			adata->res->end = addr + (ACP6x_REG_END - ACP6x_REG_START);
+			adata->acp62_audio_mode = ACP6x_PDM_MODE;
+
+			memset(&pdevinfo, 0, sizeof(pdevinfo));
+			pdevinfo[0].name = "acp_ps_pdm_dma";
+			pdevinfo[0].id = 0;
+			pdevinfo[0].parent = &pci->dev;
+			pdevinfo[0].num_res = 1;
+			pdevinfo[0].res = adata->res;
+
+			pdevinfo[1].name = "dmic-codec";
+			pdevinfo[1].id = 0;
+			pdevinfo[1].parent = &pci->dev;
+
+			pdevinfo[2].name = "acp_ps_mach";
+			pdevinfo[2].id = 0;
+			pdevinfo[2].parent = &pci->dev;
+
+			for (index = 0; index < ACP6x_DEVS; index++) {
+				adata->pdev[index] =
+					platform_device_register_full(&pdevinfo[index]);
+
+				if (IS_ERR(adata->pdev[index])) {
+					dev_err(&pci->dev,
+						"cannot register %s device\n",
+						 pdevinfo[index].name);
+					ret = PTR_ERR(adata->pdev[index]);
+					goto unregister_devs;
+				}
+				ret = devm_request_irq(&pci->dev, pci->irq, acp62_irq_handler,
+						       irqflags, "ACP_PCI_IRQ", adata);
+				if (ret) {
+					dev_err(&pci->dev, "ACP PCI IRQ request failed\n");
+					goto unregister_devs;
+				}
+			}
+		}
+		break;
+	}
+	pm_runtime_set_autosuspend_delay(&pci->dev, ACP_SUSPEND_DELAY_MS);
+	pm_runtime_use_autosuspend(&pci->dev);
+	pm_runtime_put_noidle(&pci->dev);
+	pm_runtime_allow(&pci->dev);
+	return 0;
+unregister_devs:
+	for (--index; index >= 0; index--)
+		platform_device_unregister(adata->pdev[index]);
+de_init:
+	if (acp62_deinit(adata->acp62_base, &pci->dev))
+		dev_err(&pci->dev, "ACP de-init failed\n");
+release_regions:
+	pci_release_regions(pci);
+disable_pci:
+	pci_disable_device(pci);
+
+	return ret;
+}
+
+static int __maybe_unused snd_acp62_suspend(struct device *dev)
+{
+	struct acp62_dev_data *adata;
+	int ret;
+
+	adata = dev_get_drvdata(dev);
+	ret = acp62_deinit(adata->acp62_base, dev);
+	if (ret)
+		dev_err(dev, "ACP de-init failed\n");
+	return ret;
+}
+
+static int __maybe_unused snd_acp62_resume(struct device *dev)
+{
+	struct acp62_dev_data *adata;
+	int ret;
+
+	adata = dev_get_drvdata(dev);
+	ret = acp62_init(adata->acp62_base, dev);
+	if (ret)
+		dev_err(dev, "ACP init failed\n");
+	return ret;
+}
+
+static const struct dev_pm_ops acp62_pm_ops = {
+	SET_RUNTIME_PM_OPS(snd_acp62_suspend, snd_acp62_resume, NULL)
+	SET_SYSTEM_SLEEP_PM_OPS(snd_acp62_suspend, snd_acp62_resume)
+};
+
+static void snd_acp62_remove(struct pci_dev *pci)
+{
+	struct acp62_dev_data *adata;
+	int ret, index;
+
+	adata = pci_get_drvdata(pci);
+	if (adata->acp62_audio_mode == ACP6x_PDM_MODE) {
+		for (index = 0; index < ACP6x_DEVS; index++)
+			platform_device_unregister(adata->pdev[index]);
+	}
+	ret = acp62_deinit(adata->acp62_base, &pci->dev);
+	if (ret)
+		dev_err(&pci->dev, "ACP de-init failed\n");
+	pm_runtime_forbid(&pci->dev);
+	pm_runtime_get_noresume(&pci->dev);
+	pci_release_regions(pci);
+	pci_disable_device(pci);
+}
+
+static const struct pci_device_id snd_acp62_ids[] = {
+	{ PCI_DEVICE(PCI_VENDOR_ID_AMD, ACP_DEVICE_ID),
+	.class = PCI_CLASS_MULTIMEDIA_OTHER << 8,
+	.class_mask = 0xffffff },
+	{ 0, },
+};
+MODULE_DEVICE_TABLE(pci, snd_acp62_ids);
+
+static struct pci_driver ps_acp62_driver  = {
+	.name = KBUILD_MODNAME,
+	.id_table = snd_acp62_ids,
+	.probe = snd_acp62_probe,
+	.remove = snd_acp62_remove,
+	.driver = {
+		.pm = &acp62_pm_ops,
+	}
+};
+
+module_pci_driver(ps_acp62_driver);
+
+MODULE_AUTHOR("Vijendar.Mukunda@amd.com");
+MODULE_AUTHOR("Syed.SabaKareem@amd.com");
+MODULE_DESCRIPTION("AMD ACP Pink Sardine PCI driver");
+MODULE_LICENSE("GPL v2");

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

+// SPDX-License-Identifier: GPL-2.0+
+/*
+ * Machine driver for AMD Pink Sardine platform using DMIC
+ *
+ * Copyright 2022 Advanced Micro Devices, Inc.
+ */
+
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <linux/module.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <linux/io.h>
+#include <linux/dmi.h>
+
+#include "acp62.h"
+
+#define DRV_NAME "acp_ps_mach"
+
+SND_SOC_DAILINK_DEF(acp62_pdm,
+		    DAILINK_COMP_ARRAY(COMP_CPU("acp_ps_pdm_dma.0")));
+
+SND_SOC_DAILINK_DEF(dmic_codec,
+		    DAILINK_COMP_ARRAY(COMP_CODEC("dmic-codec.0",
+						  "dmic-hifi")));
+
+SND_SOC_DAILINK_DEF(pdm_platform,
+		    DAILINK_COMP_ARRAY(COMP_PLATFORM("acp_ps_pdm_dma.0")));
+
+static struct snd_soc_dai_link acp62_dai_pdm[] = {
+	{
+		.name = "acp62-dmic-capture",
+		.stream_name = "DMIC capture",
+		.capture_only = 1,
+		SND_SOC_DAILINK_REG(acp62_pdm, dmic_codec, pdm_platform),
+	},
+};
+
+static struct snd_soc_card acp62_card = {
+	.name = "acp62",
+	.owner = THIS_MODULE,
+	.dai_link = acp62_dai_pdm,
+	.num_links = 1,
+};
+
+static int acp62_probe(struct platform_device *pdev)
+{
+	struct acp62_pdm *machine = NULL;
+	struct snd_soc_card *card;
+	int ret;
+
+	platform_set_drvdata(pdev, &acp62_card);
+	card = platform_get_drvdata(pdev);
+	acp62_card.dev = &pdev->dev;
+
+	snd_soc_card_set_drvdata(card, machine);
+	ret = devm_snd_soc_register_card(&pdev->dev, card);
+	if (ret) {
+		return dev_err_probe(&pdev->dev, ret,
+				"snd_soc_register_card(%s) failed\n",
+				card->name);
+	}
+
+	return 0;
+}
+
+static struct platform_driver acp62_mach_driver = {
+	.driver = {
+		.name = "acp_ps_mach",
+		.pm = &snd_soc_pm_ops,
+	},
+	.probe = acp62_probe,
+};
+
+module_platform_driver(acp62_mach_driver);
+
+MODULE_AUTHOR("Syed.SabaKareem@amd.com");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:" DRV_NAME);

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

@@ -172,7 +172,7 @@ static int acp5x_cs35l41_hw_params(struct snd_pcm_substream *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 num_codecs = rtd->dai_link->num_codecs;
 	unsigned int bclk_val;
 
 	ret = 0;

sound/soc/amd/yc/acp6x-mach.c

@@ -171,6 +171,20 @@ static const struct dmi_system_id yc_acp_quirk_table[] = {
 			DMI_MATCH(DMI_PRODUCT_NAME, "21J6"),
 		}
 	},
+	{
+		.driver_data = &acp6x_card,
+		.matches = {
+			DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"),
+			DMI_MATCH(DMI_PRODUCT_NAME, "82"),
+		}
+	},
+	{
+		.driver_data = &acp6x_card,
+		.matches = {
+			DMI_MATCH(DMI_BOARD_VENDOR, "ASUSTeK COMPUTER INC."),
+			DMI_MATCH(DMI_PRODUCT_NAME, "UM5302TA"),
+		}
+	},
 	{}
 };
 

sound/soc/apple/Kconfig

+config SND_SOC_APPLE_MCA
+	tristate "Apple Silicon MCA driver"
+	depends on ARCH_APPLE || COMPILE_TEST
+	select SND_DMAENGINE_PCM
+	default ARCH_APPLE
+	help
+	  This option enables an ASoC platform driver for MCA peripherals found
+	  on Apple Silicon SoCs.

sound/soc/apple/Makefile

+snd-soc-apple-mca-objs	:= mca.o
+
+obj-$(CONFIG_SND_SOC_APPLE_MCA)	+= snd-soc-apple-mca.o

sound/soc/atmel/atmel_ssc_dai.c

@@ -891,7 +891,6 @@ static int asoc_ssc_init(struct device *dev)
 int atmel_ssc_set_audio(int ssc_id)
 {
 	struct ssc_device *ssc;
-	int ret;
 
 	/* If we can grab the SSC briefly to parent the DAI device off it */
 	ssc = ssc_request(ssc_id);
@@ -903,9 +902,7 @@ int atmel_ssc_set_audio(int ssc_id)
 		ssc_info[ssc_id].ssc = ssc;
 	}
 
-	ret = asoc_ssc_init(&ssc->pdev->dev);
-
-	return ret;
+	return asoc_ssc_init(&ssc->pdev->dev);
 }
 EXPORT_SYMBOL_GPL(atmel_ssc_set_audio);
 

sound/soc/atmel/sam9g20_wm8731.c

@@ -172,7 +172,7 @@ static int at91sam9g20ek_audio_probe(struct platform_device *pdev)
 	ret = snd_soc_register_card(card);
 	if (ret) {
 		dev_err_probe(&pdev->dev, ret,
-			      "snd_soc_register_card() failed: %d\n", ret);
+			      "snd_soc_register_card() failed\n");
 		goto err;
 	}
 

sound/soc/bcm/bcm2835-i2s.c

@@ -841,14 +841,9 @@ static int bcm2835_i2s_probe(struct platform_device *pdev)
 	/* get the clock */
 	dev->clk_prepared = false;
 	dev->clk = devm_clk_get(&pdev->dev, NULL);
-	if (IS_ERR(dev->clk)) {
-		ret = PTR_ERR(dev->clk);
-		if (ret == -EPROBE_DEFER)
-			dev_dbg(&pdev->dev, "could not get clk: %d\n", ret);
-		else
-			dev_err(&pdev->dev, "could not get clk: %d\n", ret);
-		return ret;
-	}
+	if (IS_ERR(dev->clk))
+		return dev_err_probe(&pdev->dev, PTR_ERR(dev->clk),
+				     "could not get clk\n");
 
 	/* Request ioarea */
 	base = devm_platform_ioremap_resource(pdev, 0);

sound/soc/codecs/Kconfig

@@ -98,6 +98,7 @@ config SND_SOC_ALL_CODECS
 	imply SND_SOC_DA9055
 	imply SND_SOC_DMIC
 	imply SND_SOC_ES8316
+	imply SND_SOC_ES8326
 	imply SND_SOC_ES8328_SPI
 	imply SND_SOC_ES8328_I2C
 	imply SND_SOC_ES7134
@@ -205,6 +206,7 @@ config SND_SOC_ALL_CODECS
 	imply SND_SOC_SIMPLE_AMPLIFIER
 	imply SND_SOC_SIMPLE_MUX
 	imply SND_SOC_SPDIF
+	imply SND_SOC_SRC4XXX_I2C
 	imply SND_SOC_SSM2305
 	imply SND_SOC_SSM2518
 	imply SND_SOC_SSM2602_SPI
@@ -608,7 +610,7 @@ config SND_SOC_BT_SCO
 
 config SND_SOC_CPCAP
 	tristate "Motorola CPCAP codec"
-	depends on MFD_CPCAP
+	depends on MFD_CPCAP || COMPILE_TEST
 
 config SND_SOC_CQ0093VC
 	tristate
@@ -690,9 +692,15 @@ config SND_SOC_CS35L45_I2C
 	  Enable support for Cirrus Logic CS35L45 smart speaker amplifier
 	  with I2C control.
 
+config SND_SOC_CS42L42_CORE
+	tristate
+
 config SND_SOC_CS42L42
-	tristate "Cirrus Logic CS42L42 CODEC"
+	tristate "Cirrus Logic CS42L42 CODEC (I2C)"
 	depends on I2C
+	select REGMAP
+	select REGMAP_I2C
+	select SND_SOC_CS42L42_CORE
 
 config SND_SOC_CS42L51
 	tristate
@@ -714,6 +722,13 @@ config SND_SOC_CS42L73
 	tristate "Cirrus Logic CS42L73 CODEC"
 	depends on I2C
 
+config SND_SOC_CS42L83
+	tristate "Cirrus Logic CS42L83 CODEC"
+	depends on I2C
+	select REGMAP
+	select REGMAP_I2C
+	select SND_SOC_CS42L42_CORE
+
 config SND_SOC_CS4234
 	tristate "Cirrus Logic CS4234 CODEC"
 	depends on I2C
@@ -913,6 +928,10 @@ config SND_SOC_ES8316
 	tristate "Everest Semi ES8316 CODEC"
 	depends on I2C
 
+config SND_SOC_ES8326
+	tristate "Everest Semi ES8326 CODEC"
+	depends on I2C
+
 config SND_SOC_ES8328
 	tristate
 
@@ -966,7 +985,7 @@ config SND_SOC_LM49453
 
 config SND_SOC_LOCHNAGAR_SC
 	tristate "Lochnagar Sound Card"
-	depends on MFD_LOCHNAGAR
+	depends on MFD_LOCHNAGAR || COMPILE_TEST
 	help
 	  This driver support the sound card functionality of the Cirrus
 	  Logic Lochnagar audio development board.
@@ -1191,8 +1210,7 @@ config SND_SOC_RK3328
 
 config SND_SOC_RK817
 	tristate "Rockchip RK817 audio CODEC"
-	depends on MFD_RK808
-	select REGMAP_I2C
+	depends on MFD_RK808 || COMPILE_TEST
 
 config SND_SOC_RL6231
 	tristate
@@ -1471,6 +1489,18 @@ config SND_SOC_SIMPLE_MUX
 config SND_SOC_SPDIF
 	tristate "S/PDIF CODEC"
 
+config SND_SOC_SRC4XXX_I2C
+	tristate "Texas Instruments SRC4XXX DIR/DIT and SRC codecs"
+	depends on I2C
+	select SND_SOC_SRC4XXX
+	help
+	  Enable support for the TI SRC4XXX family of codecs. These include the
+	  scr4392 which has digital receivers, transmitters, and
+	  a sample rate converter, including numerous ports.
+
+config SND_SOC_SRC4XXX
+	tristate
+
 config SND_SOC_SSM2305
 	tristate "Analog Devices SSM2305 Class-D Amplifier"
 	help
@@ -1726,8 +1756,10 @@ config SND_SOC_WCD_MBHC
 config SND_SOC_WCD934X
 	tristate "WCD9340/WCD9341 Codec"
 	depends on COMMON_CLK
+	depends on SLIMBUS
+	select REGMAP_SLIMBUS
 	select SND_SOC_WCD_MBHC
-	depends on MFD_WCD934X
+	depends on MFD_WCD934X || COMPILE_TEST
 	help
 	  The WCD9340/9341 is a audio codec IC Integrated in
 	  Qualcomm SoCs like SDM845.

sound/soc/codecs/Makefile

@@ -65,11 +65,13 @@ snd-soc-cs35l45-objs := cs35l45.o
 snd-soc-cs35l45-spi-objs := cs35l45-spi.o
 snd-soc-cs35l45-i2c-objs := cs35l45-i2c.o
 snd-soc-cs42l42-objs := cs42l42.o
+snd-soc-cs42l42-i2c-objs := cs42l42-i2c.o
 snd-soc-cs42l51-objs := cs42l51.o
 snd-soc-cs42l51-i2c-objs := cs42l51-i2c.o
 snd-soc-cs42l52-objs := cs42l52.o
 snd-soc-cs42l56-objs := cs42l56.o
 snd-soc-cs42l73-objs := cs42l73.o
+snd-soc-cs42l83-i2c-objs := cs42l83-i2c.o
 snd-soc-cs4234-objs := cs4234.o
 snd-soc-cs4265-objs := cs4265.o
 snd-soc-cs4270-objs := cs4270.o
@@ -100,6 +102,7 @@ snd-soc-dmic-objs := dmic.o
 snd-soc-es7134-objs := es7134.o
 snd-soc-es7241-objs := es7241.o
 snd-soc-es8316-objs := es8316.o
+snd-soc-es8326-objs := es8326.o
 snd-soc-es8328-objs := es8328.o
 snd-soc-es8328-i2c-objs := es8328-i2c.o
 snd-soc-es8328-spi-objs := es8328-spi.o
@@ -231,6 +234,8 @@ snd-soc-sigmadsp-regmap-objs := sigmadsp-regmap.o
 snd-soc-si476x-objs := si476x.o
 snd-soc-spdif-tx-objs := spdif_transmitter.o
 snd-soc-spdif-rx-objs := spdif_receiver.o
+snd-soc-src4xxx-objs := src4xxx.o
+snd-soc-src4xxx-i2c-objs := src4xxx-i2c.o
 snd-soc-ssm2305-objs := ssm2305.o
 snd-soc-ssm2518-objs := ssm2518.o
 snd-soc-ssm2602-objs := ssm2602.o
@@ -419,12 +424,14 @@ obj-$(CONFIG_SND_SOC_CS35L45_TABLES)	+= snd-soc-cs35l45-tables.o
 obj-$(CONFIG_SND_SOC_CS35L45)	+= snd-soc-cs35l45.o
 obj-$(CONFIG_SND_SOC_CS35L45_SPI)	+= snd-soc-cs35l45-spi.o
 obj-$(CONFIG_SND_SOC_CS35L45_I2C)	+= snd-soc-cs35l45-i2c.o
-obj-$(CONFIG_SND_SOC_CS42L42)	+= snd-soc-cs42l42.o
+obj-$(CONFIG_SND_SOC_CS42L42_CORE)	+= snd-soc-cs42l42.o
+obj-$(CONFIG_SND_SOC_CS42L42)	+= snd-soc-cs42l42-i2c.o
 obj-$(CONFIG_SND_SOC_CS42L51)	+= snd-soc-cs42l51.o
 obj-$(CONFIG_SND_SOC_CS42L51_I2C)	+= snd-soc-cs42l51-i2c.o
 obj-$(CONFIG_SND_SOC_CS42L52)	+= snd-soc-cs42l52.o
 obj-$(CONFIG_SND_SOC_CS42L56)	+= snd-soc-cs42l56.o
 obj-$(CONFIG_SND_SOC_CS42L73)	+= snd-soc-cs42l73.o
+obj-$(CONFIG_SND_SOC_CS42L83)	+= snd-soc-cs42l83-i2c.o
 obj-$(CONFIG_SND_SOC_CS4234)	+= snd-soc-cs4234.o
 obj-$(CONFIG_SND_SOC_CS4265)	+= snd-soc-cs4265.o
 obj-$(CONFIG_SND_SOC_CS4270)	+= snd-soc-cs4270.o
@@ -455,6 +462,7 @@ obj-$(CONFIG_SND_SOC_DMIC)	+= snd-soc-dmic.o
 obj-$(CONFIG_SND_SOC_ES7134)	+= snd-soc-es7134.o
 obj-$(CONFIG_SND_SOC_ES7241)	+= snd-soc-es7241.o
 obj-$(CONFIG_SND_SOC_ES8316)    += snd-soc-es8316.o
+obj-$(CONFIG_SND_SOC_ES8326)    += snd-soc-es8326.o
 obj-$(CONFIG_SND_SOC_ES8328)	+= snd-soc-es8328.o
 obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o
 obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o
@@ -579,6 +587,8 @@ obj-$(CONFIG_SND_SOC_SIGMADSP_I2C)	+= snd-soc-sigmadsp-i2c.o
 obj-$(CONFIG_SND_SOC_SIGMADSP_REGMAP)	+= snd-soc-sigmadsp-regmap.o
 obj-$(CONFIG_SND_SOC_SI476X)	+= snd-soc-si476x.o
 obj-$(CONFIG_SND_SOC_SPDIF)	+= snd-soc-spdif-rx.o snd-soc-spdif-tx.o
+obj-$(CONFIG_SND_SOC_SRC4XXX)	+= snd-soc-src4xxx.o
+obj-$(CONFIG_SND_SOC_SRC4XXX_I2C)	+= snd-soc-src4xxx-i2c.o
 obj-$(CONFIG_SND_SOC_SSM2305)	+= snd-soc-ssm2305.o
 obj-$(CONFIG_SND_SOC_SSM2518)	+= snd-soc-ssm2518.o
 obj-$(CONFIG_SND_SOC_SSM2602)	+= snd-soc-ssm2602.o

sound/soc/codecs/ak4458.c

@@ -447,6 +447,13 @@ static int ak4458_hw_params(struct snd_pcm_substream *substream,
 	snd_soc_component_update_bits(component, AK4458_0B_CONTROL7,
 				      AK4458_DCHAIN_MASK, dchn);
 
+	if (ak4458->drvdata->type == AK4497) {
+		ret = snd_soc_component_update_bits(component, AK4458_09_DSD2,
+						    0x4, (ak4458->dsd_path << 2));
+		if (ret < 0)
+			return ret;
+	}
+
 	ret = ak4458_rstn_control(component, 0);
 	if (ret)
 		return ret;
@@ -629,48 +636,6 @@ static void ak4458_reset(struct ak4458_priv *ak4458, bool active)
 	}
 }
 
-static int ak4458_init(struct snd_soc_component *component)
-{
-	struct ak4458_priv *ak4458 = snd_soc_component_get_drvdata(component);
-	int ret;
-
-	/* External Mute ON */
-	if (ak4458->mute_gpiod)
-		gpiod_set_value_cansleep(ak4458->mute_gpiod, 1);
-
-	ak4458_reset(ak4458, false);
-
-	ret = snd_soc_component_update_bits(component, AK4458_00_CONTROL1,
-			    0x80, 0x80);   /* ACKS bit = 1; 10000000 */
-	if (ret < 0)
-		return ret;
-
-	if (ak4458->drvdata->type == AK4497) {
-		ret = snd_soc_component_update_bits(component, AK4458_09_DSD2,
-						    0x4, (ak4458->dsd_path << 2));
-		if (ret < 0)
-			return ret;
-	}
-
-	return ak4458_rstn_control(component, 1);
-}
-
-static int ak4458_probe(struct snd_soc_component *component)
-{
-	struct ak4458_priv *ak4458 = snd_soc_component_get_drvdata(component);
-
-	ak4458->fs = 48000;
-
-	return ak4458_init(component);
-}
-
-static void ak4458_remove(struct snd_soc_component *component)
-{
-	struct ak4458_priv *ak4458 = snd_soc_component_get_drvdata(component);
-
-	ak4458_reset(ak4458, true);
-}
-
 #ifdef CONFIG_PM
 static int __maybe_unused ak4458_runtime_suspend(struct device *dev)
 {
@@ -714,8 +679,6 @@ static int __maybe_unused ak4458_runtime_resume(struct device *dev)
 #endif /* CONFIG_PM */
 
 static const struct snd_soc_component_driver soc_codec_dev_ak4458 = {
-	.probe			= ak4458_probe,
-	.remove			= ak4458_remove,
 	.controls		= ak4458_snd_controls,
 	.num_controls		= ARRAY_SIZE(ak4458_snd_controls),
 	.dapm_widgets		= ak4458_dapm_widgets,
@@ -728,8 +691,6 @@ static const struct snd_soc_component_driver soc_codec_dev_ak4458 = {
 };
 
 static const struct snd_soc_component_driver soc_codec_dev_ak4497 = {
-	.probe			= ak4458_probe,
-	.remove			= ak4458_remove,
 	.controls		= ak4497_snd_controls,
 	.num_controls		= ARRAY_SIZE(ak4497_snd_controls),
 	.dapm_widgets		= ak4497_dapm_widgets,
@@ -820,12 +781,16 @@ static int ak4458_i2c_probe(struct i2c_client *i2c)
 
 	pm_runtime_enable(&i2c->dev);
 	regcache_cache_only(ak4458->regmap, true);
+	ak4458_reset(ak4458, false);
 
 	return 0;
 }
 
 static int ak4458_i2c_remove(struct i2c_client *i2c)
 {
+	struct ak4458_priv *ak4458 = i2c_get_clientdata(i2c);
+
+	ak4458_reset(ak4458, true);
 	pm_runtime_disable(&i2c->dev);
 
 	return 0;

sound/soc/codecs/cs42l42-i2c.c

+// SPDX-License-Identifier: GPL-2.0-only
+/*
+ * cs42l42-i2c.c -- CS42L42 ALSA SoC audio driver for I2C
+ *
+ * Copyright 2016, 2022 Cirrus Logic, Inc.
+ */
+
+#include <linux/i2c.h>
+#include <linux/module.h>
+#include <linux/regmap.h>
+#include <linux/slab.h>
+#include <linux/types.h>
+
+#include "cs42l42.h"
+
+static int cs42l42_i2c_probe(struct i2c_client *i2c_client)
+{
+	struct device *dev = &i2c_client->dev;
+	struct cs42l42_private *cs42l42;
+	struct regmap *regmap;
+	int ret;
+
+	cs42l42 = devm_kzalloc(dev, sizeof(*cs42l42), GFP_KERNEL);
+	if (!cs42l42)
+		return -ENOMEM;
+
+	regmap = devm_regmap_init_i2c(i2c_client, &cs42l42_regmap);
+	if (IS_ERR(regmap))
+		return dev_err_probe(&i2c_client->dev, PTR_ERR(regmap),
+				     "regmap_init() failed\n");
+
+	cs42l42->devid = CS42L42_CHIP_ID;
+	cs42l42->dev = dev;
+	cs42l42->regmap = regmap;
+	cs42l42->irq = i2c_client->irq;
+
+	ret = cs42l42_common_probe(cs42l42, &cs42l42_soc_component, &cs42l42_dai);
+	if (ret)
+		return ret;
+
+	return cs42l42_init(cs42l42);
+}
+
+static int cs42l42_i2c_remove(struct i2c_client *i2c_client)
+{
+	struct cs42l42_private *cs42l42 = dev_get_drvdata(&i2c_client->dev);
+
+	cs42l42_common_remove(cs42l42);
+
+	return 0;
+}
+
+static int __maybe_unused cs42l42_i2c_resume(struct device *dev)
+{
+	int ret;
+
+	ret = cs42l42_resume(dev);
+	if (ret)
+		return ret;
+
+	cs42l42_resume_restore(dev);
+
+	return 0;
+}
+
+static const struct dev_pm_ops cs42l42_i2c_pm_ops = {
+	SET_SYSTEM_SLEEP_PM_OPS(cs42l42_suspend, cs42l42_i2c_resume)
+};
+
+static const struct of_device_id __maybe_unused cs42l42_of_match[] = {
+	{ .compatible = "cirrus,cs42l42", },
+	{}
+};
+MODULE_DEVICE_TABLE(of, cs42l42_of_match);
+
+static const struct acpi_device_id __maybe_unused cs42l42_acpi_match[] = {
+	{"10134242", 0,},
+	{}
+};
+MODULE_DEVICE_TABLE(acpi, cs42l42_acpi_match);
+
+static const struct i2c_device_id cs42l42_id[] = {
+	{"cs42l42", 0},
+	{}
+};
+
+MODULE_DEVICE_TABLE(i2c, cs42l42_id);
+
+static struct i2c_driver cs42l42_i2c_driver = {
+	.driver = {
+		.name = "cs42l42",
+		.pm = &cs42l42_i2c_pm_ops,
+		.of_match_table = of_match_ptr(cs42l42_of_match),
+		.acpi_match_table = ACPI_PTR(cs42l42_acpi_match),
+		},
+	.id_table = cs42l42_id,
+	.probe_new = cs42l42_i2c_probe,
+	.remove = cs42l42_i2c_remove,
+};
+
+module_i2c_driver(cs42l42_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC CS42L42 I2C driver");
+MODULE_AUTHOR("Richard Fitzgerald <rf@opensource.cirrus.com>");
+MODULE_LICENSE("GPL");
+MODULE_IMPORT_NS(SND_SOC_CS42L42_CORE);