Merge remote-tracking branch 'asoc/for-5.12' into asoc-linus

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

@@ -71,9 +71,6 @@ properties:
             description: CPU to Codec rate channels.
             $ref: /schemas/types.yaml#/definitions/uint32
 
-        required:
-          - remote-endpoint
-
   ports:
     description: multi OF-Graph subnode
     type: object

Documentation/devicetree/bindings/sound/ingenic,codec.yaml

@@ -15,9 +15,14 @@ properties:
 
   compatible:
     oneOf:
-      - const: ingenic,jz4770-codec
-      - const: ingenic,jz4725b-codec
-      - const: ingenic,jz4740-codec
+      - enum:
+          - ingenic,jz4770-codec
+          - ingenic,jz4760-codec
+          - ingenic,jz4725b-codec
+          - ingenic,jz4740-codec
+      - items:
+          - const: ingenic,jz4760b-codec
+          - const: ingenic,jz4760-codec
 
   reg:
     maxItems: 1

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

@@ -18,6 +18,7 @@ properties:
     enum:
       - intel,keembay-i2s
       - intel,keembay-tdm
+      - intel,keembay-hdmi-i2s
 
   "#sound-dai-cells":
     const: 0
@@ -45,6 +46,16 @@ properties:
       - const: osc
       - const: apb_clk
 
+  dmas:
+    items:
+      - description: DMA TX channel
+      - description: DMA RX channel
+
+  dma-names:
+    items:
+      - const: tx
+      - const: rx
+
 required:
   - compatible
   - "#sound-dai-cells"
@@ -70,4 +81,6 @@ examples:
          interrupts = <GIC_SPI 120 IRQ_TYPE_LEVEL_HIGH>;
          clock-names = "osc", "apb_clk";
          clocks = <&scmi_clk KEEM_BAY_PSS_AUX_I2S3>, <&scmi_clk KEEM_BAY_PSS_I2S3>;
+         dmas = <&axi_dma0 29 &axi_dma0 33>;
+         dma-names = "tx", "rx";
      };

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

@@ -23,6 +23,10 @@ properties:
     $ref: "/schemas/types.yaml#/definitions/phandle"
     description: The phandle of MT8192 ASoC platform.
 
+  mediatek,hdmi-codec:
+    $ref: "/schemas/types.yaml#/definitions/phandle"
+    description: The phandle of HDMI codec.
+
 additionalProperties: false
 
 required:
@@ -35,6 +39,7 @@ examples:
     sound: mt8192-sound {
         compatible = "mediatek,mt8192_mt6359_rt1015_rt5682";
         mediatek,platform = <&afe>;
+        mediatek,hdmi-codec = <&anx_bridge_dp>;
         pinctrl-names = "aud_clk_mosi_off",
                         "aud_clk_mosi_on";
         pinctrl-0 = <&aud_clk_mosi_off>;

Documentation/devicetree/bindings/sound/nvidia,tegra-audio-graph-card.yaml

@@ -0,0 +1,190 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/nvidia,tegra-audio-graph-card.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Audio Graph based Tegra sound card driver
+
+description: |
+  This is based on generic audio graph card driver along with additional
+  customizations for Tegra platforms. It uses the same bindings with
+  additional standard clock DT bindings required for Tegra.
+
+maintainers:
+  - Jon Hunter <jonathanh@nvidia.com>
+  - Sameer Pujar <spujar@nvidia.com>
+
+allOf:
+  - $ref: audio-graph.yaml#
+
+properties:
+  compatible:
+    enum:
+      - nvidia,tegra210-audio-graph-card
+      - nvidia,tegra186-audio-graph-card
+
+  clocks:
+    minItems: 2
+
+  clock-names:
+    minItems: 2
+    items:
+      - const: pll_a
+      - const: plla_out0
+
+  assigned-clocks:
+    minItems: 1
+    maxItems: 3
+
+  assigned-clock-parents:
+    minItems: 1
+    maxItems: 3
+
+  assigned-clock-rates:
+    minItems: 1
+    maxItems: 3
+
+  iommus:
+    maxItems: 1
+
+required:
+  - clocks
+  - clock-names
+  - assigned-clocks
+  - assigned-clock-parents
+
+unevaluatedProperties: false
+
+examples:
+  - |
+    #include<dt-bindings/clock/tegra210-car.h>
+
+    tegra_sound {
+        compatible = "nvidia,tegra210-audio-graph-card";
+
+        clocks = <&tegra_car TEGRA210_CLK_PLL_A>,
+                 <&tegra_car TEGRA210_CLK_PLL_A_OUT0>;
+        clock-names = "pll_a", "plla_out0";
+
+        assigned-clocks = <&tegra_car TEGRA210_CLK_PLL_A>,
+                          <&tegra_car TEGRA210_CLK_PLL_A_OUT0>,
+                          <&tegra_car TEGRA210_CLK_EXTERN1>;
+        assigned-clock-parents = <0>, <0>, <&tegra_car TEGRA210_CLK_PLL_A_OUT0>;
+        assigned-clock-rates = <368640000>, <49152000>, <12288000>;
+
+        dais = /* FE */
+               <&admaif1_port>,
+               /* Router */
+               <&xbar_i2s1_port>,
+               /* I/O DAP Ports */
+               <&i2s1_port>;
+
+        label = "jetson-tx1-ape";
+    };
+
+    // The ports are defined for AHUB and its child devices.
+    ahub@702d0800 {
+        compatible = "nvidia,tegra210-ahub";
+        reg = <0x702d0800 0x800>;
+        clocks = <&tegra_car TEGRA210_CLK_D_AUDIO>;
+        clock-names = "ahub";
+        assigned-clocks = <&tegra_car TEGRA210_CLK_D_AUDIO>;
+        assigned-clock-parents = <&tegra_car TEGRA210_CLK_PLL_A_OUT0>;
+        #address-cells = <1>;
+        #size-cells = <1>;
+        ranges = <0x702d0000 0x702d0000 0x0000e400>;
+
+        ports {
+            #address-cells = <1>;
+            #size-cells = <0>;
+
+            port@0 {
+                reg = <0x0>;
+                xbar_admaif1_ep: endpoint {
+                    remote-endpoint = <&admaif1_ep>;
+                };
+            };
+
+            // ...
+
+            xbar_i2s1_port: port@a {
+                reg = <0xa>;
+                xbar_i2s1_ep: endpoint {
+                    remote-endpoint = <&i2s1_cif_ep>;
+                };
+            };
+        };
+
+        admaif@702d0000 {
+            compatible = "nvidia,tegra210-admaif";
+            reg = <0x702d0000 0x800>;
+            dmas = <&adma 1>,  <&adma 1>,
+                   <&adma 2>,  <&adma 2>,
+                   <&adma 3>,  <&adma 3>,
+                   <&adma 4>,  <&adma 4>,
+                   <&adma 5>,  <&adma 5>,
+                   <&adma 6>,  <&adma 6>,
+                   <&adma 7>,  <&adma 7>,
+                   <&adma 8>,  <&adma 8>,
+                   <&adma 9>,  <&adma 9>,
+                   <&adma 10>, <&adma 10>;
+            dma-names = "rx1",  "tx1",
+                        "rx2",  "tx2",
+                        "rx3",  "tx3",
+                        "rx4",  "tx4",
+                        "rx5",  "tx5",
+                        "rx6",  "tx6",
+                        "rx7",  "tx7",
+                        "rx8",  "tx8",
+                        "rx9",  "tx9",
+                        "rx10", "tx10";
+
+            ports {
+                #address-cells = <1>;
+                #size-cells = <0>;
+
+                admaif1_port: port@0 {
+                    reg = <0x0>;
+                    admaif1_ep: endpoint {
+                        remote-endpoint = <&xbar_admaif1_ep>;
+                    };
+                };
+
+                // More ADMAIF ports to follow
+            };
+        };
+
+        i2s@702d1000 {
+            compatible = "nvidia,tegra210-i2s";
+            clocks = <&tegra_car TEGRA210_CLK_I2S0>;
+            clock-names = "i2s";
+            assigned-clocks = <&tegra_car TEGRA210_CLK_I2S0>;
+            assigned-clock-parents = <&tegra_car TEGRA210_CLK_PLL_A_OUT0>;
+            assigned-clock-rates = <1536000>;
+            reg = <0x702d1000 0x100>;
+
+            ports {
+                #address-cells = <1>;
+                #size-cells = <0>;
+
+                port@0 {
+                    reg = <0x0>;
+
+                    i2s1_cif_ep: endpoint {
+                        remote-endpoint = <&xbar_i2s1_ep>;
+                    };
+                };
+
+                i2s1_port: port@1 {
+                    reg = <0x1>;
+
+                    i2s1_dap: endpoint {
+                        dai-format = "i2s";
+                    };
+                };
+            };
+        };
+    };
+
+...

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

@@ -17,6 +17,9 @@ maintainers:
   - Jon Hunter <jonathanh@nvidia.com>
   - Sameer Pujar <spujar@nvidia.com>
 
+allOf:
+  - $ref: audio-graph-port.yaml#
+
 properties:
   $nodename:
     pattern: "^dspk@[0-9a-f]*$"
@@ -55,6 +58,19 @@ properties:
       The name can be "DSPK1" or "DSPKx", where x depends on the maximum
       available instances on a Tegra SoC.
 
+  ports:
+    type: object
+    properties:
+      port@0:
+        description: |
+          DSPK ACIF (Audio Client Interface) port connected to the
+          corresponding AHUB (Audio Hub) ACIF port.
+
+      port@1:
+        description: |
+          DSPK DAP (Digital Audio Port) interface which can be connected
+          to external audio codec for playback.
+
 required:
   - compatible
   - reg
@@ -64,7 +80,7 @@ required:
   - assigned-clock-parents
   - sound-name-prefix
 
-additionalProperties: false
+unevaluatedProperties: false
 
 examples:
   - |

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

@@ -17,6 +17,9 @@ maintainers:
   - Jon Hunter <jonathanh@nvidia.com>
   - Sameer Pujar <spujar@nvidia.com>
 
+allOf:
+  - $ref: audio-graph-port.yaml#
+
 properties:
   $nodename:
     pattern: "^admaif@[0-9a-f]*$"
@@ -37,6 +40,14 @@ properties:
 
   dma-names: true
 
+  ports:
+    description: |
+      Contains list of ACIF (Audio CIF) port nodes for ADMAIF channels.
+      The number of port nodes depends on the number of ADMAIF channels
+      that SoC may have. These are interfaced with respective ACIF ports
+      in AHUB (Audio Hub). Each port is capable of data transfers in
+      both directions.
+
 if:
   properties:
     compatible:
@@ -81,7 +92,7 @@ required:
   - dmas
   - dma-names
 
-additionalProperties: false
+unevaluatedProperties: false
 
 examples:
   - |

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

@@ -17,6 +17,9 @@ maintainers:
   - Jon Hunter <jonathanh@nvidia.com>
   - Sameer Pujar <spujar@nvidia.com>
 
+allOf:
+  - $ref: audio-graph-port.yaml#
+
 properties:
   $nodename:
     pattern: "^ahub@[0-9a-f]*$"
@@ -56,6 +59,13 @@ properties:
 
   ranges: true
 
+  ports:
+    description: |
+      Contains list of ACIF (Audio CIF) port nodes for AHUB (Audio Hub).
+      These are connected to ACIF interfaces of AHUB clients. Thus the
+      number of port nodes depend on the number of clients that AHUB may
+      have depending on the SoC revision.
+
 required:
   - compatible
   - reg
@@ -67,8 +77,7 @@ required:
   - "#size-cells"
   - ranges
 
-additionalProperties:
-  type: object
+unevaluatedProperties: false
 
 examples:
   - |

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

@@ -16,6 +16,9 @@ maintainers:
   - Jon Hunter <jonathanh@nvidia.com>
   - Sameer Pujar <spujar@nvidia.com>
 
+allOf:
+  - $ref: audio-graph-port.yaml#
+
 properties:
   $nodename:
     pattern: "^dmic@[0-9a-f]*$"
@@ -56,6 +59,19 @@ properties:
       The name can be "DMIC1" or "DMIC2" ... "DMICx", where x depends
       on the maximum available instances on a Tegra SoC.
 
+  ports:
+    type: object
+    properties:
+      port@0:
+        description: |
+          DMIC ACIF (Audio Client Interface) port connected to the
+          corresponding AHUB (Audio Hub) ACIF port.
+
+      port@1:
+        description: |
+          DMIC DAP (Digital Audio Port) interface which can be connected
+          to external audio codec for capture.
+
 required:
   - compatible
   - reg
@@ -64,7 +80,7 @@ required:
   - assigned-clocks
   - assigned-clock-parents
 
-additionalProperties: false
+unevaluatedProperties: false
 
 examples:
   - |

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

@@ -16,6 +16,9 @@ maintainers:
   - Jon Hunter <jonathanh@nvidia.com>
   - Sameer Pujar <spujar@nvidia.com>
 
+allOf:
+  - $ref: audio-graph-port.yaml#
+
 properties:
   $nodename:
     pattern: "^i2s@[0-9a-f]*$"
@@ -74,6 +77,19 @@ properties:
       The name can be "I2S1" or "I2S2" ... "I2Sx", where x depends
       on the maximum available instances on a Tegra SoC.
 
+  ports:
+    type: object
+    properties:
+      port@0:
+        description: |
+          I2S ACIF (Audio Client Interface) port connected to the
+          corresponding AHUB (Audio Hub) ACIF port.
+
+      port@1:
+        description: |
+          I2S DAP (Digital Audio Port) interface which can be connected
+          to external audio codec for playback or capture.
+
 required:
   - compatible
   - reg
@@ -82,7 +98,7 @@ required:
   - assigned-clocks
   - assigned-clock-parents
 
-additionalProperties: false
+unevaluatedProperties: false
 
 examples:
   - |

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

@@ -0,0 +1,62 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/qcom,lpass-rx-macro.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: LPASS(Low Power Audio Subsystem) RX Macro audio codec DT bindings
+
+maintainers:
+  - Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
+
+properties:
+  compatible:
+    const: qcom,sm8250-lpass-rx-macro
+
+  reg:
+    maxItems: 1
+
+  "#sound-dai-cells":
+    const: 1
+
+  '#clock-cells':
+    const: 0
+
+  clocks:
+    maxItems: 5
+
+  clock-names:
+    items:
+      - const: mclk
+      - const: npl
+      - const: macro
+      - const: dcodec
+      - const: fsgen
+
+  clock-output-names:
+    items:
+      - const: mclk
+
+required:
+  - compatible
+  - reg
+  - "#sound-dai-cells"
+
+additionalProperties: false
+
+examples:
+  - |
+    #include <dt-bindings/sound/qcom,q6afe.h>
+    codec@3200000 {
+      compatible = "qcom,sm8250-lpass-rx-macro";
+      reg = <0x3200000 0x1000>;
+      #sound-dai-cells = <1>;
+      #clock-cells = <0>;
+      clocks = <&audiocc 0>,
+               <&audiocc 1>,
+               <&q6afecc LPASS_HW_MACRO_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>,
+               <&q6afecc LPASS_HW_DCODEC_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>,
+               <&vamacro>;
+      clock-names = "mclk", "npl", "macro", "dcodec", "fsgen";
+      clock-output-names = "mclk";
+    };

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

@@ -0,0 +1,67 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/qcom,lpass-tx-macro.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: LPASS(Low Power Audio Subsystem) TX Macro audio codec DT bindings
+
+maintainers:
+  - Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
+
+properties:
+  compatible:
+    const: qcom,sm8250-lpass-tx-macro
+
+  reg:
+    maxItems: 1
+
+  "#sound-dai-cells":
+    const: 1
+
+  '#clock-cells':
+    const: 0
+
+  clocks:
+    maxItems: 5
+
+  clock-names:
+    items:
+      - const: mclk
+      - const: npl
+      - const: macro
+      - const: dcodec
+      - const: fsgen
+
+  clock-output-names:
+    items:
+      - const: mclk
+
+  qcom,dmic-sample-rate:
+    description: dmic sample rate
+    $ref: /schemas/types.yaml#/definitions/uint32
+
+required:
+  - compatible
+  - reg
+  - "#sound-dai-cells"
+
+additionalProperties: false
+
+examples:
+  - |
+    #include <dt-bindings/sound/qcom,q6afe.h>
+    codec@3220000 {
+      compatible = "qcom,sm8250-lpass-tx-macro";
+      reg = <0x3220000 0x1000>;
+      #sound-dai-cells = <1>;
+      #clock-cells = <0>;
+      clocks = <&aoncc 0>,
+               <&aoncc 1>,
+               <&q6afecc LPASS_HW_MACRO_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>,
+               <&q6afecc LPASS_HW_DCODEC_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>,
+               <&vamacro>;
+      clock-names = "mclk", "npl", "macro", "dcodec", "fsgen";
+      clock-output-names = "mclk";
+      qcom,dmic-sample-rate = <600000>;
+    };

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

@@ -404,7 +404,7 @@ examples:
         /* DAI base */
         rcar_sound,dai {
             dai0 {
-                playback = <&ssi5 &src5>;
+                playback = <&ssi5>, <&src5>;
                 capture  = <&ssi6>;
             };
             dai1 {
@@ -430,8 +430,8 @@ examples:
                 bitclock-master = <&rsnd_endpoint0>;
                 frame-master = <&rsnd_endpoint0>;
 
-                playback = <&ssi0 &src0 &dvc0>;
-                capture  = <&ssi1 &src1 &dvc1>;
+                playback = <&ssi0>, <&src0>, <&dvc0>;
+                capture  = <&ssi1>, <&src1>, <&dvc1>;
             };
         };
     };

Documentation/devicetree/bindings/sound/rt5659.txt

@@ -37,10 +37,21 @@ Optional properties:
 - realtek,jd-src
   0: No JD is used
   1: using JD3 as JD source
+  2: JD source for Intel HDA header
 
 - realtek,ldo1-en-gpios : The GPIO that controls the CODEC's LDO1_EN pin.
 - realtek,reset-gpios : The GPIO that controls the CODEC's RESET pin.
 
+- sound-name-prefix: Please refer to name-prefix.txt
+
+- ports: A Codec may have a single or multiple I2S interfaces. These
+  interfaces on Codec side can be described under 'ports' or 'port'.
+  When the SoC or host device is connected to multiple interfaces of
+  the Codec, the connectivity can be described using 'ports' property.
+  If a single interface is used, then 'port' can be used. The usage
+  depends on the platform or board design.
+  Please refer to Documentation/devicetree/bindings/graph.txt
+
 Pins on the device (for linking into audio routes) for RT5659/RT5658:
 
   * DMIC L1

Documentation/devicetree/bindings/sound/sirf-audio-codec.txt

@@ -1,17 +0,0 @@
-SiRF internal audio CODEC
-
-Required properties:
-
-  - compatible : "sirf,atlas6-audio-codec" or "sirf,prima2-audio-codec"
-
-  - reg : the register address of the device.
-
-  - clocks: the clock of SiRF internal audio codec
-
-Example:
-
-audiocodec: audiocodec@b0040000 {
-	compatible = "sirf,atlas6-audio-codec";
-	reg = <0xb0040000 0x10000>;
-	clocks = <&clks 27>;
-};

Documentation/devicetree/bindings/sound/sirf-usp.txt

@@ -1,27 +0,0 @@
-* SiRF SoC USP module
-
-Required properties:
-- compatible: "sirf,prima2-usp-pcm"
-- reg: Base address and size entries:
-- dmas: List of DMA controller phandle and DMA request line ordered pairs.
-- dma-names: Identifier string for each DMA request line in the dmas property.
-  These strings correspond 1:1 with the ordered pairs in dmas.
-
-  One of the DMA channels will be responsible for transmission (should be
-  named "tx") and one for reception (should be named "rx").
-
-- clocks: USP controller clock source
-- pinctrl-names: Must contain a "default" entry.
-- pinctrl-NNN: One property must exist for each entry in pinctrl-names.
-
-Example:
-usp0: usp@b0080000 {
-	compatible = "sirf,prima2-usp-pcm";
-	reg = <0xb0080000 0x10000>;
-	clocks = <&clks 28>;
-	dmas = <&dmac1 1>, <&dmac1 2>;
-	dma-names = "rx", "tx";
-	pinctrl-names = "default";
-	pinctrl-0 = <&usp0_only_utfs_pins_a>;
-};
-

Documentation/devicetree/bindings/sound/st,stm32-i2s.yaml

@@ -54,6 +54,10 @@ properties:
   resets:
     maxItems: 1
 
+  "#clock-cells":
+    description: Configure the I2S device as MCLK clock provider.
+    const: 0
+
 required:
   - compatible
   - "#sound-dai-cells"

Documentation/devicetree/bindings/sound/wm8962.txt

@@ -9,6 +9,9 @@ Required properties:
   - reg : the I2C address of the device.
 
 Optional properties:
+
+  - clocks : The clock source of the mclk
+
   - spk-mono: This is a boolean property. If present, the SPK_MONO bit
     of R51 (Class D Control 2) gets set, indicating that the speaker is
     in mono mode.
@@ -27,6 +30,7 @@ Example:
 wm8962: codec@1a {
 	compatible = "wlf,wm8962";
 	reg = <0x1a>;
+	clocks = <&clks IMX6QDL_CLK_CKO>;
 
 	gpio-cfg = <
 		0x0000 /* 0:Default */

Documentation/devicetree/bindings/sound/zte,tdm.txt

@@ -1,30 +0,0 @@
-ZTE TDM DAI driver
-
-Required properties:
-
-- compatible : should be one of the following.
-       * zte,zx296718-tdm
-- reg : physical base address of the controller and length of memory mapped
-    region.
-- clocks : Pairs of phandle and specifier referencing the controller's clocks.
-- clock-names: "wclk" for the wclk.
-               "pclk" for the pclk.
--#clock-cells: should be 1.
-- zte,tdm-dma-sysctrl : Reference to the sysctrl controller controlling
-    the dma. includes:
-	phandle of sysctrl.
-	register offset in sysctrl for control dma.
-	mask of the register that be written to sysctrl.
-
-Example:
-
-	tdm: tdm@1487000 {
-		compatible = "zte,zx296718-tdm";
-		reg = <0x01487000 0x1000>;
-		clocks = <&audiocrm AUDIO_TDM_WCLK>, <&audiocrm AUDIO_TDM_PCLK>;
-		clock-names = "wclk", "pclk";
-		#clock-cells = <1>;
-		pinctrl-names = "default";
-		pinctrl-0 = <&tdm_global_pin>;
-		zte,tdm-dma-sysctrl = <&sysctrl 0x10c 4>;
-	};

Documentation/devicetree/bindings/sound/zte,zx-aud96p22.txt

@@ -1,24 +0,0 @@
-ZTE ZX AUD96P22 Audio Codec
-
-Required properties:
- - compatible: Must be "zte,zx-aud96p22"
- - #sound-dai-cells: Should be 0
- - reg: I2C bus slave address of AUD96P22
-
-Example:
-
-	i2c0: i2c@1486000 {
-		compatible = "zte,zx296718-i2c";
-		reg = <0x01486000 0x1000>;
-		interrupts = <GIC_SPI 35 IRQ_TYPE_LEVEL_HIGH>;
-		#address-cells = <1>;
-		#size-cells = <0>;
-		clocks = <&audiocrm AUDIO_I2C0_WCLK>;
-		clock-frequency = <1600000>;
-
-		aud96p22: codec@22 {
-			compatible = "zte,zx-aud96p22";
-			#sound-dai-cells = <0>;
-			reg = <0x22>;
-		};
-	};

Documentation/devicetree/bindings/sound/zte,zx-i2s.txt

@@ -1,45 +0,0 @@
-ZTE ZX296702 I2S controller
-
-Required properties:
- - compatible : Must be one of:
-	"zte,zx296718-i2s", "zte,zx296702-i2s"
-	"zte,zx296702-i2s"
- - reg : Must contain I2S core's registers location and length
- - clocks : Pairs of phandle and specifier referencing the controller's clocks.
- - clock-names: "wclk" for the wclk, "pclk" for the pclk to the I2S interface.
- - dmas: Pairs of phandle and specifier for the DMA channel that is used by
-   the core. The core expects two dma channels for transmit.
- - dma-names : Must be "tx" and "rx"
-
-For more details on the 'dma', 'dma-names', 'clock' and 'clock-names' properties
-please check:
-	* resource-names.txt
-	* clock/clock-bindings.txt
-	* dma/dma.txt
-
-Example:
-	i2s0: i2s@b005000 {
-		#sound-dai-cells = <0>;
-		compatible = "zte,zx296718-i2s", "zte,zx296702-i2s";
-		reg = <0x0b005000 0x1000>;
-		clocks = <&audiocrm AUDIO_I2S0_WCLK>, <&audiocrm AUDIO_I2S0_PCLK>;
-		clock-names = "wclk", "pclk";
-		interrupts = <GIC_SPI 22 IRQ_TYPE_LEVEL_HIGH>;
-		dmas = <&dma 5>, <&dma 6>;
-		dma-names = "tx", "rx";
-	};
-
-	sound {
-		compatible = "simple-audio-card";
-		simple-audio-card,name = "zx296702_snd";
-		simple-audio-card,format = "left_j";
-		simple-audio-card,bitclock-master = <&sndcodec>;
-		simple-audio-card,frame-master = <&sndcodec>;
-		sndcpu: simple-audio-card,cpu {
-			sound-dai = <&i2s0>;
-		};
-
-		sndcodec: simple-audio-card,codec {
-			sound-dai = <&acodec>;
-		};
-	};

Documentation/devicetree/bindings/sound/zte,zx-spdif.txt

@@ -1,27 +0,0 @@
-ZTE ZX296702 SPDIF controller
-
-Required properties:
- - compatible : Must be "zte,zx296702-spdif"
- - reg : Must contain SPDIF core's registers location and length
- - clocks : Pairs of phandle and specifier referencing the controller's clocks.
- - clock-names: "tx" for the clock to the SPDIF interface.
- - dmas: Pairs of phandle and specifier for the DMA channel that is used by
-   the core. The core expects one dma channel for transmit.
- - dma-names : Must be "tx"
-
-For more details on the 'dma', 'dma-names', 'clock' and 'clock-names' properties
-please check:
-	* resource-names.txt
-	* clock/clock-bindings.txt
-	* dma/dma.txt
-
-Example:
-	spdif0: spdif0@b004000 {
-		compatible = "zte,zx296702-spdif";
-		reg = <0x0b004000 0x1000>;
-		clocks = <&lsp0clk ZX296702_SPDIF0_DIV>;
-		clock-names = "tx";
-		interrupts = <GIC_SPI 21 IRQ_TYPE_LEVEL_HIGH>;
-		dmas = <&dma 4>;
-		dma-names = "tx";
-	};

drivers/mfd/arizona-core.c

@@ -797,17 +797,6 @@ const struct dev_pm_ops arizona_pm_ops = {
 EXPORT_SYMBOL_GPL(arizona_pm_ops);
 
 #ifdef CONFIG_OF
-unsigned long arizona_of_get_type(struct device *dev)
-{
-	const struct of_device_id *id = of_match_device(arizona_of_match, dev);
-
-	if (id)
-		return (unsigned long)id->data;
-	else
-		return 0;
-}
-EXPORT_SYMBOL_GPL(arizona_of_get_type);
-
 static int arizona_of_get_core_pdata(struct arizona *arizona)
 {
 	struct arizona_pdata *pdata = &arizona->pdata;

drivers/mfd/arizona-i2c.c

@@ -23,14 +23,16 @@
 static int arizona_i2c_probe(struct i2c_client *i2c,
 			     const struct i2c_device_id *id)
 {
+	const void *match_data;
 	struct arizona *arizona;
 	const struct regmap_config *regmap_config = NULL;
-	unsigned long type;
+	unsigned long type = 0;
 	int ret;
 
-	if (i2c->dev.of_node)
-		type = arizona_of_get_type(&i2c->dev);
-	else
+	match_data = device_get_match_data(&i2c->dev);
+	if (match_data)
+		type = (unsigned long)match_data;
+	else if (id)
 		type = id->driver_data;
 
 	switch (type) {
@@ -115,6 +117,7 @@ static struct i2c_driver arizona_i2c_driver = {
 
 module_i2c_driver(arizona_i2c_driver);
 
+MODULE_SOFTDEP("pre: arizona_ldo1");
 MODULE_DESCRIPTION("Arizona I2C bus interface");
 MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
 MODULE_LICENSE("GPL");

drivers/mfd/arizona-spi.c

@@ -7,7 +7,10 @@
  * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
  */
 
+#include <linux/acpi.h>
 #include <linux/err.h>
+#include <linux/gpio/consumer.h>
+#include <linux/gpio/machine.h>
 #include <linux/module.h>
 #include <linux/pm_runtime.h>
 #include <linux/regmap.h>
@@ -15,22 +18,141 @@
 #include <linux/slab.h>
 #include <linux/spi/spi.h>
 #include <linux/of.h>
+#include <uapi/linux/input-event-codes.h>
 
 #include <linux/mfd/arizona/core.h>
 
 #include "arizona.h"
 
+#ifdef CONFIG_ACPI
+const struct acpi_gpio_params reset_gpios = { 1, 0, false };
+const struct acpi_gpio_params ldoena_gpios = { 2, 0, false };
+
+static const struct acpi_gpio_mapping arizona_acpi_gpios[] = {
+	{ "reset-gpios", &reset_gpios, 1, },
+	{ "wlf,ldoena-gpios", &ldoena_gpios, 1 },
+	{ }
+};
+
+/*
+ * The ACPI resources for the device only describe external GPIO-s. They do
+ * not provide mappings for the GPIO-s coming from the Arizona codec itself.
+ */
+static const struct gpiod_lookup arizona_soc_gpios[] = {
+	{ "arizona", 2, "wlf,spkvdd-ena", 0, GPIO_ACTIVE_HIGH },
+	{ "arizona", 4, "wlf,micd-pol", 0, GPIO_ACTIVE_LOW },
+};
+
+/*
+ * The AOSP 3.5 mm Headset: Accessory Specification gives the following values:
+ * Function A Play/Pause:           0 ohm
+ * Function D Voice assistant:    135 ohm
+ * Function B Volume Up           240 ohm
+ * Function C Volume Down         470 ohm
+ * Minimum Mic DC resistance     1000 ohm
+ * Minimum Ear speaker impedance   16 ohm
+ * Note the first max value below must be less then the min. speaker impedance,
+ * to allow CTIA/OMTP detection to work. The other max values are the closest
+ * value from extcon-arizona.c:arizona_micd_levels halfway 2 button resistances.
+ */
+static const struct arizona_micd_range arizona_micd_aosp_ranges[] = {
+	{ .max =  11, .key = KEY_PLAYPAUSE },
+	{ .max = 186, .key = KEY_VOICECOMMAND },
+	{ .max = 348, .key = KEY_VOLUMEUP },
+	{ .max = 752, .key = KEY_VOLUMEDOWN },
+};
+
+static void arizona_spi_acpi_remove_lookup(void *lookup)
+{
+	gpiod_remove_lookup_table(lookup);
+}
+
+static int arizona_spi_acpi_probe(struct arizona *arizona)
+{
+	struct gpiod_lookup_table *lookup;
+	acpi_status status;
+	int ret;
+
+	/* Add mappings for the 2 ACPI declared GPIOs used for reset and ldo-ena */
+	devm_acpi_dev_add_driver_gpios(arizona->dev, arizona_acpi_gpios);
+
+	/* Add lookups for the SoCs own GPIOs used for micdet-polarity and spkVDD-enable */
+	lookup = devm_kzalloc(arizona->dev,
+			      struct_size(lookup, table, ARRAY_SIZE(arizona_soc_gpios) + 1),
+			      GFP_KERNEL);
+	if (!lookup)
+		return -ENOMEM;
+
+	lookup->dev_id = dev_name(arizona->dev);
+	memcpy(lookup->table, arizona_soc_gpios, sizeof(arizona_soc_gpios));
+
+	gpiod_add_lookup_table(lookup);
+	ret = devm_add_action_or_reset(arizona->dev, arizona_spi_acpi_remove_lookup, lookup);
+	if (ret)
+		return ret;
+
+	/* Enable 32KHz clock from SoC to codec for jack-detect */
+	status = acpi_evaluate_object(ACPI_HANDLE(arizona->dev), "CLKE", NULL, NULL);
+	if (ACPI_FAILURE(status))
+		dev_warn(arizona->dev, "Failed to enable 32KHz clk ACPI error %d\n", status);
+
+	/*
+	 * Some DSDTs wrongly declare the IRQ trigger-type as IRQF_TRIGGER_FALLING
+	 * The IRQ line will stay low when a new IRQ event happens between reading
+	 * the IRQ status flags and acknowledging them. When the IRQ line stays
+	 * low like this the IRQ will never trigger again when its type is set
+	 * to IRQF_TRIGGER_FALLING. Correct the IRQ trigger-type to fix this.
+	 *
+	 * Note theoretically it is possible that some boards are not capable
+	 * of handling active low level interrupts. In that case setting the
+	 * flag to IRQF_TRIGGER_FALLING would not be a bug (and we would need
+	 * to work around this) but so far all known usages of IRQF_TRIGGER_FALLING
+	 * are a bug in the board's DSDT.
+	 */
+	arizona->pdata.irq_flags = IRQF_TRIGGER_LOW;
+
+	/* Wait 200 ms after jack insertion */
+	arizona->pdata.micd_detect_debounce = 200;
+
+	/* Use standard AOSP values for headset-button mappings */
+	arizona->pdata.micd_ranges = arizona_micd_aosp_ranges;
+	arizona->pdata.num_micd_ranges = ARRAY_SIZE(arizona_micd_aosp_ranges);
+
+	return 0;
+}
+
+static const struct acpi_device_id arizona_acpi_match[] = {
+	{
+		.id = "WM510204",
+		.driver_data = WM5102,
+	},
+	{
+		.id = "WM510205",
+		.driver_data = WM5102,
+	},
+	{ }
+};
+MODULE_DEVICE_TABLE(acpi, arizona_acpi_match);
+#else
+static int arizona_spi_acpi_probe(struct arizona *arizona)
+{
+	return -ENODEV;
+}
+#endif
+
 static int arizona_spi_probe(struct spi_device *spi)
 {
 	const struct spi_device_id *id = spi_get_device_id(spi);
+	const void *match_data;
 	struct arizona *arizona;
 	const struct regmap_config *regmap_config = NULL;
-	unsigned long type;
+	unsigned long type = 0;
 	int ret;
 
-	if (spi->dev.of_node)
-		type = arizona_of_get_type(&spi->dev);
-	else
+	match_data = device_get_match_data(&spi->dev);
+	if (match_data)
+		type = (unsigned long)match_data;
+	else if (id)
 		type = id->driver_data;
 
 	switch (type) {
@@ -75,6 +197,12 @@ static int arizona_spi_probe(struct spi_device *spi)
 	arizona->dev = &spi->dev;
 	arizona->irq = spi->irq;
 
+	if (has_acpi_companion(&spi->dev)) {
+		ret = arizona_spi_acpi_probe(arizona);
+		if (ret)
+			return ret;
+	}
+
 	return arizona_dev_init(arizona);
 }
 
@@ -102,6 +230,7 @@ static struct spi_driver arizona_spi_driver = {
 		.name	= "arizona",
 		.pm	= &arizona_pm_ops,
 		.of_match_table	= of_match_ptr(arizona_of_match),
+		.acpi_match_table = ACPI_PTR(arizona_acpi_match),
 	},
 	.probe		= arizona_spi_probe,
 	.remove		= arizona_spi_remove,
@@ -110,6 +239,7 @@ static struct spi_driver arizona_spi_driver = {
 
 module_spi_driver(arizona_spi_driver);
 
+MODULE_SOFTDEP("pre: arizona_ldo1");
 MODULE_DESCRIPTION("Arizona SPI bus interface");
 MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
 MODULE_LICENSE("GPL");

drivers/mfd/arizona.h

@@ -50,13 +50,4 @@ int arizona_dev_exit(struct arizona *arizona);
 int arizona_irq_init(struct arizona *arizona);
 int arizona_irq_exit(struct arizona *arizona);
 
-#ifdef CONFIG_OF
-unsigned long arizona_of_get_type(struct device *dev);
-#else
-static inline unsigned long arizona_of_get_type(struct device *dev)
-{
-	return 0;
-}
-#endif
-
 #endif

include/linux/platform_data/cros_ec_commands.h

@@ -4600,6 +4600,7 @@ enum ec_codec_i2s_rx_subcmd {
 	EC_CODEC_I2S_RX_SET_SAMPLE_DEPTH = 0x2,
 	EC_CODEC_I2S_RX_SET_DAIFMT = 0x3,
 	EC_CODEC_I2S_RX_SET_BCLK = 0x4,
+	EC_CODEC_I2S_RX_RESET = 0x5,
 	EC_CODEC_I2S_RX_SUBCMD_COUNT,
 };
 

include/sound/dmaengine_pcm.h

@@ -66,6 +66,9 @@ struct dma_chan *snd_dmaengine_pcm_get_chan(struct snd_pcm_substream *substream)
  * @chan_name: Custom channel name to use when requesting DMA channel.
  * @fifo_size: FIFO size of the DAI controller in bytes
  * @flags: PCM_DAI flags, only SND_DMAENGINE_PCM_DAI_FLAG_PACK for now
+ * @peripheral_config: peripheral configuration for programming peripheral
+ * for dmaengine transfer
+ * @peripheral_size: peripheral configuration buffer size
  */
 struct snd_dmaengine_dai_dma_data {
 	dma_addr_t addr;
@@ -76,6 +79,8 @@ struct snd_dmaengine_dai_dma_data {
 	const char *chan_name;
 	unsigned int fifo_size;
 	unsigned int flags;
+	void *peripheral_config;
+	size_t peripheral_size;
 };
 
 void snd_dmaengine_pcm_set_config_from_dai_data(

include/sound/graph_card.h

@@ -9,8 +9,10 @@
 
 #include <sound/simple_card_utils.h>
 
-int graph_card_probe(struct snd_soc_card *card);
+int audio_graph_card_probe(struct snd_soc_card *card);
 
-int graph_parse_of(struct asoc_simple_priv *priv, struct device *dev);
+int audio_graph_parse_of(struct asoc_simple_priv *priv, struct device *dev);
+
+int audio_graph_remove(struct platform_device *pdev);
 
 #endif /* __GRAPH_CARD_H */

include/sound/hdmi-codec.h

@@ -34,6 +34,11 @@ struct hdmi_codec_daifmt {
 	unsigned int frame_clk_inv:1;
 	unsigned int bit_clk_master:1;
 	unsigned int frame_clk_master:1;
+	/* bit_fmt could be standard PCM format or
+	 * IEC958 encoded format. ALSA IEC958 plugin will pass
+	 * IEC958_SUBFRAME format to the underneath driver.
+	 */
+	snd_pcm_format_t bit_fmt;
 };
 
 /*

include/sound/rt5645.h

@@ -22,6 +22,8 @@ struct rt5645_platform_data {
 	bool level_trigger_irq;
 	/* Invert JD1_1 status polarity */
 	bool inv_jd1_1;
+	/* Invert HP detect status polarity */
+	bool inv_hp_pol;
 
 	/* Value to asign to snd_soc_card.long_name */
 	const char *long_name;

include/sound/soc-component.h

@@ -353,6 +353,12 @@ int snd_soc_component_test_bits(struct snd_soc_component *component,
 				unsigned int reg, unsigned int mask,
 				unsigned int value);
 
+unsigned int snd_soc_component_read_field(struct snd_soc_component *component,
+					  unsigned int reg, unsigned int mask);
+int snd_soc_component_write_field(struct snd_soc_component *component,
+				  unsigned int reg, unsigned int mask,
+				  unsigned int val);
+
 /* component wide operations */
 int snd_soc_component_set_sysclk(struct snd_soc_component *component,
 				 int clk_id, int source,

include/sound/soc-dai.h

@@ -353,9 +353,9 @@ struct snd_soc_dai_driver {
 	/* DAI capabilities */
 	struct snd_soc_pcm_stream capture;
 	struct snd_soc_pcm_stream playback;
-	unsigned int symmetric_rates:1;
+	unsigned int symmetric_rate:1;
 	unsigned int symmetric_channels:1;
-	unsigned int symmetric_samplebits:1;
+	unsigned int symmetric_sample_bits:1;
 
 	/* probe ordering - for components with runtime dependencies */
 	int probe_order;

include/sound/soc.h

@@ -685,9 +685,9 @@ struct snd_soc_dai_link {
 	unsigned int ignore_suspend:1;
 
 	/* Symmetry requirements */
-	unsigned int symmetric_rates:1;
+	unsigned int symmetric_rate:1;
 	unsigned int symmetric_channels:1;
-	unsigned int symmetric_samplebits:1;
+	unsigned int symmetric_sample_bits:1;
 
 	/* Do not create a PCM for this DAI link (Backend link) */
 	unsigned int no_pcm:1;

include/sound/sof/ext_manifest.h

@@ -58,9 +58,9 @@ struct sof_ext_man_header {
 /* Extended manifest elements types */
 enum sof_ext_man_elem_type {
 	SOF_EXT_MAN_ELEM_FW_VERSION		= 0,
-	SOF_EXT_MAN_ELEM_WINDOW			= SOF_IPC_EXT_WINDOW,
-	SOF_EXT_MAN_ELEM_CC_VERSION		= SOF_IPC_EXT_CC_INFO,
-	SOF_EXT_MAN_ELEM_DBG_ABI		= SOF_IPC_EXT_USER_ABI_INFO,
+	SOF_EXT_MAN_ELEM_WINDOW			= 1,
+	SOF_EXT_MAN_ELEM_CC_VERSION		= 2,
+	SOF_EXT_MAN_ELEM_DBG_ABI		= 4,
 	SOF_EXT_MAN_ELEM_CONFIG_DATA		= 5, /**< ABI3.17 */
 	SOF_EXT_MAN_ELEM_PLATFORM_CONFIG_DATA   = 6,
 };

sound/core/pcm_dmaengine.c

@@ -125,6 +125,8 @@ void snd_dmaengine_pcm_set_config_from_dai_data(
 	}
 
 	slave_config->slave_id = dma_data->slave_id;
+	slave_config->peripheral_config = dma_data->peripheral_config;
+	slave_config->peripheral_size = dma_data->peripheral_size;
 }
 EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_set_config_from_dai_data);
 

sound/soc/Kconfig

@@ -37,6 +37,23 @@ config SND_SOC_COMPRESS
 config SND_SOC_TOPOLOGY
 	bool
 
+config SND_SOC_TOPOLOGY_KUNIT_TESTS
+	tristate "KUnit tests for SoC topology"
+	depends on KUNIT
+	depends on SND_SOC_TOPOLOGY
+	default KUNIT_ALL_TESTS
+	help
+	  If you want to perform tests on ALSA SoC topology support say Y here.
+
+	  This builds a module which can be later manually loaded to run KUNIT
+	  test cases against soc-topology.c API. This should be primarily used
+	  by developers to test their changes to ASoC.
+
+	  Do note that it creates fake playback devices which do not interact
+	  well with userspace. When running tests one may want to disable
+	  userspace applications such as pulseaudio, to prevent unnecessary
+	  problems.
+
 config SND_SOC_ACPI
 	tristate
 
@@ -62,7 +79,6 @@ source "sound/soc/qcom/Kconfig"
 source "sound/soc/rockchip/Kconfig"
 source "sound/soc/samsung/Kconfig"
 source "sound/soc/sh/Kconfig"
-source "sound/soc/sirf/Kconfig"
 source "sound/soc/sof/Kconfig"
 source "sound/soc/spear/Kconfig"
 source "sound/soc/sprd/Kconfig"
@@ -71,12 +87,10 @@ source "sound/soc/stm/Kconfig"
 source "sound/soc/sunxi/Kconfig"
 source "sound/soc/tegra/Kconfig"
 source "sound/soc/ti/Kconfig"
-source "sound/soc/txx9/Kconfig"
 source "sound/soc/uniphier/Kconfig"
 source "sound/soc/ux500/Kconfig"
 source "sound/soc/xilinx/Kconfig"
 source "sound/soc/xtensa/Kconfig"
-source "sound/soc/zte/Kconfig"
 
 # Supported codecs
 source "sound/soc/codecs/Kconfig"

sound/soc/Makefile

@@ -7,6 +7,11 @@ ifneq ($(CONFIG_SND_SOC_TOPOLOGY),)
 snd-soc-core-objs += soc-topology.o
 endif
 
+ifneq ($(CONFIG_SND_SOC_TOPOLOGY_KUNIT_TESTS),)
+# snd-soc-test-objs := soc-topology-test.o
+obj-$(CONFIG_SND_SOC_TOPOLOGY_KUNIT_TESTS) := soc-topology-test.o
+endif
+
 ifneq ($(CONFIG_SND_SOC_GENERIC_DMAENGINE_PCM),)
 snd-soc-core-objs += soc-generic-dmaengine-pcm.o
 endif
@@ -45,7 +50,6 @@ obj-$(CONFIG_SND_SOC)	+= qcom/
 obj-$(CONFIG_SND_SOC)	+= rockchip/
 obj-$(CONFIG_SND_SOC)	+= samsung/
 obj-$(CONFIG_SND_SOC)	+= sh/
-obj-$(CONFIG_SND_SOC)	+= sirf/
 obj-$(CONFIG_SND_SOC)	+= sof/
 obj-$(CONFIG_SND_SOC)	+= spear/
 obj-$(CONFIG_SND_SOC)	+= sprd/
@@ -54,9 +58,7 @@ obj-$(CONFIG_SND_SOC)	+= stm/
 obj-$(CONFIG_SND_SOC)	+= sunxi/
 obj-$(CONFIG_SND_SOC)	+= tegra/
 obj-$(CONFIG_SND_SOC)	+= ti/
-obj-$(CONFIG_SND_SOC)	+= txx9/
 obj-$(CONFIG_SND_SOC)	+= uniphier/
 obj-$(CONFIG_SND_SOC)	+= ux500/
 obj-$(CONFIG_SND_SOC)	+= xilinx/
 obj-$(CONFIG_SND_SOC)	+= xtensa/
-obj-$(CONFIG_SND_SOC)	+= zte/

sound/soc/adi/axi-i2s.c

@@ -156,7 +156,7 @@ static const struct snd_soc_dai_ops axi_i2s_dai_ops = {
 static struct snd_soc_dai_driver axi_i2s_dai = {
 	.probe = axi_i2s_dai_probe,
 	.ops = &axi_i2s_dai_ops,
-	.symmetric_rates = 1,
+	.symmetric_rate = 1,
 };
 
 static const struct snd_soc_component_driver axi_i2s_component = {

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

@@ -140,9 +140,7 @@ static int acp3x_1015_hw_params(struct snd_pcm_substream *substream,
 	for_each_rtd_codec_dais(rtd, i, codec_dai) {
 		if (strcmp(codec_dai->name, "rt1015-aif"))
 			continue;
-		ret = snd_soc_dai_set_bclk_ratio(codec_dai, 64);
-		if (ret < 0)
-			return ret;
+
 		ret = snd_soc_dai_set_pll(codec_dai, 0, RT1015_PLL_S_BCLK,
 						64 * srate, 256 * srate);
 		if (ret < 0)

sound/soc/atmel/atmel-i2s.c

@@ -541,7 +541,7 @@ static struct snd_soc_dai_driver atmel_i2s_dai = {
 		.formats = ATMEL_I2S_FORMATS,
 	},
 	.ops = &atmel_i2s_dai_ops,
-	.symmetric_rates = 1,
+	.symmetric_rate = 1,
 };
 
 static const struct snd_soc_component_driver atmel_i2s_component = {

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

@@ -34,86 +34,21 @@
 #include "atmel-pcm.h"
 
 
-static int atmel_pcm_preallocate_dma_buffer(struct snd_pcm *pcm,
-	int stream)
-{
-	struct snd_pcm_substream *substream = pcm->streams[stream].substream;
-	struct snd_dma_buffer *buf = &substream->dma_buffer;
-	size_t size = ATMEL_SSC_DMABUF_SIZE;
-
-	buf->dev.type = SNDRV_DMA_TYPE_DEV;
-	buf->dev.dev = pcm->card->dev;
-	buf->private_data = NULL;
-	buf->area = dma_alloc_coherent(pcm->card->dev, size,
-			&buf->addr, GFP_KERNEL);
-	pr_debug("atmel-pcm: alloc dma buffer: area=%p, addr=%p, size=%zu\n",
-			(void *)buf->area, (void *)(long)buf->addr, size);
-
-	if (!buf->area)
-		return -ENOMEM;
-
-	buf->bytes = size;
-	return 0;
-}
-
-static int atmel_pcm_mmap(struct snd_soc_component *component,
-			  struct snd_pcm_substream *substream,
-			  struct vm_area_struct *vma)
-{
-	return remap_pfn_range(vma, vma->vm_start,
-		       substream->dma_buffer.addr >> PAGE_SHIFT,
-		       vma->vm_end - vma->vm_start, vma->vm_page_prot);
-}
-
 static int atmel_pcm_new(struct snd_soc_component *component,
 			 struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_card *card = rtd->card->snd_card;
-	struct snd_pcm *pcm = rtd->pcm;
 	int ret;
 
 	ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32));
 	if (ret)
 		return ret;
 
-	if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
-		pr_debug("atmel-pcm: allocating PCM playback DMA buffer\n");
-		ret = atmel_pcm_preallocate_dma_buffer(pcm,
-			SNDRV_PCM_STREAM_PLAYBACK);
-		if (ret)
-			goto out;
-	}
+	snd_pcm_set_managed_buffer_all(rtd->pcm, SNDRV_DMA_TYPE_DEV,
+				       card->dev, ATMEL_SSC_DMABUF_SIZE,
+				       ATMEL_SSC_DMABUF_SIZE);
 
-	if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
-		pr_debug("atmel-pcm: allocating PCM capture DMA buffer\n");
-		ret = atmel_pcm_preallocate_dma_buffer(pcm,
-			SNDRV_PCM_STREAM_CAPTURE);
-		if (ret)
-			goto out;
-	}
- out:
-	return ret;
-}
-
-static void atmel_pcm_free(struct snd_soc_component *component,
-			   struct snd_pcm *pcm)
-{
-	struct snd_pcm_substream *substream;
-	struct snd_dma_buffer *buf;
-	int stream;
-
-	for (stream = 0; stream < 2; stream++) {
-		substream = pcm->streams[stream].substream;
-		if (!substream)
-			continue;
-
-		buf = &substream->dma_buffer;
-		if (!buf->area)
-			continue;
-		dma_free_coherent(pcm->card->dev, buf->bytes,
-				  buf->area, buf->addr);
-		buf->area = NULL;
-	}
+	return 0;
 }
 
 /*--------------------------------------------------------------------------*\
@@ -210,9 +145,6 @@ static int atmel_pcm_hw_params(struct snd_soc_component *component,
 	/* this may get called several times by oss emulation
 	 * with different params */
 
-	snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
-	runtime->dma_bytes = params_buffer_bytes(params);
-
 	prtd->params = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream);
 	prtd->params->dma_intr_handler = atmel_pcm_dma_irq;
 
@@ -384,9 +316,7 @@ static const struct snd_soc_component_driver atmel_soc_platform = {
 	.prepare	= atmel_pcm_prepare,
 	.trigger	= atmel_pcm_trigger,
 	.pointer	= atmel_pcm_pointer,
-	.mmap		= atmel_pcm_mmap,
 	.pcm_construct	= atmel_pcm_new,
-	.pcm_destruct	= atmel_pcm_free,
 };
 
 int atmel_pcm_pdc_platform_register(struct device *dev)

sound/soc/atmel/mchp-i2s-mcc.c

@@ -859,8 +859,8 @@ static struct snd_soc_dai_driver mchp_i2s_mcc_dai = {
 		.formats = MCHP_I2SMCC_FORMATS,
 	},
 	.ops = &mchp_i2s_mcc_dai_ops,
-	.symmetric_rates = 1,
-	.symmetric_samplebits = 1,
+	.symmetric_rate = 1,
+	.symmetric_sample_bits = 1,
 	.symmetric_channels = 1,
 };
 

sound/soc/au1x/i2sc.c

@@ -210,7 +210,7 @@ static const struct snd_soc_dai_ops au1xi2s_dai_ops = {
 };
 
 static struct snd_soc_dai_driver au1xi2s_dai_driver = {
-	.symmetric_rates	= 1,
+	.symmetric_rate		= 1,
 	.playback = {
 		.rates		= AU1XI2SC_RATES,
 		.formats	= AU1XI2SC_FMTS,

sound/soc/bcm/bcm2835-i2s.c

@@ -783,8 +783,8 @@ static struct snd_soc_dai_driver bcm2835_i2s_dai = {
 				| SNDRV_PCM_FMTBIT_S32_LE
 		},
 	.ops = &bcm2835_i2s_dai_ops,
-	.symmetric_rates = 1,
-	.symmetric_samplebits = 1,
+	.symmetric_rate = 1,
+	.symmetric_sample_bits = 1,
 };
 
 static bool bcm2835_i2s_volatile_reg(struct device *dev, unsigned int reg)

sound/soc/bcm/bcm63xx-i2s-whistler.c

@@ -212,7 +212,7 @@ static struct snd_soc_dai_driver bcm63xx_i2s_dai = {
 		.formats = SNDRV_PCM_FMTBIT_S32_LE,
 	},
 	.ops = &bcm63xx_i2s_dai_ops,
-	.symmetric_rates = 1,
+	.symmetric_rate = 1,
 	.symmetric_channels = 1,
 };
 

sound/soc/bcm/cygnus-pcm.c

@@ -636,36 +636,6 @@ static int cygnus_pcm_close(struct snd_soc_component *component,
 	return 0;
 }
 
-static int cygnus_pcm_hw_params(struct snd_soc_component *component,
-				struct snd_pcm_substream *substream,
-				struct snd_pcm_hw_params *params)
-{
-	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct cygnus_aio_port *aio;
-
-	aio = cygnus_dai_get_dma_data(substream);
-	dev_dbg(asoc_rtd_to_cpu(rtd, 0)->dev, "%s  port %d\n", __func__, aio->portnum);
-
-	snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
-	runtime->dma_bytes = params_buffer_bytes(params);
-
-	return 0;
-}
-
-static int cygnus_pcm_hw_free(struct snd_soc_component *component,
-			      struct snd_pcm_substream *substream)
-{
-	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
-	struct cygnus_aio_port *aio;
-
-	aio = cygnus_dai_get_dma_data(substream);
-	dev_dbg(asoc_rtd_to_cpu(rtd, 0)->dev, "%s  port %d\n", __func__, aio->portnum);
-
-	snd_pcm_set_runtime_buffer(substream, NULL);
-	return 0;
-}
-
 static int cygnus_pcm_prepare(struct snd_soc_component *component,
 			      struct snd_pcm_substream *substream)
 {
@@ -730,87 +700,19 @@ static snd_pcm_uframes_t cygnus_pcm_pointer(struct snd_soc_component *component,
 	return bytes_to_frames(substream->runtime, res);
 }
 
-static int cygnus_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
-{
-	struct snd_pcm_substream *substream = pcm->streams[stream].substream;
-	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
-	struct snd_dma_buffer *buf = &substream->dma_buffer;
-	size_t size;
-
-	size = cygnus_pcm_hw.buffer_bytes_max;
-
-	buf->dev.type = SNDRV_DMA_TYPE_DEV;
-	buf->dev.dev = pcm->card->dev;
-	buf->private_data = NULL;
-	buf->area = dma_alloc_coherent(pcm->card->dev, size,
-			&buf->addr, GFP_KERNEL);
-
-	dev_dbg(asoc_rtd_to_cpu(rtd, 0)->dev, "%s: size 0x%zx @ %pK\n",
-				__func__, size, buf->area);
-
-	if (!buf->area) {
-		dev_err(asoc_rtd_to_cpu(rtd, 0)->dev, "%s: dma_alloc failed\n", __func__);
-		return -ENOMEM;
-	}
-	buf->bytes = size;
-
-	return 0;
-}
-
-static void cygnus_dma_free_dma_buffers(struct snd_soc_component *component,
-					struct snd_pcm *pcm)
-{
-	struct snd_pcm_substream *substream;
-	struct snd_dma_buffer *buf;
-
-	substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
-	if (substream) {
-		buf = &substream->dma_buffer;
-		if (buf->area) {
-			dma_free_coherent(pcm->card->dev, buf->bytes,
-				buf->area, buf->addr);
-			buf->area = NULL;
-		}
-	}
-
-	substream = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream;
-	if (substream) {
-		buf = &substream->dma_buffer;
-		if (buf->area) {
-			dma_free_coherent(pcm->card->dev, buf->bytes,
-				buf->area, buf->addr);
-			buf->area = NULL;
-		}
-	}
-}
-
 static int cygnus_dma_new(struct snd_soc_component *component,
 			  struct snd_soc_pcm_runtime *rtd)
 {
+	size_t size = cygnus_pcm_hw.buffer_bytes_max;
 	struct snd_card *card = rtd->card->snd_card;
-	struct snd_pcm *pcm = rtd->pcm;
-	int ret;
 
 	if (!card->dev->dma_mask)
 		card->dev->dma_mask = &cygnus_dma_dmamask;
 	if (!card->dev->coherent_dma_mask)
 		card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
 
-	if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
-		ret = cygnus_pcm_preallocate_dma_buffer(pcm,
-				SNDRV_PCM_STREAM_PLAYBACK);
-		if (ret)
-			return ret;
-	}
-
-	if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
-		ret = cygnus_pcm_preallocate_dma_buffer(pcm,
-				SNDRV_PCM_STREAM_CAPTURE);
-		if (ret) {
-			cygnus_dma_free_dma_buffers(component, pcm);
-			return ret;
-		}
-	}
+	snd_pcm_set_managed_buffer_all(rtd->pcm, SNDRV_DMA_TYPE_DEV,
+				       card->dev, size, size);
 
 	return 0;
 }
@@ -818,13 +720,10 @@ static int cygnus_dma_new(struct snd_soc_component *component,
 static struct snd_soc_component_driver cygnus_soc_platform = {
 	.open		= cygnus_pcm_open,
 	.close		= cygnus_pcm_close,
-	.hw_params	= cygnus_pcm_hw_params,
-	.hw_free	= cygnus_pcm_hw_free,
 	.prepare	= cygnus_pcm_prepare,
 	.trigger	= cygnus_pcm_trigger,
 	.pointer	= cygnus_pcm_pointer,
 	.pcm_construct	= cygnus_dma_new,
-	.pcm_destruct	= cygnus_dma_free_dma_buffers,
 };
 
 int cygnus_soc_platform_register(struct device *dev,

sound/soc/cirrus/ep93xx-i2s.c

@@ -404,7 +404,7 @@ static const struct snd_soc_dai_ops ep93xx_i2s_dai_ops = {
 #define EP93XX_I2S_FORMATS (SNDRV_PCM_FMTBIT_S32_LE)
 
 static struct snd_soc_dai_driver ep93xx_i2s_dai = {
-	.symmetric_rates= 1,
+	.symmetric_rate	= 1,
 	.probe		= ep93xx_i2s_dai_probe,
 	.playback	= {
 		.channels_min	= 2,

sound/soc/codecs/Kconfig

@@ -104,6 +104,7 @@ config SND_SOC_ALL_CODECS
 	imply SND_SOC_ISABELLE
 	imply SND_SOC_JZ4740_CODEC
 	imply SND_SOC_JZ4725B_CODEC
+	imply SND_SOC_JZ4760_CODEC
 	imply SND_SOC_JZ4770_CODEC
 	imply SND_SOC_LM4857
 	imply SND_SOC_LM49453
@@ -227,6 +228,8 @@ config SND_SOC_ALL_CODECS
 	imply SND_SOC_UDA1380
 	imply SND_SOC_WCD9335
 	imply SND_SOC_WCD934X
+	imply SND_SOC_LPASS_RX_MACRO
+	imply SND_SOC_LPASS_TX_MACRO
 	imply SND_SOC_WL1273
 	imply SND_SOC_WM0010
 	imply SND_SOC_WM1250_EV1
@@ -712,7 +715,7 @@ config SND_SOC_CX2072X
 	  Enable support for Conexant CX20721 and CX20723 codec chips.
 
 config SND_SOC_JZ4740_CODEC
-	depends on MIPS || COMPILE_TEST
+	depends on MACH_INGENIC || COMPILE_TEST
 	depends on OF
 	select REGMAP_MMIO
 	tristate "Ingenic JZ4740 internal CODEC"
@@ -724,7 +727,7 @@ config SND_SOC_JZ4740_CODEC
 	  will be called snd-soc-jz4740-codec.
 
 config SND_SOC_JZ4725B_CODEC
-	depends on MIPS || COMPILE_TEST
+	depends on MACH_INGENIC || COMPILE_TEST
 	depends on OF
 	select REGMAP
 	tristate "Ingenic JZ4725B internal CODEC"
@@ -735,8 +738,20 @@ config SND_SOC_JZ4725B_CODEC
 	  This driver can also be built as a module. If so, the module
 	  will be called snd-soc-jz4725b-codec.
 
+config SND_SOC_JZ4760_CODEC
+        depends on MACH_INGENIC || COMPILE_TEST
+        depends on OF
+        select REGMAP
+        tristate "Ingenic JZ4760 internal CODEC"
+        help
+          Enable support for the internal CODEC found in the JZ4760 SoC
+          from Ingenic.
+
+          This driver can also be built as a module. If so, the module
+          will be called snd-soc-jz4760-codec.
+
 config SND_SOC_JZ4770_CODEC
-	depends on MIPS || COMPILE_TEST
+	depends on MACH_INGENIC || COMPILE_TEST
 	depends on OF
 	select REGMAP
 	tristate "Ingenic JZ4770 internal CODEC"
@@ -1162,7 +1177,7 @@ config SND_SOC_RT5651
 	depends on I2C
 
 config SND_SOC_RT5659
-	tristate
+	tristate "Realtek RT5658/RT5659 Codec"
 	depends on I2C
 
 config SND_SOC_RT5660
@@ -1820,4 +1835,12 @@ config SND_SOC_LPASS_VA_MACRO
 	depends on COMMON_CLK
 	tristate "Qualcomm VA Macro in LPASS(Low Power Audio SubSystem)"
 
+config SND_SOC_LPASS_RX_MACRO
+	depends on COMMON_CLK
+	tristate "Qualcomm RX Macro in LPASS(Low Power Audio SubSystem)"
+
+config SND_SOC_LPASS_TX_MACRO
+	depends on COMMON_CLK
+	tristate "Qualcomm TX Macro in LPASS(Low Power Audio SubSystem)"
+
 endmenu

sound/soc/codecs/Makefile

@@ -101,11 +101,14 @@ snd-soc-inno-rk3036-objs := inno_rk3036.o
 snd-soc-isabelle-objs := isabelle.o
 snd-soc-jz4740-codec-objs := jz4740.o
 snd-soc-jz4725b-codec-objs := jz4725b.o
+snd-soc-jz4760-codec-objs := jz4760.o
 snd-soc-jz4770-codec-objs := jz4770.o
 snd-soc-l3-objs := l3.o
 snd-soc-lm4857-objs := lm4857.o
 snd-soc-lm49453-objs := lm49453.o
 snd-soc-lochnagar-sc-objs := lochnagar-sc.o
+snd-soc-lpass-rx-macro-objs := lpass-rx-macro.o
+snd-soc-lpass-tx-macro-objs := lpass-tx-macro.o
 snd-soc-lpass-wsa-macro-objs := lpass-wsa-macro.o
 snd-soc-lpass-va-macro-objs := lpass-va-macro.o
 snd-soc-madera-objs := madera.o
@@ -201,7 +204,6 @@ snd-soc-sigmadsp-objs := sigmadsp.o
 snd-soc-sigmadsp-i2c-objs := sigmadsp-i2c.o
 snd-soc-sigmadsp-regmap-objs := sigmadsp-regmap.o
 snd-soc-si476x-objs := si476x.o
-snd-soc-sirf-audio-codec-objs := sirf-audio-codec.o
 snd-soc-spdif-tx-objs := spdif_transmitter.o
 snd-soc-spdif-rx-objs := spdif_receiver.o
 snd-soc-ssm2305-objs := ssm2305.o
@@ -302,7 +304,6 @@ snd-soc-wm9713-objs := wm9713.o
 snd-soc-wm-hubs-objs := wm_hubs.o
 snd-soc-wsa881x-objs := wsa881x.o
 snd-soc-zl38060-objs := zl38060.o
-snd-soc-zx-aud96p22-objs := zx_aud96p22.o
 # Amp
 snd-soc-max9877-objs := max9877.o
 snd-soc-max98504-objs := max98504.o
@@ -418,6 +419,7 @@ obj-$(CONFIG_SND_SOC_INNO_RK3036)	+= snd-soc-inno-rk3036.o
 obj-$(CONFIG_SND_SOC_ISABELLE)	+= snd-soc-isabelle.o
 obj-$(CONFIG_SND_SOC_JZ4740_CODEC)	+= snd-soc-jz4740-codec.o
 obj-$(CONFIG_SND_SOC_JZ4725B_CODEC)	+= snd-soc-jz4725b-codec.o
+obj-$(CONFIG_SND_SOC_JZ4760_CODEC)      += snd-soc-jz4760-codec.o
 obj-$(CONFIG_SND_SOC_JZ4770_CODEC)	+= snd-soc-jz4770-codec.o
 obj-$(CONFIG_SND_SOC_L3)	+= snd-soc-l3.o
 obj-$(CONFIG_SND_SOC_LM4857)	+= snd-soc-lm4857.o
@@ -516,7 +518,6 @@ 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-$(CONFIG_SND_SOC_SIRF_AUDIO_CODEC) += sirf-audio-codec.o
 obj-$(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
@@ -618,7 +619,6 @@ obj-$(CONFIG_SND_SOC_WM_ADSP)	+= snd-soc-wm-adsp.o
 obj-$(CONFIG_SND_SOC_WM_HUBS)	+= snd-soc-wm-hubs.o
 obj-$(CONFIG_SND_SOC_WSA881X)	+= snd-soc-wsa881x.o
 obj-$(CONFIG_SND_SOC_ZL38060)	+= snd-soc-zl38060.o
-obj-$(CONFIG_SND_SOC_ZX_AUD96P22) += snd-soc-zx-aud96p22.o
 
 # Amp
 obj-$(CONFIG_SND_SOC_MAX9877)	+= snd-soc-max9877.o
@@ -627,6 +627,8 @@ obj-$(CONFIG_SND_SOC_SIMPLE_AMPLIFIER)	+= snd-soc-simple-amplifier.o
 obj-$(CONFIG_SND_SOC_TPA6130A2)	+= snd-soc-tpa6130a2.o
 obj-$(CONFIG_SND_SOC_LPASS_WSA_MACRO)	+= snd-soc-lpass-wsa-macro.o
 obj-$(CONFIG_SND_SOC_LPASS_VA_MACRO)	+= snd-soc-lpass-va-macro.o
+obj-$(CONFIG_SND_SOC_LPASS_RX_MACRO)	+= snd-soc-lpass-rx-macro.o
+obj-$(CONFIG_SND_SOC_LPASS_TX_MACRO)	+= snd-soc-lpass-tx-macro.o
 
 # Mux
 obj-$(CONFIG_SND_SOC_SIMPLE_MUX)	+= snd-soc-simple-mux.o

sound/soc/codecs/ab8500-codec.c

@@ -2384,7 +2384,7 @@ static struct snd_soc_dai_driver ab8500_codec_dai[] = {
 			.formats = AB8500_SUPPORTED_FMT,
 		},
 		.ops = &ab8500_codec_ops,
-		.symmetric_rates = 1
+		.symmetric_rate = 1
 	},
 	{
 		.name = "ab8500-codec-dai.1",
@@ -2397,7 +2397,7 @@ static struct snd_soc_dai_driver ab8500_codec_dai[] = {
 			.formats = AB8500_SUPPORTED_FMT,
 		},
 		.ops = &ab8500_codec_ops,
-		.symmetric_rates = 1
+		.symmetric_rate = 1
 	}
 };
 

sound/soc/codecs/adau1372.c

@@ -890,7 +890,7 @@ static struct snd_soc_dai_driver adau1372_dai_driver = {
 		.sig_bits = 24,
 	},
 	.ops = &adau1372_dai_ops,
-	.symmetric_rates = 1,
+	.symmetric_rate = 1,
 };
 
 static int adau1372_setup_pll(struct adau1372 *adau1372, unsigned int rate)

sound/soc/codecs/adau1373.c

@@ -1205,7 +1205,7 @@ static struct snd_soc_dai_driver adau1373_dai_driver[] = {
 			.formats = ADAU1373_FORMATS,
 		},
 		.ops = &adau1373_dai_ops,
-		.symmetric_rates = 1,
+		.symmetric_rate = 1,
 	},
 	{
 		.id = 1,
@@ -1225,7 +1225,7 @@ static struct snd_soc_dai_driver adau1373_dai_driver[] = {
 			.formats = ADAU1373_FORMATS,
 		},
 		.ops = &adau1373_dai_ops,
-		.symmetric_rates = 1,
+		.symmetric_rate = 1,
 	},
 	{
 		.id = 2,
@@ -1245,7 +1245,7 @@ static struct snd_soc_dai_driver adau1373_dai_driver[] = {
 			.formats = ADAU1373_FORMATS,
 		},
 		.ops = &adau1373_dai_ops,
-		.symmetric_rates = 1,
+		.symmetric_rate = 1,
 	},
 };
 

sound/soc/codecs/adau1701.c

@@ -653,7 +653,7 @@ static struct snd_soc_dai_driver adau1701_dai = {
 		.formats = ADAU1701_FORMATS,
 	},
 	.ops = &adau1701_dai_ops,
-	.symmetric_rates = 1,
+	.symmetric_rate = 1,
 };
 
 #ifdef CONFIG_OF

sound/soc/codecs/adau17x1.c

@@ -1095,8 +1095,7 @@ void adau17x1_remove(struct device *dev)
 {
 	struct adau *adau = dev_get_drvdata(dev);
 
-	if (adau->mclk)
-		clk_disable_unprepare(adau->mclk);
+	clk_disable_unprepare(adau->mclk);
 }
 EXPORT_SYMBOL_GPL(adau17x1_remove);
 

sound/soc/codecs/ak4554.c

@@ -56,7 +56,7 @@ static struct snd_soc_dai_driver ak4554_dai = {
 		.rates = SNDRV_PCM_RATE_8000_48000,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,
 	},
-	.symmetric_rates = 1,
+	.symmetric_rate = 1,
 };
 
 static const struct snd_soc_component_driver soc_component_dev_ak4554 = {

sound/soc/codecs/ak4613.c

@@ -575,7 +575,7 @@ static struct snd_soc_dai_driver ak4613_dai = {
 		.formats	= AK4613_PCM_FMTBIT,
 	},
 	.ops = &ak4613_dai_ops,
-	.symmetric_rates = 1,
+	.symmetric_rate = 1,
 };
 
 static int ak4613_suspend(struct snd_soc_component *component)

sound/soc/codecs/ak4641.c

@@ -499,7 +499,7 @@ static struct snd_soc_dai_driver ak4641_dai[] = {
 		.formats = AK4641_FORMATS,
 	},
 	.ops = &ak4641_i2s_dai_ops,
-	.symmetric_rates = 1,
+	.symmetric_rate = 1,
 },
 {
 	.name = "ak4641-voice",
@@ -519,7 +519,7 @@ static struct snd_soc_dai_driver ak4641_dai[] = {
 		.formats = AK4641_FORMATS,
 	},
 	.ops = &ak4641_pcm_dai_ops,
-	.symmetric_rates = 1,
+	.symmetric_rate = 1,
 },
 };
 

sound/soc/codecs/ak4642.c

@@ -516,7 +516,7 @@ static struct snd_soc_dai_driver ak4642_dai = {
 		.rates = SNDRV_PCM_RATE_8000_48000,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE },
 	.ops = &ak4642_dai_ops,
-	.symmetric_rates = 1,
+	.symmetric_rate = 1,
 };
 
 static int ak4642_suspend(struct snd_soc_component *component)

sound/soc/codecs/alc5632.c

@@ -1032,7 +1032,7 @@ static struct snd_soc_dai_driver alc5632_dai = {
 		.formats = ALC5632_FORMATS,},
 
 	.ops = &alc5632_dai_ops,
-	.symmetric_rates = 1,
+	.symmetric_rate = 1,
 };
 
 #ifdef CONFIG_PM

sound/soc/codecs/cpcap.c

@@ -16,6 +16,14 @@
 #include <sound/soc.h>
 #include <sound/tlv.h>
 
+/* Register 512 CPCAP_REG_VAUDIOC --- Audio Regulator and Bias Voltage */
+#define CPCAP_BIT_AUDIO_LOW_PWR           6
+#define CPCAP_BIT_AUD_LOWPWR_SPEED        5
+#define CPCAP_BIT_VAUDIOPRISTBY           4
+#define CPCAP_BIT_VAUDIO_MODE1            2
+#define CPCAP_BIT_VAUDIO_MODE0            1
+#define CPCAP_BIT_V_AUDIO_EN              0
+
 /* Register 513 CPCAP_REG_CC     --- CODEC */
 #define CPCAP_BIT_CDC_CLK2                15
 #define CPCAP_BIT_CDC_CLK1                14
@@ -221,6 +229,7 @@ struct cpcap_reg_info {
 };
 
 static const struct cpcap_reg_info cpcap_default_regs[] = {
+	{ CPCAP_REG_VAUDIOC, 0x003F, 0x0000 },
 	{ CPCAP_REG_CC, 0xFFFF, 0x0000 },
 	{ CPCAP_REG_CC, 0xFFFF, 0x0000 },
 	{ CPCAP_REG_CDI, 0xBFFF, 0x0000 },
@@ -1371,8 +1380,121 @@ static int cpcap_voice_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	return 0;
 }
 
-static int cpcap_voice_set_mute(struct snd_soc_dai *dai,
-				int mute, int direction)
+
+/*
+ * Configure codec for voice call if requested.
+ *
+ * We can configure most with snd_soc_dai_set_sysclk(), snd_soc_dai_set_fmt()
+ * and snd_soc_dai_set_tdm_slot(). This function configures the rest of the
+ * cpcap related hardware as CPU is not involved in the voice call.
+ */
+static int cpcap_voice_call(struct cpcap_audio *cpcap, struct snd_soc_dai *dai,
+			    bool voice_call)
+{
+	int mask, err;
+
+	/* Modem to codec VAUDIO_MODE1 */
+	mask = BIT(CPCAP_BIT_VAUDIO_MODE1);
+	err = regmap_update_bits(cpcap->regmap, CPCAP_REG_VAUDIOC,
+				 mask, voice_call ? mask : 0);
+	if (err)
+		return err;
+
+	/* Clear MIC1_MUX for call */
+	mask = BIT(CPCAP_BIT_MIC1_MUX);
+	err = regmap_update_bits(cpcap->regmap, CPCAP_REG_TXI,
+				 mask, voice_call ? 0 : mask);
+	if (err)
+		return err;
+
+	/* Set MIC2_MUX for call */
+	mask = BIT(CPCAP_BIT_MB_ON1L) | BIT(CPCAP_BIT_MB_ON1R) |
+		BIT(CPCAP_BIT_MIC2_MUX) | BIT(CPCAP_BIT_MIC2_PGA_EN);
+	err = regmap_update_bits(cpcap->regmap, CPCAP_REG_TXI,
+				 mask, voice_call ? mask : 0);
+	if (err)
+		return err;
+
+	/* Enable LDSP for call */
+	mask = BIT(CPCAP_BIT_A2_LDSP_L_EN) | BIT(CPCAP_BIT_A2_LDSP_R_EN);
+	err = regmap_update_bits(cpcap->regmap, CPCAP_REG_RXOA,
+				 mask, voice_call ? mask : 0);
+	if (err)
+		return err;
+
+	/* Enable CPCAP_BIT_PGA_CDC_EN for call */
+	mask = BIT(CPCAP_BIT_PGA_CDC_EN);
+	err = regmap_update_bits(cpcap->regmap, CPCAP_REG_RXCOA,
+				 mask, voice_call ? mask : 0);
+	if (err)
+		return err;
+
+	/* Unmute voice for call */
+	if (dai) {
+		err = snd_soc_dai_digital_mute(dai, !voice_call,
+					       SNDRV_PCM_STREAM_PLAYBACK);
+		if (err)
+			return err;
+	}
+
+	/* Set modem to codec mic CDC and HPF for call */
+	mask = BIT(CPCAP_BIT_MIC2_CDC_EN) | BIT(CPCAP_BIT_CDC_EN_RX) |
+	       BIT(CPCAP_BIT_AUDOHPF_1) | BIT(CPCAP_BIT_AUDOHPF_0) |
+	       BIT(CPCAP_BIT_AUDIHPF_1) | BIT(CPCAP_BIT_AUDIHPF_0);
+	err = regmap_update_bits(cpcap->regmap, CPCAP_REG_CC,
+				 mask, voice_call ? mask : 0);
+	if (err)
+		return err;
+
+	/* Enable modem to codec CDC for call*/
+	mask = BIT(CPCAP_BIT_CDC_CLK_EN);
+	err = regmap_update_bits(cpcap->regmap, CPCAP_REG_CDI,
+				 mask, voice_call ? mask : 0);
+
+	return err;
+}
+
+static int cpcap_voice_set_tdm_slot(struct snd_soc_dai *dai,
+				    unsigned int tx_mask, unsigned int rx_mask,
+				    int slots, int slot_width)
+{
+	struct snd_soc_component *component = dai->component;
+	struct cpcap_audio *cpcap = snd_soc_component_get_drvdata(component);
+	int err, ts_mask, mask;
+	bool voice_call;
+
+	/*
+	 * Primitive test for voice call, probably needs more checks
+	 * later on for 16-bit calls detected, Bluetooth headset etc.
+	 */
+	if (tx_mask == 0 && rx_mask == 1 && slot_width == 8)
+		voice_call = true;
+	else
+		voice_call = false;
+
+	ts_mask = 0x7 << CPCAP_BIT_MIC2_TIMESLOT0;
+	ts_mask |= 0x7 << CPCAP_BIT_MIC1_RX_TIMESLOT0;
+
+	mask = (tx_mask & 0x7) << CPCAP_BIT_MIC2_TIMESLOT0;
+	mask |= (rx_mask & 0x7) << CPCAP_BIT_MIC1_RX_TIMESLOT0;
+
+	err = regmap_update_bits(cpcap->regmap, CPCAP_REG_CDI,
+				 ts_mask, mask);
+	if (err)
+		return err;
+
+	err = cpcap_set_samprate(cpcap, CPCAP_DAI_VOICE, slot_width * 1000);
+	if (err)
+		return err;
+
+	err = cpcap_voice_call(cpcap, dai, voice_call);
+	if (err)
+		return err;
+
+	return 0;
+}
+
+static int cpcap_voice_set_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct snd_soc_component *component = dai->component;
 	struct cpcap_audio *cpcap = snd_soc_component_get_drvdata(component);
@@ -1393,6 +1515,7 @@ static const struct snd_soc_dai_ops cpcap_dai_voice_ops = {
 	.hw_params	= cpcap_voice_hw_params,
 	.set_sysclk	= cpcap_voice_set_dai_sysclk,
 	.set_fmt	= cpcap_voice_set_dai_fmt,
+	.set_tdm_slot	= cpcap_voice_set_tdm_slot,
 	.mute_stream	= cpcap_voice_set_mute,
 	.no_capture_mute = 1,
 };

sound/soc/codecs/cros_ec_codec.c

@@ -1011,6 +1011,18 @@ static int cros_ec_codec_platform_probe(struct platform_device *pdev)
 	}
 	priv->ec_capabilities = r.capabilities;
 
+	/* Reset EC codec i2s rx. */
+	p.cmd = EC_CODEC_I2S_RX_RESET;
+	ret = send_ec_host_command(priv->ec_device, EC_CMD_EC_CODEC_I2S_RX,
+				   (uint8_t *)&p, sizeof(p), NULL, 0);
+	if (ret == -ENOPROTOOPT) {
+		dev_info(dev,
+			 "Missing reset command. Please update EC firmware.\n");
+	} else if (ret) {
+		dev_err(dev, "failed to EC_CODEC_I2S_RESET: %d\n", ret);
+		return ret;
+	}
+
 	platform_set_drvdata(pdev, priv);
 
 	ret = devm_snd_soc_register_component(dev, &i2s_rx_component_driver,

sound/soc/codecs/cs35l32.c

@@ -194,7 +194,7 @@ static struct snd_soc_dai_driver cs35l32_dai[] = {
 			.formats = CS35L32_FORMATS,
 		},
 		.ops = &cs35l32_ops,
-		.symmetric_rates = 1,
+		.symmetric_rate = 1,
 	}
 };
 

sound/soc/codecs/cs35l33.c

@@ -691,7 +691,7 @@ static struct snd_soc_dai_driver cs35l33_dai = {
 			.formats = CS35L33_FORMATS,
 		},
 		.ops = &cs35l33_ops,
-		.symmetric_rates = 1,
+		.symmetric_rate = 1,
 };
 
 static int cs35l33_set_hg_data(struct snd_soc_component *component,

sound/soc/codecs/cs35l34.c

@@ -666,7 +666,7 @@ static struct snd_soc_dai_driver cs35l34_dai = {
 			.formats = CS35L34_FORMATS,
 		},
 		.ops = &cs35l34_ops,
-		.symmetric_rates = 1,
+		.symmetric_rate = 1,
 };
 
 static int cs35l34_boost_inductor(struct cs35l34_private *cs35l34,

sound/soc/codecs/cs35l35.c

@@ -692,7 +692,7 @@ static struct snd_soc_dai_driver cs35l35_dai[] = {
 			.formats = CS35L35_FORMATS,
 		},
 		.ops = &cs35l35_ops,
-		.symmetric_rates = 1,
+		.symmetric_rate = 1,
 	},
 	{
 		.name = "cs35l35-pdm",

sound/soc/codecs/cs35l36.c

@@ -995,7 +995,7 @@ static struct snd_soc_dai_driver cs35l36_dai[] = {
 			.formats = CS35L36_TX_FORMATS,
 		},
 		.ops = &cs35l36_ops,
-		.symmetric_rates = 1,
+		.symmetric_rate = 1,
 	},
 };
 

sound/soc/codecs/cs4234.c

@@ -585,7 +585,7 @@ static struct snd_soc_dai_driver cs4234_dai[] = {
 			.formats = CS4234_FORMATS,
 		},
 		.ops = &cs4234_dai_ops,
-		.symmetric_rates = 1,
+		.symmetric_rate = 1,
 	},
 };
 

sound/soc/codecs/cs4271.c

@@ -481,7 +481,7 @@ static struct snd_soc_dai_driver cs4271_dai = {
 		.formats	= CS4271_PCM_FORMATS,
 	},
 	.ops = &cs4271_dai_ops,
-	.symmetric_rates = 1,
+	.symmetric_rate = 1,
 };
 
 static int cs4271_reset(struct snd_soc_component *component)

sound/soc/codecs/cs42l56.c

@@ -1250,6 +1250,7 @@ static int cs42l56_i2c_probe(struct i2c_client *i2c_client,
 		dev_err(&i2c_client->dev,
 			"CS42L56 Device ID (%X). Expected %X\n",
 			devid, CS42L56_DEVID);
+		ret = -EINVAL;
 		goto err_enable;
 	}
 	alpha_rev = reg & CS42L56_AREV_MASK;
@@ -1307,7 +1308,7 @@ static int cs42l56_i2c_probe(struct i2c_client *i2c_client,
 	ret =  devm_snd_soc_register_component(&i2c_client->dev,
 			&soc_component_dev_cs42l56, &cs42l56_dai, 1);
 	if (ret < 0)
-		return ret;
+		goto err_enable;
 
 	return 0;
 

sound/soc/codecs/cs42l73.c

@@ -1181,7 +1181,7 @@ static struct snd_soc_dai_driver cs42l73_dai[] = {
 			.formats = CS42L73_FORMATS,
 		},
 		.ops = &cs42l73_ops,
-		.symmetric_rates = 1,
+		.symmetric_rate = 1,
 	 },
 	{
 		.name = "cs42l73-asp",
@@ -1201,7 +1201,7 @@ static struct snd_soc_dai_driver cs42l73_dai[] = {
 			.formats = CS42L73_FORMATS,
 		},
 		.ops = &cs42l73_ops,
-		.symmetric_rates = 1,
+		.symmetric_rate = 1,
 	 },
 	{
 		.name = "cs42l73-vsp",
@@ -1221,7 +1221,7 @@ static struct snd_soc_dai_driver cs42l73_dai[] = {
 			.formats = CS42L73_FORMATS,
 		},
 		.ops = &cs42l73_ops,
-		.symmetric_rates = 1,
+		.symmetric_rate = 1,
 	 }
 };
 

sound/soc/codecs/cs43130.c

@@ -1581,7 +1581,7 @@ static struct snd_soc_dai_driver cs43130_dai[] = {
 			.formats = CS43130_PCM_FORMATS,
 		},
 		.ops = &cs43130_pcm_ops,
-		.symmetric_rates = 1,
+		.symmetric_rate = 1,
 	},
 	{
 		.name = "cs43130-asp-dop",
@@ -1594,7 +1594,7 @@ static struct snd_soc_dai_driver cs43130_dai[] = {
 			.formats = CS43130_DOP_FORMATS,
 		},
 		.ops = &cs43130_dop_ops,
-		.symmetric_rates = 1,
+		.symmetric_rate = 1,
 	},
 	{
 		.name = "cs43130-xsp-dop",
@@ -1607,7 +1607,7 @@ static struct snd_soc_dai_driver cs43130_dai[] = {
 			.formats = CS43130_DOP_FORMATS,
 		},
 		.ops = &cs43130_dop_ops,
-		.symmetric_rates = 1,
+		.symmetric_rate = 1,
 	},
 	{
 		.name = "cs43130-xsp-dsd",

sound/soc/codecs/cs4341.c

@@ -189,7 +189,7 @@ static struct snd_soc_dai_driver cs4341_dai = {
 				  SNDRV_PCM_FMTBIT_S24_LE,
 	},
 	.ops			= &cs4341_dai_ops,
-	.symmetric_rates	= 1,
+	.symmetric_rate		= 1,
 };
 
 static const struct snd_soc_component_driver soc_component_cs4341 = {

sound/soc/codecs/cs4349.c

@@ -250,7 +250,7 @@ static struct snd_soc_dai_driver cs4349_dai = {
 		.formats	= CS4349_PCM_FORMATS,
 	},
 	.ops = &cs4349_dai_ops,
-	.symmetric_rates = 1,
+	.symmetric_rate = 1,
 };
 
 static const struct snd_soc_component_driver soc_component_dev_cs4349 = {

sound/soc/codecs/cs47l15.c

@@ -1160,8 +1160,8 @@ static struct snd_soc_dai_driver cs47l15_dai[] = {
 			.formats = MADERA_FORMATS,
 		 },
 		.ops = &madera_dai_ops,
-		.symmetric_rates = 1,
-		.symmetric_samplebits = 1,
+		.symmetric_rate = 1,
+		.symmetric_sample_bits = 1,
 	},
 	{
 		.name = "cs47l15-aif2",
@@ -1182,8 +1182,8 @@ static struct snd_soc_dai_driver cs47l15_dai[] = {
 			.formats = MADERA_FORMATS,
 		 },
 		.ops = &madera_dai_ops,
-		.symmetric_rates = 1,
-		.symmetric_samplebits = 1,
+		.symmetric_rate = 1,
+		.symmetric_sample_bits = 1,
 	},
 	{
 		.name = "cs47l15-aif3",
@@ -1204,8 +1204,8 @@ static struct snd_soc_dai_driver cs47l15_dai[] = {
 			.formats = MADERA_FORMATS,
 		 },
 		.ops = &madera_dai_ops,
-		.symmetric_rates = 1,
-		.symmetric_samplebits = 1,
+		.symmetric_rate = 1,
+		.symmetric_sample_bits = 1,
 	},
 	{
 		.name = "cs47l15-cpu-trace",

sound/soc/codecs/cs47l24.c

@@ -977,8 +977,8 @@ static struct snd_soc_dai_driver cs47l24_dai[] = {
 			 .formats = CS47L24_FORMATS,
 		 },
 		.ops = &arizona_dai_ops,
-		.symmetric_rates = 1,
-		.symmetric_samplebits = 1,
+		.symmetric_rate = 1,
+		.symmetric_sample_bits = 1,
 	},
 	{
 		.name = "cs47l24-aif2",
@@ -999,8 +999,8 @@ static struct snd_soc_dai_driver cs47l24_dai[] = {
 			 .formats = CS47L24_FORMATS,
 		 },
 		.ops = &arizona_dai_ops,
-		.symmetric_rates = 1,
-		.symmetric_samplebits = 1,
+		.symmetric_rate = 1,
+		.symmetric_sample_bits = 1,
 	},
 	{
 		.name = "cs47l24-aif3",
@@ -1021,8 +1021,8 @@ static struct snd_soc_dai_driver cs47l24_dai[] = {
 			 .formats = CS47L24_FORMATS,
 		 },
 		.ops = &arizona_dai_ops,
-		.symmetric_rates = 1,
-		.symmetric_samplebits = 1,
+		.symmetric_rate = 1,
+		.symmetric_sample_bits = 1,
 	},
 	{
 		.name = "cs47l24-cpu-voicectrl",

sound/soc/codecs/cs47l35.c

@@ -1368,8 +1368,8 @@ static struct snd_soc_dai_driver cs47l35_dai[] = {
 			.formats = MADERA_FORMATS,
 		 },
 		.ops = &madera_dai_ops,
-		.symmetric_rates = 1,
-		.symmetric_samplebits = 1,
+		.symmetric_rate = 1,
+		.symmetric_sample_bits = 1,
 	},
 	{
 		.name = "cs47l35-aif2",
@@ -1390,8 +1390,8 @@ static struct snd_soc_dai_driver cs47l35_dai[] = {
 			.formats = MADERA_FORMATS,
 		 },
 		.ops = &madera_dai_ops,
-		.symmetric_rates = 1,
-		.symmetric_samplebits = 1,
+		.symmetric_rate = 1,
+		.symmetric_sample_bits = 1,
 	},
 	{
 		.name = "cs47l35-aif3",
@@ -1412,8 +1412,8 @@ static struct snd_soc_dai_driver cs47l35_dai[] = {
 			.formats = MADERA_FORMATS,
 		 },
 		.ops = &madera_dai_ops,
-		.symmetric_rates = 1,
-		.symmetric_samplebits = 1,
+		.symmetric_rate = 1,
+		.symmetric_sample_bits = 1,
 	},
 	{
 		.name = "cs47l35-slim1",

sound/soc/codecs/cs47l85.c

@@ -2269,8 +2269,8 @@ static struct snd_soc_dai_driver cs47l85_dai[] = {
 			.formats = MADERA_FORMATS,
 		 },
 		.ops = &madera_dai_ops,
-		.symmetric_rates = 1,
-		.symmetric_samplebits = 1,
+		.symmetric_rate = 1,
+		.symmetric_sample_bits = 1,
 	},
 	{
 		.name = "cs47l85-aif2",
@@ -2291,8 +2291,8 @@ static struct snd_soc_dai_driver cs47l85_dai[] = {
 			.formats = MADERA_FORMATS,
 		 },
 		.ops = &madera_dai_ops,
-		.symmetric_rates = 1,
-		.symmetric_samplebits = 1,
+		.symmetric_rate = 1,
+		.symmetric_sample_bits = 1,
 	},
 	{
 		.name = "cs47l85-aif3",
@@ -2313,8 +2313,8 @@ static struct snd_soc_dai_driver cs47l85_dai[] = {
 			.formats = MADERA_FORMATS,
 		 },
 		.ops = &madera_dai_ops,
-		.symmetric_rates = 1,
-		.symmetric_samplebits = 1,
+		.symmetric_rate = 1,
+		.symmetric_sample_bits = 1,
 	},
 	{
 		.name = "cs47l85-aif4",
@@ -2335,8 +2335,8 @@ static struct snd_soc_dai_driver cs47l85_dai[] = {
 			.formats = MADERA_FORMATS,
 		 },
 		.ops = &madera_dai_ops,
-		.symmetric_rates = 1,
-		.symmetric_samplebits = 1,
+		.symmetric_rate = 1,
+		.symmetric_sample_bits = 1,
 	},
 	{
 		.name = "cs47l85-slim1",

sound/soc/codecs/cs47l90.c

@@ -2188,8 +2188,8 @@ static struct snd_soc_dai_driver cs47l90_dai[] = {
 			.formats = MADERA_FORMATS,
 		 },
 		.ops = &madera_dai_ops,
-		.symmetric_rates = 1,
-		.symmetric_samplebits = 1,
+		.symmetric_rate = 1,
+		.symmetric_sample_bits = 1,
 	},
 	{
 		.name = "cs47l90-aif2",
@@ -2210,8 +2210,8 @@ static struct snd_soc_dai_driver cs47l90_dai[] = {
 			.formats = MADERA_FORMATS,
 		 },
 		.ops = &madera_dai_ops,
-		.symmetric_rates = 1,
-		.symmetric_samplebits = 1,
+		.symmetric_rate = 1,
+		.symmetric_sample_bits = 1,
 	},
 	{
 		.name = "cs47l90-aif3",
@@ -2232,8 +2232,8 @@ static struct snd_soc_dai_driver cs47l90_dai[] = {
 			.formats = MADERA_FORMATS,
 		 },
 		.ops = &madera_dai_ops,
-		.symmetric_rates = 1,
-		.symmetric_samplebits = 1,
+		.symmetric_rate = 1,
+		.symmetric_sample_bits = 1,
 	},
 	{
 		.name = "cs47l90-aif4",
@@ -2254,8 +2254,8 @@ static struct snd_soc_dai_driver cs47l90_dai[] = {
 			.formats = MADERA_FORMATS,
 		 },
 		.ops = &madera_dai_ops,
-		.symmetric_rates = 1,
-		.symmetric_samplebits = 1,
+		.symmetric_rate = 1,
+		.symmetric_sample_bits = 1,
 	},
 	{
 		.name = "cs47l90-slim1",

sound/soc/codecs/cs47l92.c

@@ -1704,8 +1704,8 @@ static struct snd_soc_dai_driver cs47l92_dai[] = {
 			.formats = MADERA_FORMATS,
 		 },
 		.ops = &madera_dai_ops,
-		.symmetric_rates = 1,
-		.symmetric_samplebits = 1,
+		.symmetric_rate = 1,
+		.symmetric_sample_bits = 1,
 	},
 	{
 		.name = "cs47l92-aif2",
@@ -1726,8 +1726,8 @@ static struct snd_soc_dai_driver cs47l92_dai[] = {
 			.formats = MADERA_FORMATS,
 		 },
 		.ops = &madera_dai_ops,
-		.symmetric_rates = 1,
-		.symmetric_samplebits = 1,
+		.symmetric_rate = 1,
+		.symmetric_sample_bits = 1,
 	},
 	{
 		.name = "cs47l92-aif3",
@@ -1748,8 +1748,8 @@ static struct snd_soc_dai_driver cs47l92_dai[] = {
 			.formats = MADERA_FORMATS,
 		 },
 		.ops = &madera_dai_ops,
-		.symmetric_rates = 1,
-		.symmetric_samplebits = 1,
+		.symmetric_rate = 1,
+		.symmetric_sample_bits = 1,
 	},
 	{
 		.name = "cs47l92-slim1",

sound/soc/codecs/cs53l30.c

@@ -869,7 +869,7 @@ static struct snd_soc_dai_driver cs53l30_dai = {
 		.formats = CS53L30_FORMATS,
 	},
 	.ops = &cs53l30_ops,
-	.symmetric_rates = 1,
+	.symmetric_rate = 1,
 };
 
 static int cs53l30_component_probe(struct snd_soc_component *component)

sound/soc/codecs/cx2072x.c

@@ -1572,7 +1572,7 @@ static struct snd_soc_dai_driver soc_codec_cx2072x_dai[] = {
 			.formats = CX2072X_FORMATS,
 		},
 		.ops = &cx2072x_dai_ops,
-		.symmetric_rates = 1,
+		.symmetric_rate = 1,
 	},
 	{ /* plabayck only, return echo reference to Conexant DSP chip */
 		.name = "cx2072x-dsp",

sound/soc/codecs/da7210.c

@@ -1059,7 +1059,7 @@ static struct snd_soc_dai_driver da7210_dai = {
 		.formats = DA7210_FORMATS,
 	},
 	.ops = &da7210_dai_ops,
-	.symmetric_rates = 1,
+	.symmetric_rate = 1,
 };
 
 static int da7210_probe(struct snd_soc_component *component)

sound/soc/codecs/da7213.c

@@ -1551,7 +1551,7 @@ static struct snd_soc_dai_driver da7213_dai = {
 		.formats = DA7213_FORMATS,
 	},
 	.ops = &da7213_dai_ops,
-	.symmetric_rates = 1,
+	.symmetric_rate = 1,
 };
 
 static int da7213_set_auto_pll(struct snd_soc_component *component, bool enable)

sound/soc/codecs/da7218.c

@@ -2194,9 +2194,9 @@ static struct snd_soc_dai_driver da7218_dai = {
 		.formats = DA7218_FORMATS,
 	},
 	.ops = &da7218_dai_ops,
-	.symmetric_rates = 1,
+	.symmetric_rate = 1,
 	.symmetric_channels = 1,
-	.symmetric_samplebits = 1,
+	.symmetric_sample_bits = 1,
 };
 
 
@@ -2278,14 +2278,12 @@ static irqreturn_t da7218_irq_thread(int irq, void *data)
  * DT
  */
 
-#ifdef CONFIG_OF
 static const struct of_device_id da7218_of_match[] = {
 	{ .compatible = "dlg,da7217", .data = (void *) DA7217_DEV_ID },
 	{ .compatible = "dlg,da7218", .data = (void *) DA7218_DEV_ID },
 	{ }
 };
 MODULE_DEVICE_TABLE(of, da7218_of_match);
-#endif
 
 static inline int da7218_of_get_id(struct device *dev)
 {
@@ -3311,7 +3309,7 @@ MODULE_DEVICE_TABLE(i2c, da7218_i2c_id);
 static struct i2c_driver da7218_i2c_driver = {
 	.driver = {
 		.name = "da7218",
-		.of_match_table = of_match_ptr(da7218_of_match),
+		.of_match_table = da7218_of_match,
 	},
 	.probe		= da7218_i2c_probe,
 	.id_table	= da7218_i2c_id,

sound/soc/codecs/da7219.c

@@ -1692,9 +1692,9 @@ static struct snd_soc_dai_driver da7219_dai = {
 		.formats = DA7219_FORMATS,
 	},
 	.ops = &da7219_dai_ops,
-	.symmetric_rates = 1,
+	.symmetric_rate = 1,
 	.symmetric_channels = 1,
-	.symmetric_samplebits = 1,
+	.symmetric_sample_bits = 1,
 };
 
 

sound/soc/codecs/da9055.c

@@ -1347,7 +1347,7 @@ static struct snd_soc_dai_driver da9055_dai = {
 		.formats = DA9055_FORMATS,
 	},
 	.ops = &da9055_dai_ops,
-	.symmetric_rates = 1,
+	.symmetric_rate = 1,
 };
 
 static int da9055_set_bias_level(struct snd_soc_component *component,

sound/soc/codecs/es8316.c

@@ -543,7 +543,7 @@ static struct snd_soc_dai_driver es8316_dai = {
 		.formats = ES8316_FORMATS,
 	},
 	.ops = &es8316_ops,
-	.symmetric_rates = 1,
+	.symmetric_rate = 1,
 };
 
 static void es8316_enable_micbias_for_mic_gnd_short_detect(
@@ -681,6 +681,9 @@ static void es8316_disable_jack_detect(struct snd_soc_component *component)
 {
 	struct es8316_priv *es8316 = snd_soc_component_get_drvdata(component);
 
+	if (!es8316->jack)
+		return; /* Already disabled (or never enabled) */
+
 	disable_irq(es8316->irq);
 
 	mutex_lock(&es8316->lock);

sound/soc/codecs/es8328.c

@@ -715,7 +715,7 @@ static struct snd_soc_dai_driver es8328_dai = {
 		.formats = ES8328_FORMATS,
 	},
 	.ops = &es8328_dai_ops,
-	.symmetric_rates = 1,
+	.symmetric_rate = 1,
 };
 
 static int es8328_suspend(struct snd_soc_component *component)
@@ -809,8 +809,7 @@ static void es8328_remove(struct snd_soc_component *component)
 
 	es8328 = snd_soc_component_get_drvdata(component);
 
-	if (es8328->clk)
-		clk_disable_unprepare(es8328->clk);
+	clk_disable_unprepare(es8328->clk);
 
 	regulator_bulk_disable(ARRAY_SIZE(es8328->supplies),
 			       es8328->supplies);

sound/soc/codecs/hdmi-codec.c

@@ -489,6 +489,7 @@ static int hdmi_codec_hw_params(struct snd_pcm_substream *substream,
 	hp.sample_rate = params_rate(params);
 	hp.channels = params_channels(params);
 
+	cf->bit_fmt = params_format(params);
 	return hcp->hcd.ops->hw_params(dai->dev->parent, hcp->hcd.data,
 				       cf, &hp);
 }
@@ -617,7 +618,8 @@ static const struct snd_soc_dai_ops hdmi_codec_spdif_dai_ops = {
 			 SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE |\
 			 SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE |\
 			 SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE |\
-			 SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE)
+			 SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE |\
+			 SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE)
 
 static int hdmi_codec_pcm_new(struct snd_soc_pcm_runtime *rtd,
 			      struct snd_soc_dai *dai)

sound/soc/codecs/inno_rk3036.c

@@ -325,7 +325,7 @@ static struct snd_soc_dai_driver rk3036_codec_dai_driver[] = {
 			.formats = RK3036_CODEC_FMTS,
 		},
 		.ops = &rk3036_codec_dai_ops,
-		.symmetric_rates = 1,
+		.symmetric_rate = 1,
 	},
 };
 

sound/soc/codecs/jz4740.c

@@ -214,7 +214,7 @@ static struct snd_soc_dai_driver jz4740_codec_dai = {
 		.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8,
 	},
 	.ops = &jz4740_codec_dai_ops,
-	.symmetric_rates = 1,
+	.symmetric_rate = 1,
 };
 
 static void jz4740_codec_wakeup(struct regmap *regmap)

sound/soc/codecs/jz4760.c

@@ -0,0 +1,889 @@
+// SPDX-License-Identifier: GPL-2.0
+//
+// Ingenic JZ4760 CODEC driver
+//
+// Copyright (C) 2021, Christophe Branchereau <cbranchereau@gmail.com>
+// Copyright (C) 2021, Paul Cercueil <paul@crapouillou.net>
+
+#include <linux/bitfield.h>
+#include <linux/clk.h>
+#include <linux/delay.h>
+#include <linux/iopoll.h>
+#include <linux/module.h>
+#include <linux/regmap.h>
+#include <linux/time64.h>
+
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+#include <sound/soc-dapm.h>
+#include <sound/tlv.h>
+
+#define ICDC_RGADW_OFFSET		0x00
+#define ICDC_RGDATA_OFFSET		0x04
+
+/* ICDC internal register access control register(RGADW) */
+#define ICDC_RGADW_RGWR			BIT(16)
+#define	ICDC_RGADW_RGADDR_MASK		GENMASK(14, 8)
+#define	ICDC_RGADW_RGDIN_MASK		GENMASK(7, 0)
+
+/* ICDC internal register data output register (RGDATA)*/
+#define ICDC_RGDATA_IRQ			BIT(8)
+#define ICDC_RGDATA_RGDOUT_MASK		GENMASK(7, 0)
+
+/* Internal register space, accessed through regmap */
+enum {
+	JZ4760_CODEC_REG_SR,
+	JZ4760_CODEC_REG_AICR,
+	JZ4760_CODEC_REG_CR1,
+	JZ4760_CODEC_REG_CR2,
+	JZ4760_CODEC_REG_CR3,
+	JZ4760_CODEC_REG_CR4,
+	JZ4760_CODEC_REG_CCR1,
+	JZ4760_CODEC_REG_CCR2,
+	JZ4760_CODEC_REG_PMR1,
+	JZ4760_CODEC_REG_PMR2,
+	JZ4760_CODEC_REG_ICR,
+	JZ4760_CODEC_REG_IFR,
+	JZ4760_CODEC_REG_GCR1,
+	JZ4760_CODEC_REG_GCR2,
+	JZ4760_CODEC_REG_GCR3,
+	JZ4760_CODEC_REG_GCR4,
+	JZ4760_CODEC_REG_GCR5,
+	JZ4760_CODEC_REG_GCR6,
+	JZ4760_CODEC_REG_GCR7,
+	JZ4760_CODEC_REG_GCR8,
+	JZ4760_CODEC_REG_GCR9,
+	JZ4760_CODEC_REG_AGC1,
+	JZ4760_CODEC_REG_AGC2,
+	JZ4760_CODEC_REG_AGC3,
+	JZ4760_CODEC_REG_AGC4,
+	JZ4760_CODEC_REG_AGC5,
+	JZ4760_CODEC_REG_MIX1,
+	JZ4760_CODEC_REG_MIX2,
+};
+
+#define REG_AICR_DAC_ADWL_MASK		GENMASK(7, 6)
+#define REG_AICR_DAC_SERIAL		BIT(3)
+#define REG_AICR_DAC_I2S		BIT(1)
+
+#define REG_AICR_ADC_ADWL_MASK		GENMASK(5, 4)
+
+#define REG_AICR_ADC_SERIAL		BIT(2)
+#define REG_AICR_ADC_I2S		BIT(0)
+
+#define REG_CR1_HP_LOAD			BIT(7)
+#define REG_CR1_HP_MUTE			BIT(5)
+#define REG_CR1_LO_MUTE_OFFSET		4
+#define REG_CR1_BTL_MUTE_OFFSET		3
+#define REG_CR1_OUTSEL_OFFSET		0
+#define REG_CR1_OUTSEL_MASK		GENMASK(1, REG_CR1_OUTSEL_OFFSET)
+
+#define REG_CR2_DAC_MONO		BIT(7)
+#define REG_CR2_DAC_MUTE		BIT(5)
+#define REG_CR2_DAC_NOMAD		BIT(1)
+#define REG_CR2_DAC_RIGHT_ONLY		BIT(0)
+
+#define REG_CR3_ADC_INSEL_OFFSET	2
+#define REG_CR3_ADC_INSEL_MASK		GENMASK(3, REG_CR3_ADC_INSEL_OFFSET)
+#define REG_CR3_MICSTEREO_OFFSET	1
+#define REG_CR3_MICDIFF_OFFSET		0
+
+#define REG_CR4_ADC_HPF_OFFSET		7
+#define REG_CR4_ADC_RIGHT_ONLY		BIT(0)
+
+#define REG_CCR1_CRYSTAL_MASK		GENMASK(3, 0)
+
+#define REG_CCR2_DAC_FREQ_MASK		GENMASK(7, 4)
+#define REG_CCR2_ADC_FREQ_MASK		GENMASK(3, 0)
+
+#define REG_PMR1_SB			BIT(7)
+#define REG_PMR1_SB_SLEEP		BIT(6)
+#define REG_PMR1_SB_AIP_OFFSET		5
+#define REG_PMR1_SB_LINE_OFFSET		4
+#define REG_PMR1_SB_MIC1_OFFSET		3
+#define REG_PMR1_SB_MIC2_OFFSET		2
+#define REG_PMR1_SB_BYPASS_OFFSET	1
+#define REG_PMR1_SB_MICBIAS_OFFSET	0
+
+#define REG_PMR2_SB_ADC_OFFSET		4
+#define REG_PMR2_SB_HP_OFFSET		3
+#define REG_PMR2_SB_BTL_OFFSET		2
+#define REG_PMR2_SB_LOUT_OFFSET		1
+#define REG_PMR2_SB_DAC_OFFSET		0
+
+#define REG_ICR_INT_FORM_MASK		GENMASK(7, 6)
+#define REG_ICR_ALL_MASK		GENMASK(5, 0)
+#define REG_ICR_JACK_MASK		BIT(5)
+#define REG_ICR_SCMC_MASK		BIT(4)
+#define REG_ICR_RUP_MASK		BIT(3)
+#define REG_ICR_RDO_MASK		BIT(2)
+#define REG_ICR_GUP_MASK		BIT(1)
+#define REG_ICR_GDO_MASK		BIT(0)
+
+#define REG_IFR_ALL_MASK		GENMASK(5, 0)
+#define REG_IFR_JACK			BIT(6)
+#define REG_IFR_JACK_EVENT		BIT(5)
+#define REG_IFR_SCMC			BIT(4)
+#define REG_IFR_RUP			BIT(3)
+#define REG_IFR_RDO			BIT(2)
+#define REG_IFR_GUP			BIT(1)
+#define REG_IFR_GDO			BIT(0)
+
+#define REG_GCR_GAIN_OFFSET		0
+#define REG_GCR_GAIN_MAX		0x1f
+
+#define REG_GCR_RL			BIT(7)
+
+#define REG_GCR_GIM1_MASK		GENMASK(5, 3)
+#define REG_GCR_GIM2_MASK		GENMASK(2, 0)
+#define REG_GCR_GIM_GAIN_MAX		7
+
+#define REG_AGC1_EN			BIT(7)
+#define REG_AGC1_TARGET_MASK		GENMASK(5, 2)
+
+#define REG_AGC2_NG_THR_MASK		GENMASK(6, 4)
+#define REG_AGC2_HOLD_MASK		GENMASK(3, 0)
+
+#define REG_AGC3_ATK_MASK		GENMASK(7, 4)
+#define REG_AGC3_DCY_MASK		GENMASK(3, 0)
+
+#define REG_AGC4_AGC_MAX_MASK		GENMASK(4, 0)
+
+#define REG_AGC5_AGC_MIN_MASK		GENMASK(4, 0)
+
+#define REG_MIX1_MIX_REC_MASK		GENMASK(7, 6)
+#define REG_MIX1_GIMIX_MASK		GENMASK(4, 0)
+
+#define REG_MIX2_DAC_MIX_MASK		GENMASK(7, 6)
+#define REG_MIX2_GOMIX_MASK		GENMASK(4, 0)
+
+/* codec private data */
+struct jz_codec {
+	struct device *dev;
+	struct regmap *regmap;
+	void __iomem *base;
+	struct clk *clk;
+};
+
+static int jz4760_codec_set_bias_level(struct snd_soc_component *codec,
+				       enum snd_soc_bias_level level)
+{
+	struct jz_codec *jz_codec = snd_soc_component_get_drvdata(codec);
+	struct regmap *regmap = jz_codec->regmap;
+
+	switch (level) {
+	case SND_SOC_BIAS_PREPARE:
+		/* Reset all interrupt flags. */
+		regmap_write(regmap, JZ4760_CODEC_REG_IFR, REG_IFR_ALL_MASK);
+
+		regmap_clear_bits(regmap, JZ4760_CODEC_REG_PMR1, REG_PMR1_SB);
+		msleep(250);
+		regmap_clear_bits(regmap, JZ4760_CODEC_REG_PMR1, REG_PMR1_SB_SLEEP);
+		msleep(400);
+		break;
+	case SND_SOC_BIAS_STANDBY:
+		regmap_set_bits(regmap, JZ4760_CODEC_REG_PMR1, REG_PMR1_SB_SLEEP);
+		regmap_set_bits(regmap, JZ4760_CODEC_REG_PMR1, REG_PMR1_SB);
+		break;
+	default:
+		break;
+	}
+
+	return 0;
+}
+
+static int jz4760_codec_startup(struct snd_pcm_substream *substream,
+				struct snd_soc_dai *dai)
+{
+	struct snd_soc_component *codec = dai->component;
+	struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(codec);
+	int ret;
+
+	/*
+	 * SYSCLK output from the codec to the AIC is required to keep the
+	 * DMA transfer going during playback when all audible outputs have
+	 * been disabled.
+	 */
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		ret = snd_soc_dapm_force_enable_pin(dapm, "SYSCLK");
+	return 0;
+}
+
+static void jz4760_codec_shutdown(struct snd_pcm_substream *substream,
+				  struct snd_soc_dai *dai)
+{
+	struct snd_soc_component *codec = dai->component;
+	struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(codec);
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		snd_soc_dapm_disable_pin(dapm, "SYSCLK");
+}
+
+
+static int jz4760_codec_pcm_trigger(struct snd_pcm_substream *substream,
+				    int cmd, struct snd_soc_dai *dai)
+{
+	struct snd_soc_component *codec = dai->component;
+	int ret = 0;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
+			snd_soc_component_force_bias_level(codec, SND_SOC_BIAS_ON);
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		/* do nothing */
+		break;
+	default:
+		ret = -EINVAL;
+	}
+
+	return ret;
+}
+
+static int jz4760_codec_mute_stream(struct snd_soc_dai *dai, int mute, int direction)
+{
+	struct snd_soc_component *codec = dai->component;
+	struct jz_codec *jz_codec = snd_soc_component_get_drvdata(codec);
+	unsigned int gain_bit = mute ? REG_IFR_GDO : REG_IFR_GUP;
+	unsigned int val, reg;
+	int change, err;
+
+	change = snd_soc_component_update_bits(codec, JZ4760_CODEC_REG_CR2,
+					       REG_CR2_DAC_MUTE,
+					       mute ? REG_CR2_DAC_MUTE : 0);
+	if (change == 1) {
+		regmap_read(jz_codec->regmap, JZ4760_CODEC_REG_PMR2, &val);
+
+		if (val & BIT(REG_PMR2_SB_DAC_OFFSET))
+			return 1;
+
+		err = regmap_read_poll_timeout(jz_codec->regmap,
+					       JZ4760_CODEC_REG_IFR,
+					       val, val & gain_bit,
+					       1000, 1 * USEC_PER_SEC);
+		if (err) {
+			dev_err(jz_codec->dev,
+				"Timeout while setting digital mute: %d", err);
+			return err;
+		}
+
+		/* clear GUP/GDO flag */
+		regmap_write(jz_codec->regmap, JZ4760_CODEC_REG_IFR, gain_bit);
+	}
+
+	regmap_read(jz_codec->regmap, JZ4760_CODEC_REG_CR2, &reg);
+
+	return 0;
+}
+
+/* unit: 0.01dB */
+static const DECLARE_TLV_DB_MINMAX_MUTE(dac_tlv, -3100, 100);
+static const DECLARE_TLV_DB_SCALE(adc_tlv, 0, 100, 0);
+static const DECLARE_TLV_DB_MINMAX(out_tlv, -2500, 100);
+static const DECLARE_TLV_DB_SCALE(linein_tlv, -2500, 100, 0);
+
+/* Unconditional controls. */
+static const struct snd_kcontrol_new jz4760_codec_snd_controls[] = {
+	/* record gain control */
+	SOC_DOUBLE_R_TLV("PCM Capture Volume",
+			 JZ4760_CODEC_REG_GCR9, JZ4760_CODEC_REG_GCR8,
+			 REG_GCR_GAIN_OFFSET, REG_GCR_GAIN_MAX, 0, adc_tlv),
+
+	SOC_DOUBLE_R_TLV("Line In Bypass Playback Volume",
+			 JZ4760_CODEC_REG_GCR4, JZ4760_CODEC_REG_GCR3,
+			 REG_GCR_GAIN_OFFSET, REG_GCR_GAIN_MAX, 1, linein_tlv),
+
+	SOC_SINGLE("High-Pass Filter Capture Switch",
+		   JZ4760_CODEC_REG_CR4,
+		   REG_CR4_ADC_HPF_OFFSET, 1, 0),
+};
+
+static const struct snd_kcontrol_new jz4760_codec_pcm_playback_controls[] = {
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Volume",
+		.info = snd_soc_info_volsw,
+		.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ
+			| SNDRV_CTL_ELEM_ACCESS_READWRITE,
+		.tlv.p = dac_tlv,
+		.get = snd_soc_dapm_get_volsw,
+		.put = snd_soc_dapm_put_volsw,
+		.private_value = SOC_DOUBLE_R_VALUE(JZ4760_CODEC_REG_GCR6,
+						    JZ4760_CODEC_REG_GCR5,
+						    REG_GCR_GAIN_OFFSET,
+						    REG_GCR_GAIN_MAX, 1),
+	},
+};
+
+static const struct snd_kcontrol_new jz4760_codec_hp_playback_controls[] = {
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Volume",
+		.info = snd_soc_info_volsw,
+		.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ
+			| SNDRV_CTL_ELEM_ACCESS_READWRITE,
+		.tlv.p = out_tlv,
+		.get = snd_soc_dapm_get_volsw,
+		.put = snd_soc_dapm_put_volsw,
+		.private_value = SOC_DOUBLE_R_VALUE(JZ4760_CODEC_REG_GCR2,
+						    JZ4760_CODEC_REG_GCR1,
+						    REG_GCR_GAIN_OFFSET,
+						    REG_GCR_GAIN_MAX, 1),
+	},
+};
+
+static int hpout_event(struct snd_soc_dapm_widget *w,
+		       struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_component *codec = snd_soc_dapm_to_component(w->dapm);
+	struct jz_codec *jz_codec = snd_soc_component_get_drvdata(codec);
+	unsigned int val;
+	int err;
+
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		/* unmute HP */
+		regmap_clear_bits(jz_codec->regmap, JZ4760_CODEC_REG_CR1,
+				  REG_CR1_HP_MUTE);
+		break;
+
+	case SND_SOC_DAPM_POST_PMU:
+		/* wait for ramp-up complete (RUP) */
+		err = regmap_read_poll_timeout(jz_codec->regmap,
+					       JZ4760_CODEC_REG_IFR,
+					       val, val & REG_IFR_RUP,
+					       1000, 1 * USEC_PER_SEC);
+		if (err) {
+			dev_err(jz_codec->dev, "RUP timeout: %d", err);
+			return err;
+		}
+
+		/* clear RUP flag */
+		regmap_set_bits(jz_codec->regmap, JZ4760_CODEC_REG_IFR,
+				REG_IFR_RUP);
+
+		break;
+
+	case SND_SOC_DAPM_POST_PMD:
+		/* mute HP */
+		regmap_set_bits(jz_codec->regmap, JZ4760_CODEC_REG_CR1,
+				REG_CR1_HP_MUTE);
+
+		err = regmap_read_poll_timeout(jz_codec->regmap,
+					       JZ4760_CODEC_REG_IFR,
+					       val, val & REG_IFR_RDO,
+					       1000, 1 * USEC_PER_SEC);
+		if (err) {
+			dev_err(jz_codec->dev, "RDO timeout: %d", err);
+			return err;
+		}
+
+		/* clear RDO flag */
+		regmap_set_bits(jz_codec->regmap, JZ4760_CODEC_REG_IFR,
+				REG_IFR_RDO);
+
+		break;
+	}
+
+	return 0;
+}
+
+static const char * const jz4760_codec_hp_texts[] = {
+	"PCM", "Line In", "Mic 1", "Mic 2"
+};
+
+static const unsigned int jz4760_codec_hp_values[] = { 3, 2, 0, 1 };
+
+static SOC_VALUE_ENUM_SINGLE_DECL(jz4760_codec_hp_enum,
+				  JZ4760_CODEC_REG_CR1,
+				  REG_CR1_OUTSEL_OFFSET,
+				  REG_CR1_OUTSEL_MASK >> REG_CR1_OUTSEL_OFFSET,
+				  jz4760_codec_hp_texts,
+				  jz4760_codec_hp_values);
+static const struct snd_kcontrol_new jz4760_codec_hp_source =
+			SOC_DAPM_ENUM("Route", jz4760_codec_hp_enum);
+
+static const char * const jz4760_codec_cap_texts[] = {
+	"Line In", "Mic 1", "Mic 2"
+};
+
+static const unsigned int jz4760_codec_cap_values[] = { 2, 0, 1 };
+
+static SOC_VALUE_ENUM_SINGLE_DECL(jz4760_codec_cap_enum,
+				  JZ4760_CODEC_REG_CR3,
+				  REG_CR3_ADC_INSEL_OFFSET,
+				  REG_CR3_ADC_INSEL_MASK >> REG_CR3_ADC_INSEL_OFFSET,
+				  jz4760_codec_cap_texts,
+				  jz4760_codec_cap_values);
+static const struct snd_kcontrol_new jz4760_codec_cap_source =
+			SOC_DAPM_ENUM("Route", jz4760_codec_cap_enum);
+
+static const struct snd_kcontrol_new jz4760_codec_mic_controls[] = {
+	SOC_DAPM_SINGLE("Stereo Capture Switch", JZ4760_CODEC_REG_CR3,
+			REG_CR3_MICSTEREO_OFFSET, 1, 0),
+};
+
+static const struct snd_kcontrol_new jz4760_codec_line_out_switch =
+	SOC_DAPM_SINGLE("Switch", JZ4760_CODEC_REG_CR1,
+			REG_CR1_LO_MUTE_OFFSET, 0, 0);
+static const struct snd_kcontrol_new jz4760_codec_btl_out_switch =
+	SOC_DAPM_SINGLE("Switch", JZ4760_CODEC_REG_CR1,
+			REG_CR1_BTL_MUTE_OFFSET, 0, 0);
+
+static const struct snd_soc_dapm_widget jz4760_codec_dapm_widgets[] = {
+	SND_SOC_DAPM_PGA_E("HP Out", JZ4760_CODEC_REG_PMR2,
+			   REG_PMR2_SB_HP_OFFSET, 1, NULL, 0, hpout_event,
+			   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+			   SND_SOC_DAPM_POST_PMD),
+
+	SND_SOC_DAPM_SWITCH("Line Out", JZ4760_CODEC_REG_PMR2,
+			    REG_PMR2_SB_LOUT_OFFSET, 1,
+			    &jz4760_codec_line_out_switch),
+
+	SND_SOC_DAPM_SWITCH("BTL Out", JZ4760_CODEC_REG_PMR2,
+			    REG_PMR2_SB_BTL_OFFSET, 1,
+			    &jz4760_codec_btl_out_switch),
+
+	SND_SOC_DAPM_PGA("Line In", JZ4760_CODEC_REG_PMR1,
+			 REG_PMR1_SB_LINE_OFFSET, 1, NULL, 0),
+
+	SND_SOC_DAPM_MUX("Headphones Source", SND_SOC_NOPM, 0, 0,
+			 &jz4760_codec_hp_source),
+
+	SND_SOC_DAPM_MUX("Capture Source", SND_SOC_NOPM, 0, 0,
+			 &jz4760_codec_cap_source),
+
+	SND_SOC_DAPM_PGA("Mic 1", JZ4760_CODEC_REG_PMR1,
+			 REG_PMR1_SB_MIC1_OFFSET, 1, NULL, 0),
+
+	SND_SOC_DAPM_PGA("Mic 2", JZ4760_CODEC_REG_PMR1,
+			 REG_PMR1_SB_MIC2_OFFSET, 1, NULL, 0),
+
+	SND_SOC_DAPM_PGA("Mic Diff", JZ4760_CODEC_REG_CR3,
+			 REG_CR3_MICDIFF_OFFSET, 0, NULL, 0),
+
+	SND_SOC_DAPM_MIXER("Mic", SND_SOC_NOPM, 0, 0,
+			   jz4760_codec_mic_controls,
+			   ARRAY_SIZE(jz4760_codec_mic_controls)),
+
+	SND_SOC_DAPM_PGA("Line In Bypass", JZ4760_CODEC_REG_PMR1,
+			 REG_PMR1_SB_BYPASS_OFFSET, 1, NULL, 0),
+
+	SND_SOC_DAPM_ADC("ADC", "Capture", JZ4760_CODEC_REG_PMR2,
+			 REG_PMR2_SB_ADC_OFFSET, 1),
+
+	SND_SOC_DAPM_DAC("DAC", "Playback", JZ4760_CODEC_REG_PMR2,
+			 REG_PMR2_SB_DAC_OFFSET, 1),
+
+	SND_SOC_DAPM_MIXER("PCM Playback", SND_SOC_NOPM, 0, 0,
+			   jz4760_codec_pcm_playback_controls,
+			   ARRAY_SIZE(jz4760_codec_pcm_playback_controls)),
+
+	SND_SOC_DAPM_MIXER("Headphones Playback", SND_SOC_NOPM, 0, 0,
+			   jz4760_codec_hp_playback_controls,
+			   ARRAY_SIZE(jz4760_codec_hp_playback_controls)),
+
+	SND_SOC_DAPM_SUPPLY("MICBIAS", JZ4760_CODEC_REG_PMR1,
+			    REG_PMR1_SB_MICBIAS_OFFSET, 1, NULL, 0),
+
+	SND_SOC_DAPM_INPUT("MIC1P"),
+	SND_SOC_DAPM_INPUT("MIC1N"),
+	SND_SOC_DAPM_INPUT("MIC2P"),
+	SND_SOC_DAPM_INPUT("MIC2N"),
+
+	SND_SOC_DAPM_INPUT("LLINEIN"),
+	SND_SOC_DAPM_INPUT("RLINEIN"),
+
+	SND_SOC_DAPM_OUTPUT("LHPOUT"),
+	SND_SOC_DAPM_OUTPUT("RHPOUT"),
+
+	SND_SOC_DAPM_OUTPUT("LOUT"),
+	SND_SOC_DAPM_OUTPUT("ROUT"),
+
+	SND_SOC_DAPM_OUTPUT("BTLP"),
+	SND_SOC_DAPM_OUTPUT("BTLN"),
+
+	SND_SOC_DAPM_OUTPUT("SYSCLK"),
+};
+
+/* Unconditional routes. */
+static const struct snd_soc_dapm_route jz4760_codec_dapm_routes[] = {
+	{ "Mic 1", NULL, "MIC1P" },
+	{ "Mic Diff", NULL, "MIC1N" },
+	{ "Mic 1", NULL, "Mic Diff" },
+	{ "Mic 2", NULL, "MIC2P" },
+	{ "Mic Diff", NULL, "MIC2N" },
+	{ "Mic 2", NULL, "Mic Diff" },
+
+	{ "Line In", NULL, "LLINEIN" },
+	{ "Line In", NULL, "RLINEIN" },
+
+	{ "Mic", "Stereo Capture Switch", "Mic 1" },
+	{ "Mic", "Stereo Capture Switch", "Mic 2" },
+	{ "Headphones Source", "Mic 1", "Mic" },
+	{ "Headphones Source", "Mic 2", "Mic" },
+	{ "Capture Source", "Mic 1", "Mic" },
+	{ "Capture Source", "Mic 2", "Mic" },
+
+	{ "Capture Source", "Line In", "Line In" },
+	{ "Capture Source", "Mic 1", "Mic 1" },
+	{ "Capture Source", "Mic 2", "Mic 2" },
+	{ "ADC", NULL, "Capture Source" },
+
+	{ "Line In Bypass", NULL, "Line In" },
+
+	{ "Headphones Source", "Mic 1", "Mic 1" },
+	{ "Headphones Source", "Mic 2", "Mic 2" },
+	{ "Headphones Source", "Line In", "Line In Bypass" },
+	{ "Headphones Source", "PCM", "Headphones Playback" },
+	{ "HP Out", NULL, "Headphones Source" },
+
+	{ "LHPOUT", NULL, "HP Out" },
+	{ "RHPOUT", NULL, "HP Out" },
+	{ "Line Out", "Switch", "HP Out" },
+
+	{ "LOUT", NULL, "Line Out" },
+	{ "ROUT", NULL, "Line Out" },
+	{ "BTL Out", "Switch", "Line Out" },
+
+	{ "BTLP", NULL, "BTL Out"},
+	{ "BTLN", NULL, "BTL Out"},
+
+	{ "PCM Playback", "Volume", "DAC" },
+	{ "Headphones Playback", "Volume", "PCM Playback" },
+
+	{ "SYSCLK", NULL, "DAC" },
+};
+
+static void jz4760_codec_codec_init_regs(struct snd_soc_component *codec)
+{
+	struct jz_codec *jz_codec = snd_soc_component_get_drvdata(codec);
+	struct regmap *regmap = jz_codec->regmap;
+
+	/* Collect updates for later sending. */
+	regcache_cache_only(regmap, true);
+
+	/* default Amp output to PCM */
+	regmap_set_bits(regmap, JZ4760_CODEC_REG_CR1, REG_CR1_OUTSEL_MASK);
+
+	/* Disable stereo mic */
+	regmap_clear_bits(regmap, JZ4760_CODEC_REG_CR3,
+			  BIT(REG_CR3_MICSTEREO_OFFSET));
+
+	/* Set mic 1 as default source for ADC */
+	regmap_clear_bits(regmap, JZ4760_CODEC_REG_CR3,
+			  REG_CR3_ADC_INSEL_MASK);
+
+	/* ADC/DAC: serial + i2s */
+	regmap_set_bits(regmap, JZ4760_CODEC_REG_AICR,
+			REG_AICR_ADC_SERIAL | REG_AICR_ADC_I2S |
+			REG_AICR_DAC_SERIAL | REG_AICR_DAC_I2S);
+
+	/* The generated IRQ is a high level */
+	regmap_clear_bits(regmap, JZ4760_CODEC_REG_ICR, REG_ICR_INT_FORM_MASK);
+	regmap_update_bits(regmap, JZ4760_CODEC_REG_ICR, REG_ICR_ALL_MASK,
+			   REG_ICR_JACK_MASK | REG_ICR_RUP_MASK |
+			   REG_ICR_RDO_MASK  | REG_ICR_GUP_MASK |
+			   REG_ICR_GDO_MASK);
+
+	/* 12M oscillator */
+	regmap_clear_bits(regmap, JZ4760_CODEC_REG_CCR1, REG_CCR1_CRYSTAL_MASK);
+
+	/* 0: 16ohm/220uF, 1: 10kohm/1uF */
+	regmap_clear_bits(regmap, JZ4760_CODEC_REG_CR1, REG_CR1_HP_LOAD);
+
+	/* default to NOMAD */
+	regmap_set_bits(jz_codec->regmap, JZ4760_CODEC_REG_CR2,
+			REG_CR2_DAC_NOMAD);
+
+	/* disable automatic gain */
+	regmap_clear_bits(regmap, JZ4760_CODEC_REG_AGC1, REG_AGC1_EN);
+
+	/* Independent L/R DAC gain control */
+	regmap_clear_bits(regmap, JZ4760_CODEC_REG_GCR5,
+			  REG_GCR_RL);
+
+	/* Send collected updates. */
+	regcache_cache_only(regmap, false);
+	regcache_sync(regmap);
+}
+
+static int jz4760_codec_codec_probe(struct snd_soc_component *codec)
+{
+	struct jz_codec *jz_codec = snd_soc_component_get_drvdata(codec);
+
+	clk_prepare_enable(jz_codec->clk);
+
+	jz4760_codec_codec_init_regs(codec);
+
+	return 0;
+}
+
+static void jz4760_codec_codec_remove(struct snd_soc_component *codec)
+{
+	struct jz_codec *jz_codec = snd_soc_component_get_drvdata(codec);
+
+	clk_disable_unprepare(jz_codec->clk);
+}
+
+static const struct snd_soc_component_driver jz4760_codec_soc_codec_dev = {
+	.probe			= jz4760_codec_codec_probe,
+	.remove			= jz4760_codec_codec_remove,
+	.set_bias_level		= jz4760_codec_set_bias_level,
+	.controls		= jz4760_codec_snd_controls,
+	.num_controls		= ARRAY_SIZE(jz4760_codec_snd_controls),
+	.dapm_widgets		= jz4760_codec_dapm_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(jz4760_codec_dapm_widgets),
+	.dapm_routes		= jz4760_codec_dapm_routes,
+	.num_dapm_routes	= ARRAY_SIZE(jz4760_codec_dapm_routes),
+	.suspend_bias_off	= 1,
+	.use_pmdown_time	= 1,
+};
+
+static const unsigned int jz4760_codec_sample_rates[] = {
+	96000, 48000, 44100, 32000,
+	24000, 22050, 16000, 12000,
+	11025, 9600, 8000,
+};
+
+static int jz4760_codec_hw_params(struct snd_pcm_substream *substream,
+				  struct snd_pcm_hw_params *params,
+				  struct snd_soc_dai *dai)
+{
+	struct jz_codec *codec = snd_soc_component_get_drvdata(dai->component);
+	unsigned int rate, bit_width;
+
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		bit_width = 0;
+		break;
+	case SNDRV_PCM_FORMAT_S18_3LE:
+		bit_width = 1;
+		break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		bit_width = 2;
+		break;
+	case SNDRV_PCM_FORMAT_S24_3LE:
+		bit_width = 3;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	for (rate = 0; rate < ARRAY_SIZE(jz4760_codec_sample_rates); rate++) {
+		if (jz4760_codec_sample_rates[rate] == params_rate(params))
+			break;
+	}
+
+	if (rate == ARRAY_SIZE(jz4760_codec_sample_rates))
+		return -EINVAL;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		regmap_update_bits(codec->regmap, JZ4760_CODEC_REG_AICR,
+				   REG_AICR_DAC_ADWL_MASK,
+				   FIELD_PREP(REG_AICR_DAC_ADWL_MASK, bit_width));
+		regmap_update_bits(codec->regmap, JZ4760_CODEC_REG_CCR2,
+				   REG_CCR2_DAC_FREQ_MASK,
+				   FIELD_PREP(REG_CCR2_DAC_FREQ_MASK, rate));
+	} else {
+		regmap_update_bits(codec->regmap, JZ4760_CODEC_REG_AICR,
+				   REG_AICR_ADC_ADWL_MASK,
+				   FIELD_PREP(REG_AICR_ADC_ADWL_MASK, bit_width));
+		regmap_update_bits(codec->regmap, JZ4760_CODEC_REG_CCR2,
+				   REG_CCR2_ADC_FREQ_MASK,
+				   FIELD_PREP(REG_CCR2_ADC_FREQ_MASK, rate));
+	}
+
+	return 0;
+}
+
+static const struct snd_soc_dai_ops jz4760_codec_dai_ops = {
+	.startup	= jz4760_codec_startup,
+	.shutdown	= jz4760_codec_shutdown,
+	.hw_params	= jz4760_codec_hw_params,
+	.trigger	= jz4760_codec_pcm_trigger,
+	.mute_stream	= jz4760_codec_mute_stream,
+	.no_capture_mute = 1,
+};
+
+#define JZ_CODEC_FORMATS (SNDRV_PCM_FMTBIT_S16_LE  | \
+			  SNDRV_PCM_FMTBIT_S18_3LE | \
+			  SNDRV_PCM_FMTBIT_S20_3LE | \
+			  SNDRV_PCM_FMTBIT_S24_3LE)
+
+static struct snd_soc_dai_driver jz4760_codec_dai = {
+	.name = "jz4760-hifi",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_8000_96000,
+		.formats = JZ_CODEC_FORMATS,
+	},
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_8000_96000,
+		.formats = JZ_CODEC_FORMATS,
+	},
+	.ops = &jz4760_codec_dai_ops,
+};
+
+static bool jz4760_codec_volatile(struct device *dev, unsigned int reg)
+{
+	return reg == JZ4760_CODEC_REG_SR || reg == JZ4760_CODEC_REG_IFR;
+}
+
+static bool jz4760_codec_writeable(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case JZ4760_CODEC_REG_SR:
+		return false;
+	default:
+		return true;
+	}
+}
+
+static int jz4760_codec_io_wait(struct jz_codec *codec)
+{
+	u32 reg;
+
+	return readl_poll_timeout(codec->base + ICDC_RGADW_OFFSET, reg,
+				  !(reg & ICDC_RGADW_RGWR),
+				  1000, 1 * USEC_PER_SEC);
+}
+
+static int jz4760_codec_reg_read(void *context, unsigned int reg,
+				 unsigned int *val)
+{
+	struct jz_codec *codec = context;
+	unsigned int i;
+	u32 tmp;
+	int ret;
+
+	ret = jz4760_codec_io_wait(codec);
+	if (ret)
+		return ret;
+
+	tmp = readl(codec->base + ICDC_RGADW_OFFSET);
+	tmp &= ~ICDC_RGADW_RGADDR_MASK;
+	tmp |= FIELD_PREP(ICDC_RGADW_RGADDR_MASK, reg);
+	writel(tmp, codec->base + ICDC_RGADW_OFFSET);
+
+	/* wait 6+ cycles */
+	for (i = 0; i < 6; i++)
+		*val = readl(codec->base + ICDC_RGDATA_OFFSET) &
+			ICDC_RGDATA_RGDOUT_MASK;
+
+	return 0;
+}
+
+static int jz4760_codec_reg_write(void *context, unsigned int reg,
+				  unsigned int val)
+{
+	struct jz_codec *codec = context;
+	int ret;
+
+	ret = jz4760_codec_io_wait(codec);
+	if (ret)
+		return ret;
+
+	writel(ICDC_RGADW_RGWR | FIELD_PREP(ICDC_RGADW_RGADDR_MASK, reg) | val,
+	       codec->base + ICDC_RGADW_OFFSET);
+
+	ret = jz4760_codec_io_wait(codec);
+	if (ret)
+		return ret;
+
+	return 0;
+}
+
+static const u8 jz4760_codec_reg_defaults[] = {
+	0x00, 0xFC, 0x1B, 0x20, 0x00, 0x80, 0x00, 0x00,
+	0xFF, 0x1F, 0x3F, 0x00, 0x06, 0x06, 0x06, 0x06,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x34, 0x07, 0x44,
+	0x1F, 0x00, 0x00, 0x00
+};
+
+static struct regmap_config jz4760_codec_regmap_config = {
+	.reg_bits = 7,
+	.val_bits = 8,
+
+	.max_register = JZ4760_CODEC_REG_MIX2,
+	.volatile_reg = jz4760_codec_volatile,
+	.writeable_reg = jz4760_codec_writeable,
+
+	.reg_read = jz4760_codec_reg_read,
+	.reg_write = jz4760_codec_reg_write,
+
+	.reg_defaults_raw = jz4760_codec_reg_defaults,
+	.num_reg_defaults_raw = ARRAY_SIZE(jz4760_codec_reg_defaults),
+	.cache_type = REGCACHE_FLAT,
+};
+
+static int jz4760_codec_probe(struct platform_device *pdev)
+{
+	struct device *dev = &pdev->dev;
+	struct jz_codec *codec;
+	int ret;
+
+	codec = devm_kzalloc(dev, sizeof(*codec), GFP_KERNEL);
+	if (!codec)
+		return -ENOMEM;
+
+	codec->dev = dev;
+
+	codec->base = devm_platform_ioremap_resource(pdev, 0);
+	if (IS_ERR(codec->base)) {
+		ret = PTR_ERR(codec->base);
+		dev_err(dev, "Failed to ioremap mmio memory: %d\n", ret);
+		return ret;
+	}
+
+	codec->regmap = devm_regmap_init(dev, NULL, codec,
+					&jz4760_codec_regmap_config);
+	if (IS_ERR(codec->regmap))
+		return PTR_ERR(codec->regmap);
+
+	codec->clk = devm_clk_get(dev, "aic");
+	if (IS_ERR(codec->clk))
+		return PTR_ERR(codec->clk);
+
+	platform_set_drvdata(pdev, codec);
+
+	ret = devm_snd_soc_register_component(dev, &jz4760_codec_soc_codec_dev,
+					      &jz4760_codec_dai, 1);
+	if (ret) {
+		dev_err(dev, "Failed to register codec: %d\n", ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static const struct of_device_id jz4760_codec_of_matches[] = {
+	{ .compatible = "ingenic,jz4760-codec", },
+	{ /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(of, jz4760_codec_of_matches);
+
+static struct platform_driver jz4760_codec_driver = {
+	.probe			= jz4760_codec_probe,
+	.driver			= {
+		.name		= "jz4760-codec",
+		.of_match_table = jz4760_codec_of_matches,
+	},
+};
+module_platform_driver(jz4760_codec_driver);
+
+MODULE_DESCRIPTION("JZ4760 SoC internal codec driver");
+MODULE_AUTHOR("Christophe Branchereau <cbranchereau@gmail.com>");
+MODULE_AUTHOR("Paul Cercueil <paul@crapouillou.net>");
+MODULE_LICENSE("GPL v2");

sound/soc/codecs/lm49453.c

@@ -1343,7 +1343,7 @@ static struct snd_soc_dai_driver lm49453_dai[] = {
 			.formats = LM49453_FORMATS,
 		},
 		.ops = &lm49453_headset_dai_ops,
-		.symmetric_rates = 1,
+		.symmetric_rate = 1,
 	},
 	{
 		.name = "LM49453 Speaker",

sound/soc/codecs/lochnagar-sc.c

@@ -166,8 +166,8 @@ static struct snd_soc_dai_driver lochnagar_sc_dai[] = {
 			.formats = SNDRV_PCM_FMTBIT_S32_LE,
 		},
 		.ops = &lochnagar_sc_line_ops,
-		.symmetric_rates = true,
-		.symmetric_samplebits = true,
+		.symmetric_rate = true,
+		.symmetric_sample_bits = true,
 	},
 	{
 		.name = "lochnagar-usb1",
@@ -186,8 +186,8 @@ static struct snd_soc_dai_driver lochnagar_sc_dai[] = {
 			.formats = SNDRV_PCM_FMTBIT_S32_LE,
 		},
 		.ops = &lochnagar_sc_usb_ops,
-		.symmetric_rates = true,
-		.symmetric_samplebits = true,
+		.symmetric_rate = true,
+		.symmetric_sample_bits = true,
 	},
 	{
 		.name = "lochnagar-usb2",
@@ -206,8 +206,8 @@ static struct snd_soc_dai_driver lochnagar_sc_dai[] = {
 			.formats = SNDRV_PCM_FMTBIT_S32_LE,
 		},
 		.ops = &lochnagar_sc_usb_ops,
-		.symmetric_rates = true,
-		.symmetric_samplebits = true,
+		.symmetric_rate = true,
+		.symmetric_sample_bits = true,
 	},
 };
 

sound/soc/codecs/lpass-wsa-macro.c

@@ -40,9 +40,11 @@
 #define CDC_WSA_TOP_I2S_CLK			(0x00A4)
 #define CDC_WSA_TOP_I2S_RESET			(0x00A8)
 #define CDC_WSA_RX_INP_MUX_RX_INT0_CFG0		(0x0100)
-#define CDC_WSA_RX_INTX_1_MIX_INP2_SEL_MASK	GENMASK(5, 3)
-#define CDC_WSA_RX_INTX_2_SEL_MASK		GENMASK(2, 0)
+#define CDC_WSA_RX_INTX_1_MIX_INP0_SEL_MASK	GENMASK(2, 0)
+#define CDC_WSA_RX_INTX_1_MIX_INP1_SEL_MASK	GENMASK(5, 3)
 #define CDC_WSA_RX_INP_MUX_RX_INT0_CFG1		(0x0104)
+#define CDC_WSA_RX_INTX_2_SEL_MASK		GENMASK(2, 0)
+#define CDC_WSA_RX_INTX_1_MIX_INP2_SEL_MASK	GENMASK(5, 3)
 #define CDC_WSA_RX_INP_MUX_RX_INT1_CFG0		(0x0108)
 #define CDC_WSA_RX_INP_MUX_RX_INT1_CFG1		(0x010C)
 #define CDC_WSA_RX_INP_MUX_RX_MIX_CFG0		(0x0110)
@@ -229,8 +231,6 @@
 #define NUM_INTERPOLATORS 2
 #define WSA_NUM_CLKS_MAX	5
 #define WSA_MACRO_MCLK_FREQ 19200000
-#define WSA_MACRO_MUX_INP_SHFT 0x3
-#define WSA_MACRO_MUX_INP_MASK1 0x07
 #define WSA_MACRO_MUX_INP_MASK2 0x38
 #define WSA_MACRO_MUX_CFG_OFFSET 0x8
 #define WSA_MACRO_MUX_CFG1_OFFSET 0x4
@@ -843,7 +843,6 @@ static int wsa_macro_set_prim_interpolator_rate(struct snd_soc_dai *dai,
 	u32 j, port;
 	u16 int_mux_cfg0, int_mux_cfg1;
 	u16 int_fs_reg;
-	u8 int_mux_cfg0_val, int_mux_cfg1_val;
 	u8 inp0_sel, inp1_sel, inp2_sel;
 	struct snd_soc_component *component = dai->component;
 	struct wsa_macro *wsa = snd_soc_component_get_drvdata(component);
@@ -865,15 +864,13 @@ static int wsa_macro_set_prim_interpolator_rate(struct snd_soc_dai *dai,
 		 */
 		for (j = 0; j < NUM_INTERPOLATORS; j++) {
 			int_mux_cfg1 = int_mux_cfg0 + WSA_MACRO_MUX_CFG1_OFFSET;
-			int_mux_cfg0_val = snd_soc_component_read(component,
-								  int_mux_cfg0);
-			int_mux_cfg1_val = snd_soc_component_read(component,
-								  int_mux_cfg1);
-			inp0_sel = int_mux_cfg0_val & WSA_MACRO_MUX_INP_MASK1;
-			inp1_sel = (int_mux_cfg0_val >> WSA_MACRO_MUX_INP_SHFT) &
-						WSA_MACRO_MUX_INP_MASK1;
-			inp2_sel = (int_mux_cfg1_val >> WSA_MACRO_MUX_INP_SHFT) &
-						WSA_MACRO_MUX_INP_MASK1;
+			inp0_sel = snd_soc_component_read_field(component, int_mux_cfg0, 
+								CDC_WSA_RX_INTX_1_MIX_INP0_SEL_MASK);
+			inp1_sel = snd_soc_component_read_field(component, int_mux_cfg0, 
+								CDC_WSA_RX_INTX_1_MIX_INP1_SEL_MASK);
+			inp2_sel = snd_soc_component_read_field(component, int_mux_cfg1,
+								CDC_WSA_RX_INTX_1_MIX_INP2_SEL_MASK);
+
 			if ((inp0_sel == int_1_mix1_inp + INTn_1_INP_SEL_RX0) ||
 			    (inp1_sel == int_1_mix1_inp + INTn_1_INP_SEL_RX0) ||
 			    (inp2_sel == int_1_mix1_inp + INTn_1_INP_SEL_RX0)) {
@@ -912,9 +909,9 @@ static int wsa_macro_set_mix_interpolator_rate(struct snd_soc_dai *dai,
 
 		int_mux_cfg1 = CDC_WSA_RX_INP_MUX_RX_INT0_CFG1;
 		for (j = 0; j < NUM_INTERPOLATORS; j++) {
-			int_mux_cfg1_val = snd_soc_component_read(component,
-							int_mux_cfg1) &
-							WSA_MACRO_MUX_INP_MASK1;
+			int_mux_cfg1_val = snd_soc_component_read_field(component, int_mux_cfg1,
+									CDC_WSA_RX_INTX_2_SEL_MASK);
+
 			if (int_mux_cfg1_val == int_2_inp + INTn_2_INP_SEL_RX0) {
 				int_fs_reg = CDC_WSA_RX0_RX_PATH_MIX_CTL +
 					WSA_MACRO_RX_PATH_OFFSET * j;
@@ -1410,25 +1407,25 @@ static bool wsa_macro_adie_lb(struct snd_soc_component *component,
 			      int interp_idx)
 {
 	u16 int_mux_cfg0,  int_mux_cfg1;
-	u8 int_mux_cfg0_val, int_mux_cfg1_val;
 	u8 int_n_inp0, int_n_inp1, int_n_inp2;
 
 	int_mux_cfg0 = CDC_WSA_RX_INP_MUX_RX_INT0_CFG0 + interp_idx * 8;
 	int_mux_cfg1 = int_mux_cfg0 + 4;
-	int_mux_cfg0_val = snd_soc_component_read(component, int_mux_cfg0);
-	int_mux_cfg1_val = snd_soc_component_read(component, int_mux_cfg1);
 
-	int_n_inp0 = int_mux_cfg0_val & 0x0F;
+	int_n_inp0 = snd_soc_component_read_field(component, int_mux_cfg0,
+						  CDC_WSA_RX_INTX_1_MIX_INP0_SEL_MASK);
 	if (int_n_inp0 == INTn_1_INP_SEL_DEC0 ||
 		int_n_inp0 == INTn_1_INP_SEL_DEC1)
 		return true;
 
-	int_n_inp1 = int_mux_cfg0_val >> 4;
+	int_n_inp1 = snd_soc_component_read_field(component, int_mux_cfg0,
+						  CDC_WSA_RX_INTX_1_MIX_INP1_SEL_MASK);
 	if (int_n_inp1 == INTn_1_INP_SEL_DEC0 ||
 		int_n_inp1 == INTn_1_INP_SEL_DEC1)
 		return true;
 
-	int_n_inp2 = int_mux_cfg1_val >> 4;
+	int_n_inp2 = snd_soc_component_read_field(component, int_mux_cfg1,
+						  CDC_WSA_RX_INTX_1_MIX_INP2_SEL_MASK);
 	if (int_n_inp2 == INTn_1_INP_SEL_DEC0 ||
 		int_n_inp2 == INTn_1_INP_SEL_DEC1)
 		return true;

sound/soc/codecs/max98373-sdw.c

@@ -262,6 +262,8 @@ static __maybe_unused int max98373_suspend(struct device *dev)
 	return 0;
 }
 
+#define MAX98373_PROBE_TIMEOUT 5000
+
 static __maybe_unused int max98373_resume(struct device *dev)
 {
 	struct sdw_slave *slave = dev_to_sdw_dev(dev);
@@ -275,7 +277,7 @@ static __maybe_unused int max98373_resume(struct device *dev)
 		goto regmap_sync;
 
 	time = wait_for_completion_timeout(&slave->initialization_complete,
-					   msecs_to_jiffies(2000));
+					   msecs_to_jiffies(MAX98373_PROBE_TIMEOUT));
 	if (!time) {
 		dev_err(dev, "Initialization not complete, timed out\n");
 		return -ETIMEDOUT;

sound/soc/codecs/max9860.c

@@ -489,7 +489,7 @@ static struct snd_soc_dai_driver max9860_dai = {
 			   SNDRV_PCM_FMTBIT_S32_LE,
 	},
 	.ops = &max9860_dai_ops,
-	.symmetric_rates = 1,
+	.symmetric_rate = 1,
 };
 
 static int max9860_set_bias_level(struct snd_soc_component *component,