Merge tag 'sound-5.7-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
"This became again a busy development cycle. There are few ALSA core
updates (merely API cleanups and sparse fixes), with the majority of
other changes are found in ASoC scene.

Here are some highlights:

ALSA core:
- More helper macros for sparse warning fixes (e.g. bitwise types)
- Slight optimization of PCM OSS locks
- Make common handling for PCM / compress buffers (for SOF)

ASoC:
- Lots of code refactoring and modernization for (still ongoing)
componentization works
- Conversion of SND_SOC_ALL_CODECS to use imply
- Continued refactoring and fixing of the Intel SOF/SST support,
including the initial (but still incomplete) SoundWire support
- SoundWire and more advanced clocking support for Realtek RT5682
- Support for amlogic GX, Meson 8, Meson 8B and T9015 DAC, Broadcom
DSL/PON, Ingenic JZ4760 and JZ4770, Realtek RL6231, and TI TAS2563
and TLV320ADCX140

HD-audio:
- Optimizations in HDMI jack handling
- A few new quirks and fixups for Realtek codecs

USB-audio:
- Delayed registration support
- New quirks for Motu, Kingston, Presonus"

* tag 'sound-5.7-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (415 commits)
ALSA: usb-audio: Fix case when USB MIDI interface has more than one extra endpoint descriptor
Revert "ALSA: uapi: Drop asound.h inclusion from asoc.h"
ALSA: hda/realtek - Remove now-unnecessary XPS 13 headphone noise fixups
ALSA: hda/realtek - Set principled PC Beep configuration for ALC256
ALSA: doc: Document PC Beep Hidden Register on Realtek ALC256
ALSA: hda/realtek - a fake key event is triggered by running shutup
ALSA: hda: default enable CA0132 DSP support
ASoC: amd: acp3x-pcm-dma: clean up two indentation issues
ASoC: tlv320adcx140: Remove undocumented property
ASoC: Intel: sof_sdw: Add Volteer support with RT5682 SNDW helper function
ASoC: Intel: common: add match table for TGL RT5682 SoundWire driver
ASoC: Intel: boards: add sof_sdw machine driver
ASoC: Intel: soc-acpi: update topology and driver name for SoundWire platforms
ASoC: rt5682: move DAI clock registry to I2S mode
ASoC: pxa: magician: convert to use i2c_new_client_device()
ASoC: SOF: Intel: hda-ctrl: add reset cycle before parsing capabilities
Asoc: SOF: Intel: hda: check SoundWire wakeen interrupt in irq thread
ASoC: SOF: Intel: hda: add WAKEEN interrupt support for SoundWire
ASoC: SOF: Intel: hda: add parameter to control SoundWire clock stop quirks
ASoC: SOF: Intel: hda: merge IPC, stream and SoundWire interrupt handlers
...

Documentation/devicetree/bindings/sound/amlogic,aiu.yaml

···
+# SPDX-License-Identifier: GPL-2.0
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/amlogic,aiu.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Amlogic AIU audio output controller
+
+maintainers:
+  - Jerome Brunet <[e-mail]>
+
+properties:
+  $nodename:
+    pattern: "^audio-controller@.*"
+
+  "#sound-dai-cells":
+    const: 2
+
+  compatible:
+    items:
+      - enum:
+        - amlogic,aiu-gxbb
+        - amlogic,aiu-gxl
+        - amlogic,aiu-meson8
+        - amlogic,aiu-meson8b
+      - const:
+          amlogic,aiu
+
+  clocks:
+    items:
+      - description: AIU peripheral clock
+      - description: I2S peripheral clock
+      - description: I2S output clock
+      - description: I2S master clock
+      - description: I2S mixer clock
+      - description: SPDIF peripheral clock
+      - description: SPDIF output clock
+      - description: SPDIF master clock
+      - description: SPDIF master clock multiplexer
+
+  clock-names:
+    items:
+      - const: pclk
+      - const: i2s_pclk
+      - const: i2s_aoclk
+      - const: i2s_mclk
+      - const: i2s_mixer
+      - const: spdif_pclk
+      - const: spdif_aoclk
+      - const: spdif_mclk
+      - const: spdif_mclk_sel
+
+  interrupts:
+    items:
+      - description: I2S interrupt line
+      - description: SPDIF interrupt line
+
+  interrupt-names:
+    items:
+      - const: i2s
+      - const: spdif
+
+  reg:
+    maxItems: 1
+
+  resets:
+    maxItems: 1
+
+required:
+  - "#sound-dai-cells"
+  - compatible
+  - clocks
+  - clock-names
+  - interrupts
+  - interrupt-names
+  - reg
+  - resets
+
+examples:
+  - |
+    #include <dt-bindings/clock/gxbb-clkc.h>
+    #include <dt-bindings/interrupt-controller/irq.h>
+    #include <dt-bindings/interrupt-controller/arm-gic.h>
+    #include <dt-bindings/reset/amlogic,meson-gxbb-reset.h>
+
+    aiu: audio-controller@5400 {
+        compatible = "amlogic,aiu-gxl", "amlogic,aiu";
+        #sound-dai-cells = <2>;
+        reg = <0x0 0x5400 0x0 0x2ac>;
+        interrupts = <GIC_SPI 48 IRQ_TYPE_EDGE_RISING>,
+                     <GIC_SPI 50 IRQ_TYPE_EDGE_RISING>;
+        interrupt-names = "i2s", "spdif";
+        clocks = <&clkc CLKID_AIU_GLUE>,
+                 <&clkc CLKID_I2S_OUT>,
+                 <&clkc CLKID_AOCLK_GATE>,
+                 <&clkc CLKID_CTS_AMCLK>,
+                 <&clkc CLKID_MIXER_IFACE>,
+                 <&clkc CLKID_IEC958>,
+                 <&clkc CLKID_IEC958_GATE>,
+                 <&clkc CLKID_CTS_MCLK_I958>,
+                 <&clkc CLKID_CTS_I958>;
+        clock-names = "pclk",
+                      "i2s_pclk",
+                      "i2s_aoclk",
+                      "i2s_mclk",
+                      "i2s_mixer",
+                      "spdif_pclk",
+                      "spdif_aoclk",
+                      "spdif_mclk",
+                      "spdif_mclk_sel";
+        resets = <&reset RESET_AIU>;
+    };
+

Documentation/devicetree/bindings/sound/amlogic,g12a-toacodec.yaml

···
+# SPDX-License-Identifier: GPL-2.0
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/amlogic,g12a-toacodec.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Amlogic G12a Internal DAC Control Glue
+
+maintainers:
+  - Jerome Brunet <[e-mail]>
+
+properties:
+  $nodename:
+    pattern: "^audio-controller@.*"
+
+  "#sound-dai-cells":
+    const: 1
+
+  compatible:
+    oneOf:
+      - items:
+        - const:
+            amlogic,g12a-toacodec
+      - items:
+        - enum:
+          - amlogic,sm1-toacodec
+        - const:
+            amlogic,g12a-toacodec
+
+  reg:
+    maxItems: 1
+
+  resets:
+    maxItems: 1
+
+required:
+  - "#sound-dai-cells"
+  - compatible
+  - reg
+  - resets
+
+examples:
+  - |
+    #include <dt-bindings/reset/amlogic,meson-g12a-audio-reset.h>
+
+    toacodec: audio-controller@740 {
+        compatible = "amlogic,g12a-toacodec";
+        reg = <0x0 0x740 0x0 0x4>;
+        #sound-dai-cells = <1>;
+        resets = <&clkc_audio AUD_RESET_TOACODEC>;
+    };

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

···
+# SPDX-License-Identifier: GPL-2.0
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/amlogic,gx-sound-card.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Amlogic GX sound card
+
+maintainers:
+  - Jerome Brunet <[e-mail]>
+
+properties:
+  compatible:
+    items:
+      - const: amlogic,gx-sound-card
+
+  audio-aux-devs:
+    $ref: /schemas/types.yaml#/definitions/phandle-array
+    description: list of auxiliary devices
+
+  audio-routing:
+    $ref: /schemas/types.yaml#/definitions/non-unique-string-array
+    minItems: 2
+    description: |-
+      A list of the connections between audio components. Each entry is a
+      pair of strings, the first being the connection's sink, the second
+      being the connection's source.
+
+  audio-widgets:
+    $ref: /schemas/types.yaml#/definitions/non-unique-string-array
+    minItems: 2
+    description: |-
+      A list off component DAPM widget. Each entry is a pair of strings,
+      the first being the widget type, the second being the widget name
+
+  model:
+    $ref: /schemas/types.yaml#/definitions/string
+    description: User specified audio sound card name
+
+patternProperties:
+  "^dai-link-[0-9]+$":
+    type: object
+    description: |-
+      dai-link child nodes:
+        Container for dai-link level properties and the CODEC sub-nodes.
+        There should be at least one (and probably more) subnode of this type
+
+    properties:
+      dai-format:
+        $ref: /schemas/types.yaml#/definitions/string
+        enum: [ i2s, left-j, dsp_a ]
+
+      mclk-fs:
+        $ref: /schemas/types.yaml#/definitions/uint32
+        description: |-
+          Multiplication factor between the frame rate and master clock
+          rate
+
+      sound-dai:
+        $ref: /schemas/types.yaml#/definitions/phandle
+        description: phandle of the CPU DAI
+
+    patternProperties:
+      "^codec-[0-9]+$":
+        type: object
+        description: |-
+          Codecs:
+          dai-link representing backend links should have at least one subnode.
+          One subnode for each codec of the dai-link. dai-link representing
+          frontend links have no codec, therefore have no subnodes
+
+        properties:
+          sound-dai:
+            $ref: /schemas/types.yaml#/definitions/phandle
+            description: phandle of the codec DAI
+
+        required:
+          - sound-dai
+
+    required:
+      - sound-dai
+
+required:
+  - model
+  - dai-link-0
+
+examples:
+  - |
+    sound {
+        compatible = "amlogic,gx-sound-card";
+        model = "GXL-ACME-S905X-FOO";
+        audio-aux-devs = <&amp>;
+        audio-routing = "I2S ENCODER I2S IN", "I2S FIFO Playback";
+
+        dai-link-0 {
+               sound-dai = <&i2s_fifo>;
+        };
+
+        dai-link-1 {
+                sound-dai = <&i2s_encoder>;
+                dai-format = "i2s";
+                mclk-fs = <256>;
+
+                codec-0 {
+                        sound-dai = <&codec0>;
+                };
+
+                codec-1 {
+                        sound-dai = <&codec1>;
+                };
+        };
+    };
+

Documentation/devicetree/bindings/sound/amlogic,t9015.yaml

···
+# SPDX-License-Identifier: GPL-2.0
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/amlogic,t9015.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Amlogic T9015 Internal Audio DAC
+
+maintainers:
+  - Jerome Brunet <[e-mail]>
+
+properties:
+  $nodename:
+    pattern: "^audio-controller@.*"
+
+  "#sound-dai-cells":
+    const: 0
+
+  compatible:
+    items:
+      - const: amlogic,t9015
+
+  clocks:
+    items:
+      - description: Peripheral clock
+
+  clock-names:
+    items:
+      - const: pclk
+
+  reg:
+    maxItems: 1
+
+  resets:
+    maxItems: 1
+
+required:
+  - "#sound-dai-cells"
+  - compatible
+  - reg
+  - clocks
+  - clock-names
+  - resets
+
+examples:
+  - |
+    #include <dt-bindings/clock/g12a-clkc.h>
+    #include <dt-bindings/reset/amlogic,meson-g12a-reset.h>
+
+    acodec: audio-controller@32000 {
+        compatible = "amlogic,t9015";
+        reg = <0x0 0x32000 0x0 0x14>;
+        #sound-dai-cells = <0>;
+        clocks = <&clkc CLKID_AUDIO_CODEC>;
+        clock-names = "pclk";
+        resets = <&reset RESET_AUDIO_CODEC>;
+    };
+

Documentation/devicetree/bindings/sound/brcm,bcm63xx-audio.txt

···
+Broadcom DSL/PON BCM63xx Audio I2S controller
+
+Required properties:
+- compatible:     Should be "brcm,bcm63xx-i2s".
+- #address-cells: 32bit valued, 1 cell.
+- #size-cells:    32bit valued, 0 cell.
+- reg:            Should contain audio registers location and length
+- interrupts:     Should contain the interrupt for the controller.
+- clocks:         Must contain an entry for each entry in clock-names.
+                  Please refer to clock-bindings.txt.
+- clock-names:    One of each entry matching the clocks phandles list:
+                  - "i2sclk" (generated clock) Required.
+                  - "i2sosc" (fixed 200MHz clock) Required.
+
+(1) : The generated clock is required only when any of TX and RX
+      works on Master Mode.
+(2) : The fixed 200MHz clock is from internal chip and always on
+
+Example:
+
+		i2s: bcm63xx-i2s {
+			#address-cells = <1>;
+			#size-cells = <0>;
+			compatible = "brcm,bcm63xx-i2s";
+			reg = <0xFF802080 0xFF>;
+			interrupts = <GIC_SPI 84 IRQ_TYPE_LEVEL_HIGH>;
+			clocks = <&i2sclk>, <&osc>;
+			clock-names = "i2sclk","i2sosc";
+		};

Documentation/devicetree/bindings/sound/cirrus,cs42l51.yaml

···
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/cirrus,cs42l51.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: CS42L51 audio codec DT bindings
+
+maintainers:
+  - Olivier Moysan <[e-mail]>
+
+properties:
+  compatible:
+      const: cirrus,cs42l51
+
+  reg:
+    maxItems: 1
+
+  "#sound-dai-cells":
+    const: 0
+
+  clocks:
+    maxItems: 1
+
+  clock-names:
+    items:
+      - const: MCLK
+
+  reset-gpios:
+    maxItems: 1
+
+  VL-supply:
+    description: phandle to voltage regulator of digital interface section
+
+  VD-supply:
+    description: phandle to voltage regulator of digital internal section
+
+  VA-supply:
+    description: phandle to voltage regulator of analog internal section
+
+  VAHP-supply:
+    description: phandle to voltage regulator of headphone
+
+required:
+  - compatible
+  - reg
+  - "#sound-dai-cells"
+
+examples:
+  - |
+    #include <dt-bindings/gpio/gpio.h>
+    i2c@0 {
+      #address-cells = <1>;
+      #size-cells = <0>;
+
+      cs42l51@4a {
+        compatible = "cirrus,cs42l51";
+        reg = <0x4a>;
+        #sound-dai-cells = <0>;
+        clocks = <&mclk_prov>;
+        clock-names = "MCLK";
+        VL-supply = <&reg_audio>;
+        VD-supply = <&reg_audio>;
+        VA-supply = <&reg_audio>;
+        VAHP-supply = <&reg_audio>;
+        reset-gpios = <&gpiog 9 GPIO_ACTIVE_LOW>;
+      };
+    };
+...

Documentation/devicetree/bindings/sound/cs42l51.txt

···
-CS42L51 audio CODEC
-
-Required properties:
-
-  - compatible : "cirrus,cs42l51"
-
-  - reg : the I2C address of the device for I2C.
-
-Optional properties:
-  - VL-supply, VD-supply, VA-supply, VAHP-supply: power supplies for the device,
-    as covered in Documentation/devicetree/bindings/regulator/regulator.txt.
-
-  - reset-gpios : GPIO specification for the reset pin. If specified, it will be
-    deasserted before starting the communication with the codec.
-
-  - clocks : a list of phandles + clock-specifiers, one for each entry in
-    clock-names
-
-  - clock-names : must contain "MCLK"
-
-Example:
-
-cs42l51: cs42l51@4a {
-	compatible = "cirrus,cs42l51";
-	reg = <0x4a>;
-	clocks = <&mclk_prov>;
-	clock-names = "MCLK";
-	VL-supply = <&reg_audio>;
-	VD-supply = <&reg_audio>;
-	VA-supply = <&reg_audio>;
-	VAHP-supply = <&reg_audio>;
-	reset-gpios = <&gpiog 9 GPIO_ACTIVE_LOW>;
-};

Documentation/devicetree/bindings/sound/google,cros-ec-codec.txt

···
-Audio codec controlled by ChromeOS EC
-
-Google's ChromeOS EC codec is a digital mic codec provided by the
-Embedded Controller (EC) and is controlled via a host-command interface.
-
-An EC codec node should only be found as a sub-node of the EC node (see
-Documentation/devicetree/bindings/mfd/cros-ec.txt).
-
-Required properties:
-- compatible: Must contain "google,cros-ec-codec"
-- #sound-dai-cells: Should be 1. The cell specifies number of DAIs.
-
-Optional properties:
-- reg: Pysical base address and length of shared memory region from EC.
-       It contains 3 unsigned 32-bit integer.  The first 2 integers
-       combine to become an unsigned 64-bit physical address.  The last
-       one integer is length of the shared memory.
-- memory-region: Shared memory region to EC.  A "shared-dma-pool".  See
-                 ../reserved-memory/reserved-memory.txt for details.
-
-Example:
-
-{
-	...
-
-	reserved_mem: reserved_mem {
-		compatible = "shared-dma-pool";
-		reg = <0 0x52800000 0 0x100000>;
-		no-map;
-	};
-}
-
-cros-ec@0 {
-	compatible = "google,cros-ec-spi";
-
-	...
-
-	cros_ec_codec: ec-codec {
-		compatible = "google,cros-ec-codec";
-		#sound-dai-cells = <1>;
-		reg = <0x0 0x10500000 0x80000>;
-		memory-region = <&reserved_mem>;
-	};
-};

Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml

···
+# SPDX-License-Identifier: GPL-2.0-only
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/google,cros-ec-codec.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Audio codec controlled by ChromeOS EC
+
+maintainers:
+  - Cheng-Yi Chiang <[e-mail]>
+
+description: |
+  Google's ChromeOS EC codec is a digital mic codec provided by the
+  Embedded Controller (EC) and is controlled via a host-command interface.
+  An EC codec node should only be found as a sub-node of the EC node (see
+  Documentation/devicetree/bindings/mfd/cros-ec.txt).
+
+properties:
+  compatible:
+    const: google,cros-ec-codec
+
+  "#sound-dai-cells":
+    const: 1
+
+  reg:
+    items:
+      - description: |
+          Physical base address and length of shared memory region from EC.
+          It contains 3 unsigned 32-bit integer. The first 2 integers
+          combine to become an unsigned 64-bit physical address.
+          The last one integer is the length of the shared memory.
+
+  memory-region:
+    $ref: '/schemas/types.yaml#/definitions/phandle'
+    description: |
+      Shared memory region to EC.  A "shared-dma-pool".
+      See ../reserved-memory/reserved-memory.txt for details.
+
+required:
+  - compatible
+  - '#sound-dai-cells'
+
+additionalProperties: false
+
+examples:
+  - |
+    reserved_mem: reserved-mem@52800000 {
+        compatible = "shared-dma-pool";
+        reg = <0x52800000 0x100000>;
+        no-map;
+    };
+    spi {
+        #address-cells = <1>;
+        #size-cells = <0>;
+        cros-ec@0 {
+            compatible = "google,cros-ec-spi";
+            #address-cells = <2>;
+            #size-cells = <1>;
+            reg = <0>;
+            cros_ec_codec: ec-codec@10500000 {
+                compatible = "google,cros-ec-codec";
+                #sound-dai-cells = <1>;
+                reg = <0x0 0x10500000 0x80000>;
+                memory-region = <&reserved_mem>;
+            };
+        };
+    };

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

···
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/ingenic,aic.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Ingenic SoCs AC97 / I2S Controller (AIC) DT bindings
+
+maintainers:
+  - Paul Cercueil <[e-mail]>
+
+properties:
+  $nodename:
+    pattern: '^audio-controller@'
+
+  compatible:
+    oneOf:
+      - enum:
+        - ingenic,jz4740-i2s
+        - ingenic,jz4760-i2s
+        - ingenic,jz4770-i2s
+        - ingenic,jz4780-i2s
+      - items:
+        - const: ingenic,jz4725b-i2s
+        - const: ingenic,jz4740-i2s
+
+  '#sound-dai-cells':
+    const: 0
+
+  reg:
+    maxItems: 1
+
+  interrupts:
+    maxItems: 1
+
+  clocks:
+    items:
+      - description: AIC clock
+      - description: I2S clock
+      - description: EXT clock
+      - description: PLL/2 clock
+
+  clock-names:
+    items:
+      - const: aic
+      - const: i2s
+      - const: ext
+      - const: pll half
+
+  dmas:
+    items:
+      - description: DMA controller phandle and request line for I2S RX
+      - description: DMA controller phandle and request line for I2S TX
+
+  dma-names:
+    items:
+      - const: rx
+      - const: tx
+
+additionalProperties: false
+
+required:
+  - compatible
+  - reg
+  - interrupts
+  - clocks
+  - clock-names
+  - dmas
+  - dma-names
+  - '#sound-dai-cells'
+
+examples:
+  - |
+    #include <dt-bindings/clock/jz4740-cgu.h>
+    aic: audio-controller@10020000 {
+      compatible = "ingenic,jz4740-i2s";
+      reg = <0x10020000 0x38>;
+
+      #sound-dai-cells = <0>;
+
+      interrupt-parent = <&intc>;
+      interrupts = <18>;
+
+      clocks = <&cgu JZ4740_CLK_AIC>,
+               <&cgu JZ4740_CLK_I2S>,
+               <&cgu JZ4740_CLK_EXT>,
+               <&cgu JZ4740_CLK_PLL_HALF>;
+      clock-names = "aic", "i2s", "ext", "pll half";
+
+      dmas = <&dmac 25 0xffffffff>, <&dmac 24 0xffffffff>;
+      dma-names = "rx", "tx";
+    };

Documentation/devicetree/bindings/sound/ingenic,jz4740-i2s.txt

···
-Ingenic JZ4740 I2S controller
-
-Required properties:
-- compatible : "ingenic,jz4740-i2s" or "ingenic,jz4780-i2s"
-- reg : I2S registers location and length
-- clocks : AIC and I2S PLL clock specifiers.
-- clock-names: "aic" and "i2s"
-- dmas: DMA controller phandle and DMA request line for I2S Tx and Rx channels
-- dma-names: Must be "tx" and "rx"
-
-Example:
-
-i2s: i2s@10020000 {
-	compatible = "ingenic,jz4740-i2s";
-	reg = <0x10020000 0x94>;
-
-	clocks = <&cgu JZ4740_CLK_AIC>, <&cgu JZ4740_CLK_I2SPLL>;
-	clock-names = "aic", "i2s";
-
-	dmas = <&dma 2>, <&dma 3>;
-	dma-names = "tx", "rx";
-
-};

Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt

···
   * Headphone Jack
   * Int Spk
   * Mic Jack
+  * Int Mic
 
 - nvidia,i2s-controller : The phandle of the Tegra I2S1 controller
 - nvidia,audio-codec : The phandle of the WM8903 audio codec

Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt

···
 - clock-names: should be "pclk".
 - spk-depop-time-ms: speak depop time msec.
 
+Optional properties:
+
+- mute-gpios: GPIO specifier for external line driver control (typically the
+              dedicated GPIO_MUTE pin)
+
 Example for rk3328 internal codec:
 
 codec: codec@ff410000 {
···
 	rockchip,grf = <&grf>;
 	clocks = <&cru PCLK_ACODEC>;
 	clock-names = "pclk";
+	mute-gpios = <&grf_gpio 0 GPIO_ACTIVE_LOW>;
 	spk-depop-time-ms = 100;
-	status = "disabled";
 };

Documentation/devicetree/bindings/sound/rockchip-i2s.txt

···
-* Rockchip I2S controller
-
-The I2S bus (Inter-IC sound bus) is a serial link for digital
-audio data transfer between devices in the system.
-
-Required properties:
-
-- compatible: should be one of the following:
-   - "rockchip,rk3066-i2s": for rk3066
-   - "rockchip,px30-i2s", "rockchip,rk3066-i2s": for px30
-   - "rockchip,rk3036-i2s", "rockchip,rk3066-i2s": for rk3036
-   - "rockchip,rk3188-i2s", "rockchip,rk3066-i2s": for rk3188
-   - "rockchip,rk3228-i2s", "rockchip,rk3066-i2s": for rk3228
-   - "rockchip,rk3288-i2s", "rockchip,rk3066-i2s": for rk3288
-   - "rockchip,rk3328-i2s", "rockchip,rk3066-i2s": for rk3328
-   - "rockchip,rk3366-i2s", "rockchip,rk3066-i2s": for rk3366
-   - "rockchip,rk3368-i2s", "rockchip,rk3066-i2s": for rk3368
-   - "rockchip,rk3399-i2s", "rockchip,rk3066-i2s": for rk3399
-- reg: physical base address of the controller and length of memory mapped
-  region.
-- interrupts: should contain the I2S interrupt.
-- dmas: DMA specifiers for tx and rx dma. See the DMA client binding,
-	Documentation/devicetree/bindings/dma/dma.txt
-- dma-names: should include "tx" and "rx".
-- clocks: a list of phandle + clock-specifer pairs, one for each entry in clock-names.
-- clock-names: should contain the following:
-   - "i2s_hclk": clock for I2S BUS
-   - "i2s_clk" : clock for I2S controller
-- rockchip,playback-channels: max playback channels, if not set, 8 channels default.
-- rockchip,capture-channels: max capture channels, if not set, 2 channels default.
-
-Required properties for controller which support multi channels
-playback/capture:
-
-- rockchip,grf: the phandle of the syscon node for GRF register.
-
-Example for rk3288 I2S controller:
-
-i2s@ff890000 {
-	compatible = "rockchip,rk3288-i2s", "rockchip,rk3066-i2s";
-	reg = <0xff890000 0x10000>;
-	interrupts = <GIC_SPI 85 IRQ_TYPE_LEVEL_HIGH>;
-	dmas = <&pdma1 0>, <&pdma1 1>;
-	dma-names = "tx", "rx";
-	clock-names = "i2s_hclk", "i2s_clk";
-	clocks = <&cru HCLK_I2S0>, <&cru SCLK_I2S0>;
-	rockchip,playback-channels = <8>;
-	rockchip,capture-channels = <2>;
-};

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

···
+# SPDX-License-Identifier: GPL-2.0
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/rockchip-i2s.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Rockchip I2S controller
+
+description:
+  The I2S bus (Inter-IC sound bus) is a serial link for digital
+  audio data transfer between devices in the system.
+
+maintainers:
+  - Heiko Stuebner <[e-mail]>
+
+properties:
+  compatible:
+    oneOf:
+      - const: rockchip,rk3066-i2s
+      - items:
+          - enum:
+            - rockchip,px30-i2s
+            - rockchip,rk3036-i2s
+            - rockchip,rk3188-i2s
+            - rockchip,rk3228-i2s
+            - rockchip,rk3288-i2s
+            - rockchip,rk3328-i2s
+            - rockchip,rk3366-i2s
+            - rockchip,rk3368-i2s
+            - rockchip,rk3399-i2s
+          - const: rockchip,rk3066-i2s
+
+  reg:
+    maxItems: 1
+
+  interrupts:
+    maxItems: 1
+
+  clocks:
+    items:
+      - description: clock for I2S controller
+      - description: clock for I2S BUS
+
+  clock-names:
+    items:
+      - const: i2s_clk
+      - const: i2s_hclk
+
+  dmas:
+    items:
+      - description: TX DMA Channel
+      - description: RX DMA Channel
+
+  dma-names:
+    items:
+      - const: tx
+      - const: rx
+
+  rockchip,capture-channels:
+    allOf:
+      - $ref: /schemas/types.yaml#/definitions/uint32
+    default: 2
+    description:
+      Max capture channels, if not set, 2 channels default.
+
+  rockchip,playback-channels:
+    allOf:
+      - $ref: /schemas/types.yaml#/definitions/uint32
+    default: 8
+    description:
+      Max playback channels, if not set, 8 channels default.
+
+  rockchip,grf:
+    $ref: /schemas/types.yaml#/definitions/phandle
+    description:
+      The phandle of the syscon node for the GRF register.
+      Required property for controllers which support multi channel
+      playback/capture.
+
+  "#sound-dai-cells":
+    const: 0
+
+required:
+  - compatible
+  - reg
+  - interrupts
+  - clocks
+  - clock-names
+  - dmas
+  - dma-names
+  - "#sound-dai-cells"
+
+additionalProperties: false
+
+examples:
+  - |
+    #include <dt-bindings/clock/rk3288-cru.h>
+    #include <dt-bindings/interrupt-controller/arm-gic.h>
+    #include <dt-bindings/interrupt-controller/irq.h>
+    i2s@ff890000 {
+      compatible = "rockchip,rk3288-i2s", "rockchip,rk3066-i2s";
+      reg = <0xff890000 0x10000>;
+      interrupts = <GIC_SPI 85 IRQ_TYPE_LEVEL_HIGH>;
+      clocks = <&cru SCLK_I2S0>, <&cru HCLK_I2S0>;
+      clock-names = "i2s_clk", "i2s_hclk";
+      dmas = <&pdma1 0>, <&pdma1 1>;
+      dma-names = "tx", "rx";
+      rockchip,capture-channels = <2>;
+      rockchip,playback-channels = <8>;
+      #sound-dai-cells = <0>;
+    };

Documentation/devicetree/bindings/sound/rt5682.txt

···
   The delay time is realtek,btndet-delay value multiple of 8.192 ms.
   If absent, the default is 16.
 
+- #clock-cells : Should be set to '<1>',  wclk and bclk sources provided.
+- clock-output-names : Name given for DAI clocks output.
+
+- clocks : phandle and clock specifier for codec MCLK.
+- clock-names : Clock name string for 'clocks' attribute, should be "mclk".
+
+- realtek,dmic-clk-rate-hz : Set the clock rate (hz) for the requirement of
+  the particular DMIC.
+
+- realtek,dmic-delay-ms : Set the delay time (ms) for the requirement of
+  the particular DMIC.
+
 Pins on the device (for linking into audio routes) for RT5682:
 
   * DMIC L1
···
 	realtek,dmic1-clk-pin = <1>;
 	realtek,jd-src = <1>;
 	realtek,btndet-delay = <16>;
+
+	#clock-cells = <1>;
+	clock-output-names = "rt5682-dai-wclk", "rt5682-dai-bclk";
+
+	clocks = <&osc>;
+	clock-names = "mclk";
 };

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

···
-STMicroelectronics STM32 SPI/I2S Controller
-
-The SPI/I2S block supports I2S/PCM protocols when configured on I2S mode.
-Only some SPI instances support I2S.
-
-Required properties:
-  - compatible: Must be "st,stm32h7-i2s"
-  - reg: Offset and length of the device's register set.
-  - interrupts: Must contain the interrupt line id.
-  - clocks: Must contain phandle and clock specifier pairs for each entry
-	in clock-names.
-  - clock-names: Must contain "i2sclk", "pclk", "x8k" and "x11k".
-	"i2sclk": clock which feeds the internal clock generator
-	"pclk": clock which feeds the peripheral bus interface
-	"x8k": I2S parent clock for sampling rates multiple of 8kHz.
-	"x11k": I2S parent clock for sampling rates multiple of 11.025kHz.
-  - dmas: DMA specifiers for tx and rx dma.
-    See Documentation/devicetree/bindings/dma/stm32-dma.txt.
-  - dma-names: Identifier for each DMA request line. Must be "tx" and "rx".
-  - pinctrl-names: should contain only value "default"
-  - pinctrl-0: see Documentation/devicetree/bindings/pinctrl/st,stm32-pinctrl.yaml
-
-Optional properties:
-  - resets: Reference to a reset controller asserting the reset controller
-
-The device node should contain one 'port' child node with one child 'endpoint'
-node, according to the bindings defined in Documentation/devicetree/bindings/
-graph.txt.
-
-Example:
-sound_card {
-	compatible = "audio-graph-card";
-	dais = <&i2s2_port>;
-};
-
-i2s2: audio-controller@40003800 {
-	compatible = "st,stm32h7-i2s";
-	reg = <0x40003800 0x400>;
-	interrupts = <36>;
-	clocks = <&rcc PCLK1>, <&rcc SPI2_CK>, <&rcc PLL1_Q>, <&rcc PLL2_P>;
-	clock-names = "pclk", "i2sclk",  "x8k", "x11k";
-	dmas = <&dmamux2 2 39 0x400 0x1>,
-           <&dmamux2 3 40 0x400 0x1>;
-	dma-names = "rx", "tx";
-	pinctrl-names = "default";
-	pinctrl-0 = <&pinctrl_i2s2>;
-
-	i2s2_port: port@0 {
-		cpu_endpoint: endpoint {
-			remote-endpoint = <&codec_endpoint>;
-			format = "i2s";
-		};
-	};
-};
-
-audio-codec {
-	codec_port: port@0 {
-		codec_endpoint: endpoint {
-			remote-endpoint = <&cpu_endpoint>;
-		};
-	};
-};

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

···
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/st,stm32-i2s.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: STMicroelectronics STM32 SPI/I2S Controller
+
+maintainers:
+  - Olivier Moysan <[e-mail]>
+
+description:
+  The SPI/I2S block supports I2S/PCM protocols when configured on I2S mode.
+  Only some SPI instances support I2S.
+
+properties:
+  compatible:
+    enum:
+      - st,stm32h7-i2s
+
+  "#sound-dai-cells":
+    const: 0
+
+  reg:
+    maxItems: 1
+
+  clocks:
+    items:
+      - description: clock feeding the peripheral bus interface.
+      - description: clock feeding the internal clock generator.
+      - description: I2S parent clock for sampling rates multiple of 8kHz.
+      - description: I2S parent clock for sampling rates multiple of 11.025kHz.
+
+  clock-names:
+    items:
+      - const: pclk
+      - const: i2sclk
+      - const: x8k
+      - const: x11k
+
+  interrupts:
+    maxItems: 1
+
+  dmas:
+    items:
+      - description: audio capture DMA.
+      - description: audio playback DMA.
+
+  dma-names:
+    items:
+      - const: rx
+      - const: tx
+
+  resets:
+    maxItems: 1
+
+required:
+  - compatible
+  - "#sound-dai-cells"
+  - reg
+  - clocks
+  - clock-names
+  - interrupts
+  - dmas
+  - dma-names
+
+additionalProperties: false
+
+examples:
+  - |
+    #include <dt-bindings/interrupt-controller/arm-gic.h>
+    #include <dt-bindings/clock/stm32mp1-clks.h>
+    i2s2: audio-controller@4000b000 {
+        compatible = "st,stm32h7-i2s";
+        #sound-dai-cells = <0>;
+        reg = <0x4000b000 0x400>;
+        clocks = <&rcc SPI2>, <&rcc SPI2_K>, <&rcc PLL3_Q>, <&rcc PLL3_R>;
+        clock-names = "pclk", "i2sclk", "x8k", "x11k";
+        interrupts = <GIC_SPI 35 IRQ_TYPE_LEVEL_HIGH>;
+        dmas = <&dmamux1 39 0x400 0x01>,
+               <&dmamux1 40 0x400 0x01>;
+        dma-names = "rx", "tx";
+        pinctrl-names = "default";
+        pinctrl-0 = <&i2s2_pins_a>;
+    };
+
+...

Documentation/devicetree/bindings/sound/st,stm32-spdifrx.txt

···
-STMicroelectronics STM32 S/PDIF receiver (SPDIFRX).
-
-The SPDIFRX peripheral, is designed to receive an S/PDIF flow compliant with
-IEC-60958 and IEC-61937.
-
-Required properties:
-  - compatible: should be "st,stm32h7-spdifrx"
-  - reg: cpu DAI IP base address and size
-  - clocks: must contain an entry for kclk (used as S/PDIF signal reference)
-  - clock-names: must contain "kclk"
-  - interrupts: cpu DAI interrupt line
-  - dmas: DMA specifiers for audio data DMA and iec control flow DMA
-    See STM32 DMA bindings, Documentation/devicetree/bindings/dma/st,stm32-dma.yaml
-  - dma-names: two dmas have to be defined, "rx" and "rx-ctrl"
-
-Optional properties:
-  - resets: Reference to a reset controller asserting the SPDIFRX
-
-The device node should contain one 'port' child node with one child 'endpoint'
-node, according to the bindings defined in Documentation/devicetree/bindings/
-graph.txt.
-
-Example:
-spdifrx: spdifrx@40004000 {
-	compatible = "st,stm32h7-spdifrx";
-	reg = <0x40004000 0x400>;
-	clocks = <&rcc SPDIFRX_CK>;
-	clock-names = "kclk";
-	interrupts = <97>;
-	dmas = <&dmamux1 2 93 0x400 0x0>,
-	       <&dmamux1 3 94 0x400 0x0>;
-	dma-names = "rx", "rx-ctrl";
-	pinctrl-0 = <&spdifrx_pins>;
-	pinctrl-names = "default";
-
-	spdifrx_port: port {
-		cpu_endpoint: endpoint {
-			remote-endpoint = <&codec_endpoint>;
-		};
-	};
-};
-
-spdif_in: spdif-in {
-	compatible = "linux,spdif-dir";
-
-	codec_port: port {
-		codec_endpoint: endpoint {
-			remote-endpoint = <&cpu_endpoint>;
-		};
-	};
-};
-
-soundcard {
-	compatible = "audio-graph-card";
-	dais = <&spdifrx_port>;
-};

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

···
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/st,stm32-spdifrx.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: STMicroelectronics STM32 S/PDIF receiver (SPDIFRX)
+
+maintainers:
+  - Olivier Moysan <[e-mail]>
+
+description: |
+  The SPDIFRX peripheral, is designed to receive an S/PDIF flow compliant with
+  IEC-60958 and IEC-61937.
+
+properties:
+  compatible:
+    enum:
+      - st,stm32h7-spdifrx
+
+  "#sound-dai-cells":
+    const: 0
+
+  reg:
+    maxItems: 1
+
+  clocks:
+    maxItems: 1
+
+  clock-names:
+    items:
+      - const: kclk
+
+  interrupts:
+    maxItems: 1
+
+  dmas:
+    items:
+      - description: audio data capture DMA
+      - description: IEC status bits capture DMA
+
+  dma-names:
+    items:
+      - const: rx
+      - const: rx-ctrl
+
+  resets:
+    maxItems: 1
+
+required:
+  - compatible
+  - "#sound-dai-cells"
+  - reg
+  - clocks
+  - clock-names
+  - interrupts
+  - dmas
+  - dma-names
+
+additionalProperties: false
+
+examples:
+  - |
+    #include <dt-bindings/interrupt-controller/arm-gic.h>
+    #include <dt-bindings/clock/stm32mp1-clks.h>
+    spdifrx: spdifrx@40004000 {
+        compatible = "st,stm32h7-spdifrx";
+        #sound-dai-cells = <0>;
+        reg = <0x40004000 0x400>;
+        clocks = <&rcc SPDIF_K>;
+        clock-names = "kclk";
+        interrupts = <GIC_SPI 97 IRQ_TYPE_LEVEL_HIGH>;
+        dmas = <&dmamux1 2 93 0x400 0x0>,
+               <&dmamux1 3 94 0x400 0x0>;
+        dma-names = "rx", "rx-ctrl";
+        pinctrl-0 = <&spdifrx_pins>;
+        pinctrl-names = "default";
+    };
+
+...

Documentation/devicetree/bindings/sound/tas2562.txt

···
 Required properties:
  - #address-cells  - Should be <1>.
  - #size-cells     - Should be <0>.
- - compatible:	   - Should contain "ti,tas2562".
+ - compatible:	   - Should contain "ti,tas2562", "ti,tas2563".
  - reg:		   - The i2c address. Should be 0x4c, 0x4d, 0x4e or 0x4f.
  - ti,imon-slot-no:- TDM TX current sense time slot.
 

Documentation/devicetree/bindings/sound/tlv320adcx140.yaml

···
+# SPDX-License-Identifier: (GPL-2.0+ OR BSD-2-Clause)
+# Copyright (C) 2019 Texas Instruments Incorporated
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/tlv320adcx140.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Texas Instruments TLV320ADCX140 Quad Channel Analog-to-Digital Converter
+
+maintainers:
+  - Dan Murphy <[e-mail]>
+
+description: |
+  The TLV320ADCX140 are multichannel (4-ch analog recording or 8-ch digital
+  PDM microphones recording), high-performance audio, analog-to-digital
+  converter (ADC) with analog inputs supporting up to 2V RMS. The TLV320ADCX140
+  family supports line and  microphone Inputs, and offers a programmable
+  microphone bias or supply voltage generation.
+
+  Specifications can be found at:
+    http://www.ti.com/lit/ds/symlink/tlv320adc3140.pdf
+    http://www.ti.com/lit/ds/symlink/tlv320adc5140.pdf
+    http://www.ti.com/lit/ds/symlink/tlv320adc6140.pdf
+
+properties:
+  compatible:
+    oneOf:
+      - const: ti,tlv320adc3140
+      - const: ti,tlv320adc5140
+      - const: ti,tlv320adc6140
+
+  reg:
+    maxItems: 1
+    description: |
+       I2C addresss of the device can be one of these 0x4c, 0x4d, 0x4e or 0x4f
+
+  reset-gpios:
+    description: |
+       GPIO used for hardware reset.
+
+  areg-supply:
+      description: |
+       Regulator with AVDD at 3.3V.  If not defined then the internal regulator
+       is enabled.
+
+  ti,mic-bias-source:
+    description: |
+       Indicates the source for MIC Bias.
+       0 - Mic bias is set to VREF
+       1 - Mic bias is set to VREF × 1.096
+       6 - Mic bias is set to AVDD
+    allOf:
+      - $ref: /schemas/types.yaml#/definitions/uint32
+      - enum: [0, 1, 6]
+
+  ti,vref-source:
+    description: |
+       Indicates the source for MIC Bias.
+       0 - Set VREF to 2.75V
+       1 - Set VREF to 2.5V
+       2 - Set VREF to 1.375V
+    allOf:
+      - $ref: /schemas/types.yaml#/definitions/uint32
+      - enum: [0, 1, 2]
+
+required:
+  - compatible
+  - reg
+
+examples:
+  - |
+    #include <dt-bindings/gpio/gpio.h>
+    i2c0 {
+      #address-cells = <1>;
+      #size-cells = <0>;
+      codec: codec@4c {
+        compatible = "ti,tlv320adc5140";
+        reg = <0x4c>;
+        ti,mic-bias-source = <6>;
+        reset-gpios = <&gpio0 14 GPIO_ACTIVE_HIGH>;
+      };
+    };

Documentation/sound/alsa-configuration.rst

···
     buffers.  If mmap is used on such architectures, turn off this
     option, so that the DMA-coherent buffers are allocated and used
     instead.
+delayed_register
+    The option is needed for devices that have multiple streams
+    defined in multiple USB interfaces.  The driver may invoke
+    registrations multiple times (once per interface) and this may
+    lead to the insufficient device enumeration.
+    This option receives an array of strings, and you can pass
+    ID:INTERFACE like ``0123abcd:4`` for performing the delayed
+    registration to the given device.  In this example, when a USB
+    device 0123:abcd is probed, the driver waits the registration
+    until the USB interface 4 gets probed.
+    The driver prints a message like "Found post-registration device
+    assignment: 1234abcd:04" for such a device, so that user can
+    notice the need.
 
 This module supports multiple devices, autoprobe and hotplugging.
 

Documentation/sound/hd-audio/index.rst

···
    models
    controls
    dp-mst
+   realtek-pc-beep

Documentation/sound/hd-audio/models.rst

···
     ALC298 fixups on Dell AIO machines
 alc275-dell-xps
     ALC275 fixups on Dell XPS models
-alc256-dell-xps13
-    ALC256 fixups on Dell XPS13
 lenovo-spk-noise
     Workaround for speaker noise on Lenovo machines
 lenovo-hotkey

Documentation/sound/hd-audio/realtek-pc-beep.rst

···
+===============================
+Realtek PC Beep Hidden Register
+===============================
+
+This file documents the "PC Beep Hidden Register", which is present in certain
+Realtek HDA codecs and controls a muxer and pair of passthrough mixers that can
+route audio between pins but aren't themselves exposed as HDA widgets. As far
+as I can tell, these hidden routes are designed to allow flexible PC Beep output
+for codecs that don't have mixer widgets in their output paths. Why it's easier
+to hide a mixer behind an undocumented vendor register than to just expose it
+as a widget, I have no idea.
+
+Register Description
+====================
+
+The register is accessed via processing coefficient 0x36 on NID 20h. Bits not
+identified below have no discernible effect on my machine, a Dell XPS 13 9350::
+
+  MSB                           LSB
+  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+  | |h|S|L|         | B |R|       | Known bits
+  +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
+  |0|0|1|1|  0x7  |0|0x0|1|  0x7  | Reset value
+  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+1Ah input select (B): 2 bits
+  When zero, expose the PC Beep line (from the internal beep generator, when
+  enabled with the Set Beep Generation verb on NID 01h, or else from the
+  external PCBEEP pin) on the 1Ah pin node. When nonzero, expose the headphone
+  jack (or possibly Line In on some machines) input instead. If PC Beep is
+  selected, the 1Ah boost control has no effect.
+
+Amplify 1Ah loopback, left (L): 1 bit
+  Amplify the left channel of 1Ah before mixing it into outputs as specified
+  by h and S bits. Does not affect the level of 1Ah exposed to other widgets.
+
+Amplify 1Ah loopback, right (R): 1 bit
+  Amplify the right channel of 1Ah before mixing it into outputs as specified
+  by h and S bits. Does not affect the level of 1Ah exposed to other widgets.
+
+Loopback 1Ah to 21h [active low] (h): 1 bit
+  When zero, mix 1Ah (possibly with amplification, depending on L and R bits)
+  into 21h (headphone jack on my machine). Mixed signal respects the mute
+  setting on 21h.
+
+Loopback 1Ah to 14h (S): 1 bit
+  When one, mix 1Ah (possibly with amplification, depending on L and R bits)
+  into 14h (internal speaker on my machine). Mixed signal **ignores** the mute
+  setting on 14h and is present whenever 14h is configured as an output.
+
+Path diagrams
+=============
+
+1Ah input selection (DIV is the PC Beep divider set on NID 01h)::
+
+  <Beep generator>   <PCBEEP pin>    <Headphone jack>
+          |                |                |
+          +--DIV--+--!DIV--+       {1Ah boost control}
+                  |                         |
+                  +--(b == 0)--+--(b != 0)--+
+                               |
+               >1Ah (Beep/Headphone Mic/Line In)<
+
+Loopback of 1Ah to 21h/14h::
+
+               <1Ah (Beep/Headphone Mic/Line In)>
+                               |
+                        {amplify if L/R}
+                               |
+                  +-----!h-----+-----S-----+
+                  |                        |
+          {21h mute control}               |
+                  |                        |
+          >21h (Headphone)<     >14h (Internal Speaker)<
+
+Background
+==========
+
+All Realtek HDA codecs have a vendor-defined widget with node ID 20h which
+provides access to a bank of registers that control various codec functions.
+Registers are read and written via the standard HDA processing coefficient
+verbs (Set/Get Coefficient Index, Set/Get Processing Coefficient). The node is
+named "Realtek Vendor Registers" in public datasheets' verb listings and,
+apart from that, is entirely undocumented.
+
+This particular register, exposed at coefficient 0x36 and named in commits from
+Realtek, is of note: unlike most registers, which seem to control detailed
+amplifier parameters not in scope of the HDA specification, it controls audio
+routing which could just as easily have been defined using standard HDA mixer
+and selector widgets.
+
+Specifically, it selects between two sources for the input pin widget with Node
+ID (NID) 1Ah: the widget's signal can come either from an audio jack (on my
+laptop, a Dell XPS 13 9350, it's the headphone jack, but comments in Realtek
+commits indicate that it might be a Line In on some machines) or from the PC
+Beep line (which is itself multiplexed between the codec's internal beep
+generator and external PCBEEP pin, depending on if the beep generator is
+enabled via verbs on NID 01h). Additionally, it can mix (with optional
+amplification) that signal onto the 21h and/or 14h output pins.
+
+The register's reset value is 0x3717, corresponding to PC Beep on 1Ah that is
+then amplified and mixed into both the headphones and the speakers. Not only
+does this violate the HDA specification, which says that "[a vendor defined
+beep input pin] connection may be maintained *only* while the Link reset
+(**RST#**) is asserted", it means that we cannot ignore the register if we care
+about the input that 1Ah would otherwise expose or if the PCBEEP trace is
+poorly shielded and picks up chassis noise (both of which are the case on my
+machine).
+
+Unfortunately, there are lots of ways to get this register configuration wrong.
+Linux, it seems, has gone through most of them. For one, the register resets
+after S3 suspend: judging by existing code, this isn't the case for all vendor
+registers, and it's led to some fixes that improve behavior on cold boot but
+don't last after suspend. Other fixes have successfully switched the 1Ah input
+away from PC Beep but have failed to disable both loopback paths. On my
+machine, this means that the headphone input is amplified and looped back to
+the headphone output, which uses the exact same pins! As you might expect, this
+causes terrible headphone noise, the character of which is controlled by the
+1Ah boost control. (If you've seen instructions online to fix XPS 13 headphone
+noise by changing "Headphone Mic Boost" in ALSA, now you know why.)
+
+The information here has been obtained through black-box reverse engineering of
+the ALC256 codec's behavior and is not guaranteed to be correct. It likely
+also applies for the ALC255, ALC257, ALC235, and ALC236, since those codecs
+seem to be close relatives of the ALC256. (They all share one initialization
+function.) Additionally, other codecs like the ALC225 and ALC285 also have this
+register, judging by existing fixups in ``patch_realtek.c``, but specific
+data (e.g. node IDs, bit positions, pin mappings) for those codecs may differ
+from what I've described here.

Documentation/sound/soc/codec-to-codec.rst

···
 dai names ending with "Playback" and "Capture" respectively as dapm core
 will link and power those dais based on the name.
 
-Note that in current device tree there is no way to mark a dai_link
-as codec to codec. However, it may change in future.
+A dai_link in a "simple-audio-card" will automatically be detected as
+codec to codec when all DAIs on the link belong to codec components.
+The dai_link will be initialized with the subset of stream parameters
+(channels, format, sample rate) supported by all DAIs on the link. Since
+there is no way to provide these parameters in the device tree, this is
+mostly useful for communication with simple fixed-function codecs, such
+as a Bluetooth controller or cellular modem.

MAINTAINERS

···
 F:	sound/soc/codecs/cros_ec_codec.*
 
 CIRRUS LOGIC AUDIO CODEC DRIVERS
-M:	Brian Austin <[e-mail]>
-M:	Paul Handrigan <[e-mail]>
+M:	James Schulman <[e-mail]>
+M:	David Rhodes <[e-mail]>
 L:	[e-mail] (moderated for non-subscribers)
 S:	Maintained
 F:	sound/soc/codecs/cs*
···
 F:	sound/soc/
 F:	include/dt-bindings/sound/
 F:	include/sound/soc*
+
+SOUND - SOUND OPEN FIRMWARE (SOF) DRIVERS
+M:	Pierre-Louis Bossart <[e-mail]>
+M:	Liam Girdwood <[e-mail]>
+M:	Ranjani Sridharan <[e-mail]>
+M:	Kai Vehmanen <[e-mail]>
+M:	Daniel Baluta <[e-mail]>
+L:	[e-mail] (moderated for non-subscribers)
+W:	https://github.com/thesofproject/linux/
+S:	Supported
+F:	sound/soc/sof/
 
 SOUNDWIRE SUBSYSTEM
 M:	Vinod Koul <[e-mail]>

drivers/gpu/drm/mediatek/mtk_hdmi.c

···
 #include <linux/io.h>
 #include <linux/kernel.h>
 #include <linux/mfd/syscon.h>
+#include <linux/mutex.h>
 #include <linux/of_platform.h>
 #include <linux/of.h>
 #include <linux/of_gpio.h>
···
 	bool audio_enable;
 	bool powered;
 	bool enabled;
+	hdmi_codec_plugged_cb plugged_cb;
+	struct device *codec_dev;
+	struct mutex update_plugged_status_lock;
 };
 
 static inline struct mtk_hdmi *hdmi_ctx_from_bridge(struct drm_bridge *b)
···
 	clk_disable_unprepare(hdmi->clk[MTK_HDMI_CLK_AUD_SPDIF]);
 }
 
+static enum drm_connector_status
+mtk_hdmi_update_plugged_status(struct mtk_hdmi *hdmi)
+{
+	bool connected;
+
+	mutex_lock(&hdmi->update_plugged_status_lock);
+	connected = mtk_cec_hpd_high(hdmi->cec_dev);
+	if (hdmi->plugged_cb && hdmi->codec_dev)
+		hdmi->plugged_cb(hdmi->codec_dev, connected);
+	mutex_unlock(&hdmi->update_plugged_status_lock);
+
+	return connected ?
+	       connector_status_connected : connector_status_disconnected;
+}
+
 static enum drm_connector_status hdmi_conn_detect(struct drm_connector *conn,
 						  bool force)
 {
 	struct mtk_hdmi *hdmi = hdmi_ctx_from_conn(conn);
-
-	return mtk_cec_hpd_high(hdmi->cec_dev) ?
-	       connector_status_connected : connector_status_disconnected;
+	return mtk_hdmi_update_plugged_status(hdmi);
 }
 
 static void hdmi_conn_destroy(struct drm_connector *conn)
···
 	return 0;
 }
 
+static int mtk_hdmi_audio_hook_plugged_cb(struct device *dev, void *data,
+					  hdmi_codec_plugged_cb fn,
+					  struct device *codec_dev)
+{
+	struct mtk_hdmi *hdmi = data;
+
+	mutex_lock(&hdmi->update_plugged_status_lock);
+	hdmi->plugged_cb = fn;
+	hdmi->codec_dev = codec_dev;
+	mutex_unlock(&hdmi->update_plugged_status_lock);
+
+	mtk_hdmi_update_plugged_status(hdmi);
+
+	return 0;
+}
+
 static const struct hdmi_codec_ops mtk_hdmi_audio_codec_ops = {
 	.hw_params = mtk_hdmi_audio_hw_params,
 	.audio_startup = mtk_hdmi_audio_startup,
 	.audio_shutdown = mtk_hdmi_audio_shutdown,
 	.digital_mute = mtk_hdmi_audio_digital_mute,
 	.get_eld = mtk_hdmi_audio_get_eld,
+	.hook_plugged_cb = mtk_hdmi_audio_hook_plugged_cb,
 };
 
-static void mtk_hdmi_register_audio_driver(struct device *dev)
+static int mtk_hdmi_register_audio_driver(struct device *dev)
 {
+	struct mtk_hdmi *hdmi = dev_get_drvdata(dev);
 	struct hdmi_codec_pdata codec_data = {
 		.ops = &mtk_hdmi_audio_codec_ops,
 		.max_i2s_channels = 2,
 		.i2s = 1,
+		.data = hdmi,
 	};
 	struct platform_device *pdev;
 
···
 					     PLATFORM_DEVID_AUTO, &codec_data,
 					     sizeof(codec_data));
 	if (IS_ERR(pdev))
-		return;
+		return PTR_ERR(pdev);
 
 	DRM_INFO("%s driver bound to HDMI\n", HDMI_CODEC_DRV_NAME);
+	return 0;
 }
 
 static int mtk_drm_hdmi_probe(struct platform_device *pdev)
···
 		return ret;
 	}
 
+	mutex_init(&hdmi->update_plugged_status_lock);
 	platform_set_drvdata(pdev, hdmi);
 
 	ret = mtk_hdmi_output_init(hdmi);
···
 		return ret;
 	}
 
-	mtk_hdmi_register_audio_driver(dev);
+	ret = mtk_hdmi_register_audio_driver(dev);
+	if (ret) {
+		dev_err(dev, "Failed to register audio driver: %d\n", ret);
+		return ret;
+	}
 
 	hdmi->bridge.funcs = &mtk_hdmi_bridge_funcs;
 	hdmi->bridge.of_node = pdev->dev.of_node;

drivers/soundwire/qcom.c

···
 	struct qcom_swrm_ctrl *ctrl = dev_get_drvdata(dai->dev);
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct sdw_stream_runtime *sruntime;
+	struct snd_soc_dai *codec_dai;
 	int ret, i;
 
 	sruntime = sdw_alloc_stream(dai->name);
···
 
 	ctrl->sruntime[dai->id] = sruntime;
 
-	for (i = 0; i < rtd->num_codecs; i++) {
-		ret = snd_soc_dai_set_sdw_stream(rtd->codec_dais[i], sruntime,
+	for_each_rtd_codec_dais(rtd, i, codec_dai) {
+		ret = snd_soc_dai_set_sdw_stream(codec_dai, sruntime,
 						 substream->stream);
 		if (ret < 0 && ret != -ENOTSUPP) {
 			dev_err(dai->dev, "Failed to set sdw stream on %s",
-				rtd->codec_dais[i]->name);
+				codec_dai->name);
 			sdw_release_stream(sruntime);
 			return ret;
 		}

drivers/soundwire/stream.c

···
 		return ret;
 	}
 
-	/* Program DPN_BlockCtrl1 register */
-	ret = sdw_write(s_rt->slave, addr2, (p_params->bps - 1));
-	if (ret < 0) {
-		dev_err(&s_rt->slave->dev,
-			"DPN_BlockCtrl1 register write failed for port %d\n",
-			t_params->port_num);
-		return ret;
+	if (!dpn_prop->read_only_wordlength) {
+		/* Program DPN_BlockCtrl1 register */
+		ret = sdw_write(s_rt->slave, addr2, (p_params->bps - 1));
+		if (ret < 0) {
+			dev_err(&s_rt->slave->dev,
+				"DPN_BlockCtrl1 register write failed for port %d\n",
+				t_params->port_num);
+			return ret;
+		}
 	}
 
 	/* Program DPN_SampleCtrl1 register */

drivers/spi/Kconfig

···
 
 config SPI_PXA2XX
 	tristate "PXA2xx SSP SPI master"
-	depends on (ARCH_PXA || ARCH_MMP || PCI || ACPI)
+	depends on ARCH_PXA || ARCH_MMP || PCI || ACPI || COMPILE_TEST
 	select PXA_SSP if ARCH_PXA || ARCH_MMP
 	help
 	  This enables using a PXA2xx or Sodaville SSP port as a SPI master

include/dt-bindings/sound/meson-aiu.h

···
+/* SPDX-License-Identifier: GPL-2.0 */
+#ifndef __DT_MESON_AIU_H
+#define __DT_MESON_AIU_H
+
+#define AIU_CPU			0
+#define AIU_HDMI		1
+#define AIU_ACODEC		2
+
+#define CPU_I2S_FIFO		0
+#define CPU_SPDIF_FIFO		1
+#define CPU_I2S_ENCODER		2
+#define CPU_SPDIF_ENCODER	3
+
+#define CTRL_I2S		0
+#define CTRL_PCM		1
+#define CTRL_OUT		2
+
+#endif /* __DT_MESON_AIU_H */

include/dt-bindings/sound/meson-g12a-toacodec.h

···
+/* SPDX-License-Identifier: GPL-2.0 */
+#ifndef __DT_MESON_G12A_TOACODEC_H
+#define __DT_MESON_G12A_TOACODEC_H
+
+#define TOACODEC_IN_A	0
+#define TOACODEC_IN_B	1
+#define TOACODEC_IN_C	2
+#define TOACODEC_OUT	3
+
+#endif /* __DT_MESON_G12A_TOACODEC_H */

include/linux/soundwire/sdw.h

···
  * @max_async_buffer: Number of samples that this port can buffer in
  * asynchronous modes
  * @block_pack_mode: Type of block port mode supported
+ * @read_only_wordlength: Read Only wordlength field in DPN_BlockCtrl1 register
  * @port_encoding: Payload Channel Sample encoding schemes supported
  * @audio_modes: Audio modes supported
  */
···
 	u32 modes;
 	u32 max_async_buffer;
 	bool block_pack_mode;
+	bool read_only_wordlength;
 	u32 port_encoding;
 	struct sdw_dpn_audio_mode *audio_modes;
 };

include/linux/usb/audio-v2.h

···
 	__u8 bmaControls[]; /* variable length */
 } __attribute__((packed));
 
+/* 4.7.2.10 Effect Unit Descriptor */
+
+struct uac2_effect_unit_descriptor {
+	__u8 bLength;
+	__u8 bDescriptorType;
+	__u8 bDescriptorSubtype;
+	__u8 bUnitID;
+	__le16 wEffectType;
+	__u8 bSourceID;
+	__u8 bmaControls[]; /* variable length */
+} __attribute__((packed));
+
 /* 4.9.2 Class-Specific AS Interface Descriptor */
 
 struct uac2_as_header_descriptor {

include/sound/compress_driver.h

···
  * struct snd_compr_runtime: runtime stream description
  * @state: stream state
  * @ops: pointer to DSP callbacks
- * @dma_buffer_p: runtime dma buffer pointer
  * @buffer: pointer to kernel buffer, valid only when not in mmap mode or
  *	DSP doesn't implement copy
  * @buffer_size: size of the above buffer
···
  * @total_bytes_transferred: cumulative bytes transferred by offload DSP
  * @sleep: poll sleep
  * @private_data: driver private data pointer
+ * @dma_area: virtual buffer address
+ * @dma_addr: physical buffer address (not accessible from main CPU)
+ * @dma_bytes: size of DMA area
+ * @dma_buffer_p: runtime dma buffer pointer
  */
 struct snd_compr_runtime {
 	snd_pcm_state_t state;
 	struct snd_compr_ops *ops;
-	struct snd_dma_buffer *dma_buffer_p;
 	void *buffer;
 	u64 buffer_size;
 	u32 fragment_size;
···
 	u64 total_bytes_transferred;
 	wait_queue_head_t sleep;
 	void *private_data;
+
+	unsigned char *dma_area;
+	dma_addr_t dma_addr;
+	size_t dma_bytes;
+	struct snd_dma_buffer *dma_buffer_p;
 };
 
 /**
···
  * @metadata_set: metadata set flag, true when set
  * @next_track: has userspace signal next track transition, true when set
  * @private_data: pointer to DSP private data
+ * @dma_buffer: allocated buffer if any
  */
 struct snd_compr_stream {
 	const char *name;
···
 	bool metadata_set;
 	bool next_track;
 	void *private_data;
+	struct snd_dma_buffer dma_buffer;
 };
 
 /**
···
 
 /**
  * snd_compr_set_runtime_buffer - Set the Compress runtime buffer
- * @substream: compress substream to set
+ * @stream: compress stream to set
  * @bufp: the buffer information, NULL to clear
  *
  * Copy the buffer information to runtime buffer when @bufp is non-NULL.
  * Otherwise it clears the current buffer information.
  */
-static inline void snd_compr_set_runtime_buffer(
-					struct snd_compr_stream *substream,
-					struct snd_dma_buffer *bufp)
+static inline void
+snd_compr_set_runtime_buffer(struct snd_compr_stream *stream,
+			     struct snd_dma_buffer *bufp)
 {
-	struct snd_compr_runtime *runtime = substream->runtime;
+	struct snd_compr_runtime *runtime = stream->runtime;
 
-	runtime->dma_buffer_p = bufp;
+	if (bufp) {
+		runtime->dma_buffer_p = bufp;
+		runtime->dma_area = bufp->area;
+		runtime->dma_addr = bufp->addr;
+		runtime->dma_bytes = bufp->bytes;
+	} else {
+		runtime->dma_buffer_p = NULL;
+		runtime->dma_area = NULL;
+		runtime->dma_addr = 0;
+		runtime->dma_bytes = 0;
+	}
 }
+
+int snd_compr_malloc_pages(struct snd_compr_stream *stream, size_t size);
+int snd_compr_free_pages(struct snd_compr_stream *stream);
 
 int snd_compr_stop_error(struct snd_compr_stream *stream,
 			 snd_pcm_state_t state);

include/sound/core.h

···
 void snd_device_disconnect_all(struct snd_card *card);
 void snd_device_free(struct snd_card *card, void *device_data);
 void snd_device_free_all(struct snd_card *card);
+int snd_device_get_state(struct snd_card *card, void *device_data);
 
 /* isadma.c */
 

include/sound/hdaudio.h

···
 	struct snd_pcm_substream *substream;	/* assigned substream,
 						 * set in PCM open
 						 */
+	struct snd_compr_stream *cstream;
 	unsigned int format_val;	/* format value to be set in the
 					 * controller and the codec
 					 */
···
 	bool locked:1;
 	bool stripe:1;			/* apply stripe control */
 
+	u64 curr_pos;
 	/* timestamp */
 	unsigned long start_wallclk;	/* start + minimum wallclk */
 	unsigned long period_wallclk;	/* wallclk for period */

include/sound/pcm.h

···
 #define snd_pcm_group_for_each_entry(s, substream) \
 	list_for_each_entry(s, &substream->group->substreams, link_list)
 
+#define for_each_pcm_streams(stream)			\
+	for (stream  = SNDRV_PCM_STREAM_PLAYBACK;	\
+	     stream <= SNDRV_PCM_STREAM_LAST;		\
+	     stream++)
+
 /**
  * snd_pcm_running - Check whether the substream is in a running state
  * @substream: substream to check
···
 	return __snd_pcm_lib_xfer(substream, bufs, false, frames, true);
 }
 
-int snd_pcm_limit_hw_rates(struct snd_pcm_runtime *runtime);
+int snd_pcm_hw_limit_rates(struct snd_pcm_hardware *hw);
+
+static inline int
+snd_pcm_limit_hw_rates(struct snd_pcm_runtime *runtime)
+{
+	return snd_pcm_hw_limit_rates(&runtime->hw);
+}
+
 unsigned int snd_pcm_rate_to_rate_bit(unsigned int rate);
 unsigned int snd_pcm_rate_bit_to_rate(unsigned int rate_bit);
 unsigned int snd_pcm_rate_mask_intersect(unsigned int rates_a,
···
 {
 	return 1ULL << (__force int) pcm_format;
 }
+
+/**
+ * pcm_for_each_format - helper to iterate for each format type
+ * @f: the iterator variable in snd_pcm_format_t type
+ */
+#define pcm_for_each_format(f)						\
+	for ((f) = SNDRV_PCM_FORMAT_FIRST;				\
+	     (__force int)(f) <= (__force int)SNDRV_PCM_FORMAT_LAST;	\
+	     (f) = (__force snd_pcm_format_t)((__force int)(f) + 1))
 
 /* printk helpers */
 #define pcm_err(pcm, fmt, args...) \

include/sound/pcm_params.h

···
 	return mask->bits[MASK_OFS(val)] & MASK_BIT(val);
 }
 
+/* Most of drivers need only this one */
+static inline int snd_mask_test_format(const struct snd_mask *mask,
+				       snd_pcm_format_t format)
+{
+	return snd_mask_test(mask, (__force unsigned int)format);
+}
+
 static inline int snd_mask_single(const struct snd_mask *mask)
 {
 	int i, c = 0;

include/sound/rt5682.h

···
 	RT5682_JD1,
 };
 
+enum rt5682_dai_clks {
+	RT5682_DAI_WCLK_IDX,
+	RT5682_DAI_BCLK_IDX,
+	RT5682_DAI_NUM_CLKS,
+};
+
 struct rt5682_platform_data {
 
 	int ldo1_en; /* GPIO for LDO1_EN */
···
 	enum rt5682_dmic1_clk_pin dmic1_clk_pin;
 	enum rt5682_jd_src jd_src;
 	unsigned int btndet_delay;
+	unsigned int dmic_clk_rate;
+	unsigned int dmic_delay;
+
+	const char *dai_clk_names[RT5682_DAI_NUM_CLKS];
 };
 
 #endif

include/sound/soc-acpi.h

···
 };
 
 /**
- * snd_soc_acpi_link_adr: ACPI-based list of _ADR, with a variable
- * number of devices per link
- *
+ * snd_soc_acpi_endpoint - endpoint descriptor
+ * @num: endpoint number (mandatory, unique per device)
+ * @aggregated: 0 (independent) or 1 (logically grouped)
+ * @group_position: zero-based order (only when @aggregated is 1)
+ * @group_id: platform-unique group identifier (only when @aggregrated is 1)
+ */
+struct snd_soc_acpi_endpoint {
+	u8 num;
+	u8 aggregated;
+	u8 group_position;
+	u8 group_id;
+};
+
+/**
+ * snd_soc_acpi_adr_device - descriptor for _ADR-enumerated device
+ * @adr: 64 bit ACPI _ADR value
+ * @num_endpoints: number of endpoints for this device
+ * @endpoints: array of endpoints
+ */
+struct snd_soc_acpi_adr_device {
+	const u64 adr;
+	const u8 num_endpoints;
+	const struct snd_soc_acpi_endpoint *endpoints;
+};
+
+/**
+ * snd_soc_acpi_link_adr - ACPI-based list of _ADR enumerated devices
  * @mask: one bit set indicates the link this list applies to
- * @num_adr: ARRAY_SIZE of adr
- * @adr: array of _ADR (represented as u64).
+ * @num_adr: ARRAY_SIZE of devices
+ * @adr_d: array of devices
+ *
+ * The number of devices per link can be more than 1, e.g. in SoundWire
+ * multi-drop configurations.
  */
 
 struct snd_soc_acpi_link_adr {
 	const u32 mask;
 	const u32 num_adr;
-	const u64 *adr;
+	const struct snd_soc_acpi_adr_device *adr_d;
 };
 
 /**

include/sound/soc-dai.h

···
 
 	int (*set_sdw_stream)(struct snd_soc_dai *dai,
 			void *stream, int direction);
+	void *(*get_sdw_stream)(struct snd_soc_dai *dai, int direction);
+
 	/*
 	 * DAI digital mute - optional.
 	 * Called by soc-core to minimise any pops.
···
 	struct snd_soc_dai_driver *driver;
 
 	/* DAI runtime info */
-	unsigned int capture_active;		/* stream usage count */
-	unsigned int playback_active;		/* stream usage count */
-	unsigned int probed:1;
+	unsigned int stream_active[SNDRV_PCM_STREAM_LAST + 1]; /* usage count */
 
 	unsigned int active;
 
···
 	unsigned int rx_mask;
 
 	struct list_head list;
+
+	/* bit field */
+	unsigned int probed:1;
+	unsigned int started:1;
 };
+
+static inline struct snd_soc_pcm_stream *
+snd_soc_dai_get_pcm_stream(const struct snd_soc_dai *dai, int stream)
+{
+	return (stream == SNDRV_PCM_STREAM_PLAYBACK) ?
+		&dai->driver->playback : &dai->driver->capture;
+}
+
+static inline
+struct snd_soc_dapm_widget *snd_soc_dai_get_widget(
+	struct snd_soc_dai *dai, int stream)
+{
+	return (stream == SNDRV_PCM_STREAM_PLAYBACK) ?
+		dai->playback_widget : dai->capture_widget;
+}
 
 static inline void *snd_soc_dai_get_dma_data(const struct snd_soc_dai *dai,
 					     const struct snd_pcm_substream *ss)
···
 		return dai->driver->ops->set_sdw_stream(dai, stream, direction);
 	else
 		return -ENOTSUPP;
+}
+
+/**
+ * snd_soc_dai_get_sdw_stream() - Retrieves SDW stream from DAI
+ * @dai: DAI
+ * @direction: Stream direction(Playback/Capture)
+ *
+ * This routine only retrieves that was previously configured
+ * with snd_soc_dai_get_sdw_stream()
+ *
+ * Returns pointer to stream or -ENOTSUPP if callback is not supported;
+ */
+static inline void *snd_soc_dai_get_sdw_stream(struct snd_soc_dai *dai,
+					       int direction)
+{
+	if (dai->driver->ops->get_sdw_stream)
+		return dai->driver->ops->get_sdw_stream(dai, direction);
+	else
+		return ERR_PTR(-ENOTSUPP);
 }
 
 #endif

include/sound/soc-dapm.h

···
 	struct snd_soc_dapm_widget_list **list,
 	bool (*custom_stop_condition)(struct snd_soc_dapm_widget *,
 				      enum snd_soc_dapm_direction));
+void snd_soc_dapm_dai_free_widgets(struct snd_soc_dapm_widget_list **list);
 
 struct snd_soc_dapm_context *snd_soc_dapm_kcontrol_dapm(
 	struct snd_kcontrol *kcontrol);
···
 	int num_widgets;
 	struct snd_soc_dapm_widget *widgets[0];
 };
+
+#define for_each_dapm_widgets(list, i, widget)				\
+	for ((i) = 0;							\
+	     (i) < list->num_widgets && (widget = list->widgets[i]);	\
+	     (i)++)
 
 struct snd_soc_dapm_stats {
 	int power_checks;

include/sound/soc-dpcm.h

···
 struct snd_pcm_substream *
 	snd_soc_dpcm_get_substream(struct snd_soc_pcm_runtime *be, int stream);
 
-/* get the BE runtime state */
-enum snd_soc_dpcm_state
-	snd_soc_dpcm_be_get_state(struct snd_soc_pcm_runtime *be, int stream);
-
-/* set the BE runtime state */
-void snd_soc_dpcm_be_set_state(struct snd_soc_pcm_runtime *be, int stream,
-	enum snd_soc_dpcm_state state);
-
-/* internal use only */
-int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute);
-int soc_dpcm_runtime_update(struct snd_soc_card *);
+/* update audio routing between PCMs and any DAI links */
+int snd_soc_dpcm_runtime_update(struct snd_soc_card *card);
 
 #ifdef CONFIG_DEBUG_FS
 void soc_dpcm_debugfs_add(struct snd_soc_pcm_runtime *rtd);
···
 
 int dpcm_path_get(struct snd_soc_pcm_runtime *fe,
 	int stream, struct snd_soc_dapm_widget_list **list_);
+void dpcm_path_put(struct snd_soc_dapm_widget_list **list);
 int dpcm_process_paths(struct snd_soc_pcm_runtime *fe,
 	int stream, struct snd_soc_dapm_widget_list **list, int new);
 int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream);
···
 int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream);
 int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir,
 	int event);
-
-static inline void dpcm_path_put(struct snd_soc_dapm_widget_list **list)
-{
-	kfree(*list);
-}
-
 
 #endif

include/sound/soc.h

···
 void snd_soc_runtime_activate(struct snd_soc_pcm_runtime *rtd, int stream);
 void snd_soc_runtime_deactivate(struct snd_soc_pcm_runtime *rtd, int stream);
 
+int snd_soc_runtime_calc_hw(struct snd_soc_pcm_runtime *rtd,
+			    struct snd_pcm_hardware *hw, int stream);
+
 int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd,
 	unsigned int dai_fmt);
 
···
 	     ((platform) = &link->platforms[i]);			\
 	     (i)++)
 
+#define for_each_link_cpus(link, i, cpu)				\
+	for ((i) = 0;							\
+	     ((i) < link->num_cpus) && ((cpu) = &link->cpus[i]);	\
+	     (i)++)
+
 /*
  * Sample 1 : Single CPU/Codec/Platform
  *
···
 	const struct snd_soc_dapm_route *of_dapm_routes;
 	int num_of_dapm_routes;
 	bool fully_routed;
+	bool disable_route_checks;
 
 	/* lists of probed devices belonging to this card */
 	struct list_head component_dev_list;
···
 #define for_each_card_components(card, component)			\
 	list_for_each_entry(component, &(card)->component_dev_list, card_list)
 
+#define for_each_card_dapms(card, dapm)					\
+	list_for_each_entry(dapm, &card->dapm_list, list)
+
+#define for_each_card_widgets(card, w)\
+	list_for_each_entry(w, &card->widgets, list)
+#define for_each_card_widgets_safe(card, w, _w)	\
+	list_for_each_entry_safe(w, _w, &card->widgets, list)
+
 /* SoC machine DAI configuration, glues a codec and cpu DAI together */
 struct snd_soc_pcm_runtime {
 	struct device *dev;
···
 	struct snd_compr *compr;
 	struct snd_soc_dai *codec_dai;
 	struct snd_soc_dai *cpu_dai;
+	struct snd_soc_dai **dais;
 
 	struct snd_soc_dai **codec_dais;
 	unsigned int num_codecs;
+
+	struct snd_soc_dai **cpu_dais;
+	unsigned int num_cpus;
 
 	struct delayed_work delayed_work;
 	void (*close_delayed_work_func)(struct snd_soc_pcm_runtime *rtd);
···
 	int num_components;
 	struct snd_soc_component *components[0]; /* CPU/Codec/Platform */
 };
+/* see soc_new_pcm_runtime()  */
+#define asoc_rtd_to_cpu(rtd, n)   (rtd)->dais[n]
+#define asoc_rtd_to_codec(rtd, n) (rtd)->dais[n + (rtd)->num_cpus]
+
 #define for_each_rtd_components(rtd, i, component)			\
 	for ((i) = 0;							\
 	     ((i) < rtd->num_components) && ((component) = rtd->components[i]);\
 	     (i)++)
-#define for_each_rtd_codec_dai(rtd, i, dai)\
-	for ((i) = 0;						       \
-	     ((i) < rtd->num_codecs) && ((dai) = rtd->codec_dais[i]); \
+#define for_each_rtd_cpu_dais(rtd, i, dai)				\
+	for ((i) = 0;							\
+	     ((i) < rtd->num_cpus) && ((dai) = rtd->cpu_dais[i]);	\
 	     (i)++)
-#define for_each_rtd_codec_dai_rollback(rtd, i, dai)		\
+#define for_each_rtd_cpu_dais_rollback(rtd, i, dai)		\
+	for (; (--(i) >= 0) && ((dai) = rtd->cpu_dais[i]);)
+#define for_each_rtd_codec_dais(rtd, i, dai)				\
+	for ((i) = 0;							\
+	     ((i) < rtd->num_codecs) && ((dai) = rtd->codec_dais[i]);	\
+	     (i)++)
+#define for_each_rtd_codec_dais_rollback(rtd, i, dai)		\
 	for (; (--(i) >= 0) && ((dai) = rtd->codec_dais[i]);)
+#define for_each_rtd_dais(rtd, i, dai)					\
+	for ((i) = 0;							\
+	     ((i) < (rtd)->num_cpus + (rtd)->num_codecs) &&		\
+		     ((dai) = (rtd)->dais[i]);				\
+	     (i)++)
 
 void snd_soc_close_delayed_work(struct snd_soc_pcm_runtime *rtd);
 

include/sound/sof/dai-intel.h

···
 	uint32_t link_dma_ch;
 } __packed;
 
+/* ALH Configuration Request - SOF_IPC_DAI_ALH_CONFIG */
+struct sof_ipc_dai_alh_params {
+	struct sof_ipc_hdr hdr;
+	uint32_t stream_id;
+
+	/* reserved for future use */
+	uint32_t reserved[15];
+} __packed;
+
 /* DMIC Configuration Request - SOF_IPC_DAI_DMIC_CONFIG */
 
 /* This struct is defined per 2ch PDM controller available in the platform.
···
 
 	/**< variable number of pdm controller config */
 	struct sof_ipc_dai_dmic_pdm_ctrl pdm[0];
-} __packed;
-
-/* ALH Configuration Request - SOF_IPC_DAI_ALH_CONFIG */
-struct sof_ipc_dai_alh_params {
-	struct sof_ipc_hdr hdr;
-	uint32_t stream_id;
-
-	/* reserved for future use */
-	uint32_t reserved[15];
 } __packed;
 
 #endif

include/sound/sof/header.h

···
 #define SOF_IPC_GLB_TRACE_MSG			SOF_GLB_TYPE(0x9U)
 #define SOF_IPC_GLB_GDB_DEBUG                   SOF_GLB_TYPE(0xAU)
 #define SOF_IPC_GLB_TEST_MSG			SOF_GLB_TYPE(0xBU)
+#define SOF_IPC_GLB_PROBE			SOF_GLB_TYPE(0xCU)
 
 /*
  * DSP Command Message Types
···
 #define SOF_IPC_STREAM_POSITION			SOF_CMD_TYPE(0x00a)
 #define SOF_IPC_STREAM_VORBIS_PARAMS		SOF_CMD_TYPE(0x010)
 #define SOF_IPC_STREAM_VORBIS_FREE		SOF_CMD_TYPE(0x011)
+
+/* probe */
+#define SOF_IPC_PROBE_INIT			SOF_CMD_TYPE(0x001)
+#define SOF_IPC_PROBE_DEINIT			SOF_CMD_TYPE(0x002)
+#define SOF_IPC_PROBE_DMA_ADD			SOF_CMD_TYPE(0x003)
+#define SOF_IPC_PROBE_DMA_INFO			SOF_CMD_TYPE(0x004)
+#define SOF_IPC_PROBE_DMA_REMOVE		SOF_CMD_TYPE(0x005)
+#define SOF_IPC_PROBE_POINT_ADD		SOF_CMD_TYPE(0x006)
+#define SOF_IPC_PROBE_POINT_INFO		SOF_CMD_TYPE(0x007)
+#define SOF_IPC_PROBE_POINT_REMOVE		SOF_CMD_TYPE(0x008)
 
 /* trace */
 #define SOF_IPC_TRACE_DMA_PARAMS		SOF_CMD_TYPE(0x001)

include/sound/sof/info.h

···
 
 /* extended data types that can be appended onto end of sof_ipc_fw_ready */
 enum sof_ipc_ext_data {
-	SOF_IPC_EXT_DMA_BUFFER = 0,
-	SOF_IPC_EXT_WINDOW,
-	SOF_IPC_EXT_CC_INFO,
+	SOF_IPC_EXT_UNUSED		= 0,
+	SOF_IPC_EXT_WINDOW		= 1,
+	SOF_IPC_EXT_CC_INFO		= 2,
 };
 
 /* FW version - SOF_IPC_GLB_VERSION */
···
 	struct sof_ipc_cmd_hdr hdr;
 	uint32_t type;		/**< SOF_IPC_EXT_ */
 } __packed;
-
-struct sof_ipc_dma_buffer_elem {
-	struct sof_ipc_hdr hdr;
-	uint32_t type;		/**< SOF_IPC_REGION_ */
-	uint32_t id;		/**< platform specific - used to map to host memory */
-	struct sof_ipc_host_buffer buffer;
-} __packed;
-
-/* extended data DMA buffers for IPC, trace and debug */
-struct sof_ipc_dma_buffer_data {
-	struct sof_ipc_ext_data_hdr ext_hdr;
-	uint32_t num_buffers;
-
-	/* host files in buffer[n].buffer */
-	struct sof_ipc_dma_buffer_elem buffer[];
-}  __packed;
 
 struct sof_ipc_window_elem {
 	struct sof_ipc_hdr hdr;

include/sound/sof/topology.h

···
 	uint32_t id;
 	enum sof_comp_type type;
 	uint32_t pipeline_id;
+	uint32_t core;
 
 	/* reserved for future use */
-	uint32_t reserved[2];
+	uint32_t reserved[1];
 } __packed;
 
 /*

include/uapi/sound/asoc.h

···
 #define __LINUX_UAPI_SND_ASOC_H
 
 #include <linux/types.h>
+#include <sound/asound.h>
 
 /*
  * Maximum number of channels topology kcontrol can represent.

include/uapi/sound/compress_offload.h

···
 #include <sound/compress_params.h>
 
 
-#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 1, 2)
+#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 2, 0)
 /**
  * struct snd_compressed_buffer - compressed buffer
  * @fragment_size: size of buffer fragment in bytes

include/uapi/sound/compress_params.h

···
 #define SND_AUDIOCODEC_G723_1                ((__u32) 0x0000000C)
 #define SND_AUDIOCODEC_G729                  ((__u32) 0x0000000D)
 #define SND_AUDIOCODEC_BESPOKE               ((__u32) 0x0000000E)
-#define SND_AUDIOCODEC_MAX                   SND_AUDIOCODEC_BESPOKE
+#define SND_AUDIOCODEC_ALAC                  ((__u32) 0x0000000F)
+#define SND_AUDIOCODEC_APE                   ((__u32) 0x00000010)
+#define SND_AUDIOCODEC_MAX                   SND_AUDIOCODEC_APE
 
 /*
  * Profile and modes are listed with bit masks. This allows for a
···
 #define SND_AUDIOPROFILE_WMA8                ((__u32) 0x00000002)
 #define SND_AUDIOPROFILE_WMA9                ((__u32) 0x00000004)
 #define SND_AUDIOPROFILE_WMA10               ((__u32) 0x00000008)
+#define SND_AUDIOPROFILE_WMA9_PRO            ((__u32) 0x00000010)
+#define SND_AUDIOPROFILE_WMA9_LOSSLESS       ((__u32) 0x00000020)
+#define SND_AUDIOPROFILE_WMA10_LOSSLESS      ((__u32) 0x00000040)
 
 #define SND_AUDIOMODE_WMA_LEVEL1             ((__u32) 0x00000001)
 #define SND_AUDIOMODE_WMA_LEVEL2             ((__u32) 0x00000002)
···
 	__u16 reserved;
 } __attribute__((packed, aligned(4)));
 
+struct snd_dec_wma {
+	__u32 encoder_option;
+	__u32 adv_encoder_option;
+	__u32 adv_encoder_option2;
+	__u32 reserved;
+} __attribute__((packed, aligned(4)));
+
+struct snd_dec_alac {
+	__u32 frame_length;
+	__u8 compatible_version;
+	__u8 pb;
+	__u8 mb;
+	__u8 kb;
+	__u32 max_run;
+	__u32 max_frame_bytes;
+} __attribute__((packed, aligned(4)));
+
+struct snd_dec_ape {
+	__u16 compatible_version;
+	__u16 compression_level;
+	__u32 format_flags;
+	__u32 blocks_per_frame;
+	__u32 final_frame_blocks;
+	__u32 total_frames;
+	__u32 seek_table_present;
+} __attribute__((packed, aligned(4)));
+
 union snd_codec_options {
 	struct snd_enc_wma wma;
 	struct snd_enc_vorbis vorbis;
···
 	struct snd_enc_flac flac;
 	struct snd_enc_generic generic;
 	struct snd_dec_flac flac_d;
+	struct snd_dec_wma wma_d;
+	struct snd_dec_alac alac_d;
+	struct snd_dec_ape ape_d;
 } __attribute__((packed, aligned(4)));
 
 /** struct snd_codec_desc - description of codec capabilities

include/uapi/sound/sof/abi.h

···
 
 /* SOF ABI version major, minor and patch numbers */
 #define SOF_ABI_MAJOR 3
-#define SOF_ABI_MINOR 12
+#define SOF_ABI_MINOR 13
 #define SOF_ABI_PATCH 0
 
 /* SOF ABI version number. Format within 32bit word is MMmmmppp */

sound/arm/pxa2xx-pcm-lib.c

···
 	struct dma_slave_config config;
 	int ret;
 
-	dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+	dma_params = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream);
 	if (!dma_params)
 		return 0;
 
···
 		return ret;
 
 	snd_dmaengine_pcm_set_config_from_dai_data(substream,
-			snd_soc_dai_get_dma_data(rtd->cpu_dai, substream),
+			snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream),
 			&config);
 
 	ret = dmaengine_slave_config(chan, &config);
···
 
 	runtime->hw = pxa2xx_pcm_hardware;
 
-	dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+	dma_params = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream);
 	if (!dma_params)
 		return 0;
 
···
 		return ret;
 
 	return snd_dmaengine_pcm_open(
-		substream, dma_request_slave_channel(rtd->cpu_dai->dev,
+		substream, dma_request_slave_channel(asoc_rtd_to_cpu(rtd, 0)->dev,
 						     dma_params->chan_name));
 }
 EXPORT_SYMBOL(pxa2xx_pcm_open);

sound/core/compress_offload.c

···
 }
 #endif /* !COMPR_CODEC_CAPS_OVERFLOW */
 
+int snd_compr_malloc_pages(struct snd_compr_stream *stream, size_t size)
+{
+	struct snd_dma_buffer *dmab;
+	int ret;
+
+	if (snd_BUG_ON(!(stream) || !(stream)->runtime))
+		return -EINVAL;
+	dmab = kzalloc(sizeof(*dmab), GFP_KERNEL);
+	if (!dmab)
+		return -ENOMEM;
+	dmab->dev = stream->dma_buffer.dev;
+	ret = snd_dma_alloc_pages(dmab->dev.type, dmab->dev.dev, size, dmab);
+	if (ret < 0) {
+		kfree(dmab);
+		return ret;
+	}
+
+	snd_compr_set_runtime_buffer(stream, dmab);
+	stream->runtime->dma_bytes = size;
+	return 1;
+}
+EXPORT_SYMBOL(snd_compr_malloc_pages);
+
+int snd_compr_free_pages(struct snd_compr_stream *stream)
+{
+	struct snd_compr_runtime *runtime = stream->runtime;
+
+	if (snd_BUG_ON(!(stream) || !(stream)->runtime))
+		return -EINVAL;
+	if (runtime->dma_area == NULL)
+		return 0;
+	if (runtime->dma_buffer_p != &stream->dma_buffer) {
+		/* It's a newly allocated buffer. Release it now. */
+		snd_dma_free_pages(runtime->dma_buffer_p);
+		kfree(runtime->dma_buffer_p);
+	}
+
+	snd_compr_set_runtime_buffer(stream, NULL);
+	return 0;
+}
+EXPORT_SYMBOL(snd_compr_free_pages);
+
 /* revisit this with snd_pcm_preallocate_xxx */
 static int snd_compr_allocate_buffer(struct snd_compr_stream *stream,
 		struct snd_compr_params *params)

sound/core/device.c

···
 	list_for_each_entry_safe_reverse(dev, next, &card->devices, list)
 		__snd_device_free(dev);
 }
+
+/**
+ * snd_device_get_state - Get the current state of the given device
+ * @card: the card instance
+ * @device_data: the data pointer to release
+ *
+ * Returns the current state of the given device object.  For the valid
+ * device, either @SNDRV_DEV_BUILD, @SNDRV_DEV_REGISTERED or
+ * @SNDRV_DEV_DISCONNECTED is returned.
+ * Or for a non-existing device, -1 is returned as an error.
+ */
+int snd_device_get_state(struct snd_card *card, void *device_data)
+{
+	struct snd_device *dev;
+
+	dev = look_for_dev(card, device_data);
+	if (dev)
+		return dev->state;
+	return -1;
+}
+EXPORT_SYMBOL_GPL(snd_device_get_state);

sound/core/info.c

···
  */
 int snd_info_get_line(struct snd_info_buffer *buffer, char *line, int len)
 {
-	int c = -1;
+	int c;
 
 	if (snd_BUG_ON(!buffer || !buffer->buffer))
 		return 1;

sound/core/oss/pcm_oss.c

···
 		sformat = snd_pcm_plug_slave_format(format, sformat_mask);
 
 	if ((__force int)sformat < 0 ||
-	    !snd_mask_test(sformat_mask, (__force int)sformat)) {
-		for (sformat = (__force snd_pcm_format_t)0;
-		     (__force int)sformat <= (__force int)SNDRV_PCM_FORMAT_LAST;
-		     sformat = (__force snd_pcm_format_t)((__force int)sformat + 1)) {
-			if (snd_mask_test(sformat_mask, (__force int)sformat) &&
+	    !snd_mask_test_format(sformat_mask, sformat)) {
+		pcm_for_each_format(sformat) {
+			if (snd_mask_test_format(sformat_mask, sformat) &&
 			    snd_pcm_oss_format_to(sformat) >= 0)
-				break;
+				goto format_found;
 		}
-		if ((__force int)sformat > (__force int)SNDRV_PCM_FORMAT_LAST) {
-			pcm_dbg(substream->pcm, "Cannot find a format!!!\n");
-			err = -EINVAL;
-			goto failure;
-		}
+		pcm_dbg(substream->pcm, "Cannot find a format!!!\n");
+		err = -EINVAL;
+		goto failure;
 	}
+ format_found:
 	err = _snd_pcm_hw_param_set(sparams, SNDRV_PCM_HW_PARAM_FORMAT, (__force int)sformat, 0);
 	if (err < 0)
 		goto failure;
···
 			if (ret < 0)
 				break;
 		}
+		mutex_unlock(&runtime->oss.params_lock);
 		ret = __snd_pcm_lib_xfer(substream, (void *)ptr, true,
 					 frames, in_kernel);
+		mutex_lock(&runtime->oss.params_lock);
 		if (ret != -EPIPE && ret != -ESTRPIPE)
 			break;
 		/* test, if we can't store new data, because the stream */
···
 		ret = snd_pcm_oss_capture_position_fixup(substream, &delay);
 		if (ret < 0)
 			break;
+		mutex_unlock(&runtime->oss.params_lock);
 		ret = __snd_pcm_lib_xfer(substream, (void *)ptr, true,
 					 frames, in_kernel);
+		mutex_lock(&runtime->oss.params_lock);
 		if (ret == -EPIPE) {
 			if (runtime->status->state == SNDRV_PCM_STATE_DRAINING) {
 				ret = snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_DROP, NULL);

sound/core/oss/pcm_plugin.c

···
 	return 0;
 }
 
+static snd_pcm_sframes_t calc_dst_frames(struct snd_pcm_substream *plug,
+					 snd_pcm_sframes_t frames)
+{
+	struct snd_pcm_plugin *plugin, *plugin_next;
+
+	plugin = snd_pcm_plug_first(plug);
+	while (plugin && frames > 0) {
+		plugin_next = plugin->next;
+		if (plugin->dst_frames) {
+			frames = plugin->dst_frames(plugin, frames);
+			if (frames < 0)
+				return frames;
+		}
+		if (frames > plugin->buf_frames)
+			frames = plugin->buf_frames;
+		plugin = plugin_next;
+	}
+	return frames;
+}
+
+static snd_pcm_sframes_t calc_src_frames(struct snd_pcm_substream *plug,
+					 snd_pcm_sframes_t frames)
+{
+	struct snd_pcm_plugin *plugin, *plugin_prev;
+
+	plugin = snd_pcm_plug_last(plug);
+	while (plugin && frames > 0) {
+		if (frames > plugin->buf_frames)
+			frames = plugin->buf_frames;
+		plugin_prev = plugin->prev;
+		if (plugin->src_frames) {
+			frames = plugin->src_frames(plugin, frames);
+			if (frames < 0)
+				return frames;
+		}
+		plugin = plugin_prev;
+	}
+	return frames;
+}
+
 snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *plug, snd_pcm_uframes_t drv_frames)
 {
-	struct snd_pcm_plugin *plugin, *plugin_prev, *plugin_next;
-	int stream;
-
 	if (snd_BUG_ON(!plug))
 		return -ENXIO;
-	if (drv_frames == 0)
-		return 0;
-	stream = snd_pcm_plug_stream(plug);
-	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		plugin = snd_pcm_plug_last(plug);
-		while (plugin && drv_frames > 0) {
-			if (drv_frames > plugin->buf_frames)
-				drv_frames = plugin->buf_frames;
-			plugin_prev = plugin->prev;
-			if (plugin->src_frames)
-				drv_frames = plugin->src_frames(plugin, drv_frames);
-			plugin = plugin_prev;
-		}
-	} else if (stream == SNDRV_PCM_STREAM_CAPTURE) {
-		plugin = snd_pcm_plug_first(plug);
-		while (plugin && drv_frames > 0) {
-			plugin_next = plugin->next;
-			if (plugin->dst_frames)
-				drv_frames = plugin->dst_frames(plugin, drv_frames);
-			if (drv_frames > plugin->buf_frames)
-				drv_frames = plugin->buf_frames;
-			plugin = plugin_next;
-		}
-	} else
+	switch (snd_pcm_plug_stream(plug)) {
+	case SNDRV_PCM_STREAM_PLAYBACK:
+		return calc_src_frames(plug, drv_frames);
+	case SNDRV_PCM_STREAM_CAPTURE:
+		return calc_dst_frames(plug, drv_frames);
+	default:
 		snd_BUG();
-	return drv_frames;
+		return -EINVAL;
+	}
 }
 
 snd_pcm_sframes_t snd_pcm_plug_slave_size(struct snd_pcm_substream *plug, snd_pcm_uframes_t clt_frames)
 {
-	struct snd_pcm_plugin *plugin, *plugin_prev, *plugin_next;
-	snd_pcm_sframes_t frames;
-	int stream;
-	
 	if (snd_BUG_ON(!plug))
 		return -ENXIO;
-	if (clt_frames == 0)
-		return 0;
-	frames = clt_frames;
-	stream = snd_pcm_plug_stream(plug);
-	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		plugin = snd_pcm_plug_first(plug);
-		while (plugin && frames > 0) {
-			plugin_next = plugin->next;
-			if (plugin->dst_frames) {
-				frames = plugin->dst_frames(plugin, frames);
-				if (frames < 0)
-					return frames;
-			}
-			if (frames > plugin->buf_frames)
-				frames = plugin->buf_frames;
-			plugin = plugin_next;
-		}
-	} else if (stream == SNDRV_PCM_STREAM_CAPTURE) {
-		plugin = snd_pcm_plug_last(plug);
-		while (plugin) {
-			if (frames > plugin->buf_frames)
-				frames = plugin->buf_frames;
-			plugin_prev = plugin->prev;
-			if (plugin->src_frames) {
-				frames = plugin->src_frames(plugin, frames);
-				if (frames < 0)
-					return frames;
-			}
-			plugin = plugin_prev;
-		}
-	} else
+	switch (snd_pcm_plug_stream(plug)) {
+	case SNDRV_PCM_STREAM_PLAYBACK:
+		return calc_dst_frames(plug, clt_frames);
+	case SNDRV_PCM_STREAM_CAPTURE:
+		return calc_src_frames(plug, clt_frames);
+	default:
 		snd_BUG();
-	return frames;
+		return -EINVAL;
+	}
 }
 
 static int snd_pcm_plug_formats(const struct snd_mask *mask,

sound/core/oss/rate.c

···
 	unsigned int pos;
 	rate_f func;
 	snd_pcm_sframes_t old_src_frames, old_dst_frames;
-	struct rate_channel channels[0];
+	struct rate_channel channels[];
 };
 
 static void rate_init(struct snd_pcm_plugin *plugin)

sound/core/pcm.c

···
 		str = "none";
 	else
 		str = strs[pcm->dev_class];
-        return snprintf(buf, PAGE_SIZE, "%s\n", str);
+	return sprintf(buf, "%s\n", str);
 }
 
 static DEVICE_ATTR(pcm_class, 0444, show_pcm_class, NULL);

sound/core/pcm_dmaengine.c

···
 snd_pcm_uframes_t snd_dmaengine_pcm_pointer(struct snd_pcm_substream *substream)
 {
 	struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream);
+	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct dma_tx_state state;
 	enum dma_status status;
 	unsigned int buf_size;
···
 		buf_size = snd_pcm_lib_buffer_bytes(substream);
 		if (state.residue > 0 && state.residue <= buf_size)
 			pos = buf_size - state.residue;
+
+		runtime->delay = bytes_to_frames(runtime,
+						 state.in_flight_bytes);
 	}
 
-	return bytes_to_frames(substream->runtime, pos);
+	return bytes_to_frames(runtime, pos);
 }
 EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_pointer);
 
···
 		 * default assumption is that it supports 1, 2 and 4 bytes
 		 * widths.
 		 */
-		for (i = SNDRV_PCM_FORMAT_FIRST; i <= SNDRV_PCM_FORMAT_LAST; i++) {
+		pcm_for_each_format(i) {
 			int bits = snd_pcm_format_physical_width(i);
 
 			/*

sound/core/pcm_misc.c

···
 /* we do lots of calculations on snd_pcm_format_t; shut up sparse */
 #define INT	__force int
 
+static bool valid_format(snd_pcm_format_t format)
+{
+	return (INT)format >= 0 && (INT)format <= (INT)SNDRV_PCM_FORMAT_LAST;
+}
+
 static const struct pcm_format_data pcm_formats[(INT)SNDRV_PCM_FORMAT_LAST+1] = {
 	[SNDRV_PCM_FORMAT_S8] = {
 		.width = 8, .phys = 8, .le = -1, .signd = 1,
···
 int snd_pcm_format_signed(snd_pcm_format_t format)
 {
 	int val;
-	if ((INT)format < 0 || (INT)format > (INT)SNDRV_PCM_FORMAT_LAST)
+	if (!valid_format(format))
 		return -EINVAL;
 	if ((val = pcm_formats[(INT)format].signd) < 0)
 		return -EINVAL;
···
 int snd_pcm_format_little_endian(snd_pcm_format_t format)
 {
 	int val;
-	if ((INT)format < 0 || (INT)format > (INT)SNDRV_PCM_FORMAT_LAST)
+	if (!valid_format(format))
 		return -EINVAL;
 	if ((val = pcm_formats[(INT)format].le) < 0)
 		return -EINVAL;
···
 int snd_pcm_format_width(snd_pcm_format_t format)
 {
 	int val;
-	if ((INT)format < 0 || (INT)format > (INT)SNDRV_PCM_FORMAT_LAST)
+	if (!valid_format(format))
 		return -EINVAL;
 	if ((val = pcm_formats[(INT)format].width) == 0)
 		return -EINVAL;
···
 int snd_pcm_format_physical_width(snd_pcm_format_t format)
 {
 	int val;
-	if ((INT)format < 0 || (INT)format > (INT)SNDRV_PCM_FORMAT_LAST)
+	if (!valid_format(format))
 		return -EINVAL;
 	if ((val = pcm_formats[(INT)format].phys) == 0)
 		return -EINVAL;
···
  */
 const unsigned char *snd_pcm_format_silence_64(snd_pcm_format_t format)
 {
-	if ((INT)format < 0 || (INT)format > (INT)SNDRV_PCM_FORMAT_LAST)
+	if (!valid_format(format))
 		return NULL;
 	if (! pcm_formats[(INT)format].phys)
 		return NULL;
···
 	unsigned char *dst;
 	const unsigned char *pat;
 
-	if ((INT)format < 0 || (INT)format > (INT)SNDRV_PCM_FORMAT_LAST)
+	if (!valid_format(format))
 		return -EINVAL;
 	if (samples == 0)
 		return 0;
···
 EXPORT_SYMBOL(snd_pcm_format_set_silence);
 
 /**
- * snd_pcm_limit_hw_rates - determine rate_min/rate_max fields
- * @runtime: the runtime instance
+ * snd_pcm_hw_limit_rates - determine rate_min/rate_max fields
+ * @hw: the pcm hw instance
  *
  * Determines the rate_min and rate_max fields from the rates bits of
- * the given runtime->hw.
+ * the given hw.
  *
  * Return: Zero if successful.
  */
-int snd_pcm_limit_hw_rates(struct snd_pcm_runtime *runtime)
+int snd_pcm_hw_limit_rates(struct snd_pcm_hardware *hw)
 {
 	int i;
 	for (i = 0; i < (int)snd_pcm_known_rates.count; i++) {
-		if (runtime->hw.rates & (1 << i)) {
-			runtime->hw.rate_min = snd_pcm_known_rates.list[i];
+		if (hw->rates & (1 << i)) {
+			hw->rate_min = snd_pcm_known_rates.list[i];
 			break;
 		}
 	}
 	for (i = (int)snd_pcm_known_rates.count - 1; i >= 0; i--) {
-		if (runtime->hw.rates & (1 << i)) {
-			runtime->hw.rate_max = snd_pcm_known_rates.list[i];
+		if (hw->rates & (1 << i)) {
+			hw->rate_max = snd_pcm_known_rates.list[i];
 			break;
 		}
 	}
 	return 0;
 }
-EXPORT_SYMBOL(snd_pcm_limit_hw_rates);
+EXPORT_SYMBOL(snd_pcm_hw_limit_rates);
 
 /**
  * snd_pcm_rate_to_rate_bit - converts sample rate to SNDRV_PCM_RATE_xxx bit

sound/core/pcm_native.c

···
 	return err;
 }
 
+/* macro for simplified cast */
+#define PARAM_MASK_BIT(b)	(1U << (__force int)(b))
+
 static bool hw_support_mmap(struct snd_pcm_substream *substream)
 {
 	if (!(substream->runtime->hw.info & SNDRV_PCM_INFO_MMAP))
···
 			return -EINVAL;
 
 		/* This parameter is not requested to change by a caller. */
-		if (!(params->rmask & (1 << k)))
+		if (!(params->rmask & PARAM_MASK_BIT(k)))
 			continue;
 
 		if (trace_hw_mask_param_enabled())
···
 
 		/* Set corresponding flag so that the caller gets it. */
 		trace_hw_mask_param(substream, k, 0, &old_mask, m);
-		params->cmask |= 1 << k;
+		params->cmask |= PARAM_MASK_BIT(k);
 	}
 
 	return 0;
···
 			return -EINVAL;
 
 		/* This parameter is not requested to change by a caller. */
-		if (!(params->rmask & (1 << k)))
+		if (!(params->rmask & PARAM_MASK_BIT(k)))
 			continue;
 
 		if (trace_hw_interval_param_enabled())
···
 
 		/* Set corresponding flag so that the caller gets it. */
 		trace_hw_interval_param(substream, k, 0, &old_interval, i);
-		params->cmask |= 1 << k;
+		params->cmask |= PARAM_MASK_BIT(k);
 	}
 
 	return 0;
···
 	 * have 0 so that the parameters are never changed anymore.
 	 */
 	for (k = 0; k <= SNDRV_PCM_HW_PARAM_LAST_INTERVAL; k++)
-		vstamps[k] = (params->rmask & (1 << k)) ? 1 : 0;
+		vstamps[k] = (params->rmask & PARAM_MASK_BIT(k)) ? 1 : 0;
 
 	/* Due to the above design, actual sequence number starts at 2. */
 	stamp = 2;
···
 					hw_param_interval(params, r->var));
 			}
 
-			params->cmask |= (1 << r->var);
+			params->cmask |= PARAM_MASK_BIT(r->var);
 			vstamps[r->var] = stamp;
 			again = true;
 		}
···
 
 	params->info = 0;
 	params->fifo_size = 0;
-	if (params->rmask & (1 << SNDRV_PCM_HW_PARAM_SAMPLE_BITS))
+	if (params->rmask & PARAM_MASK_BIT(SNDRV_PCM_HW_PARAM_SAMPLE_BITS))
 		params->msbits = 0;
-	if (params->rmask & (1 << SNDRV_PCM_HW_PARAM_RATE)) {
+	if (params->rmask & PARAM_MASK_BIT(SNDRV_PCM_HW_PARAM_RATE)) {
 		params->rate_num = 0;
 		params->rate_den = 0;
 	}
···
 static int snd_pcm_hw_rule_format(struct snd_pcm_hw_params *params,
 				  struct snd_pcm_hw_rule *rule)
 {
-	unsigned int k;
+	snd_pcm_format_t k;
 	const struct snd_interval *i =
 				hw_param_interval_c(params, rule->deps[0]);
 	struct snd_mask m;
 	struct snd_mask *mask = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
 	snd_mask_any(&m);
-	for (k = 0; k <= SNDRV_PCM_FORMAT_LAST; ++k) {
+	pcm_for_each_format(k) {
 		int bits;
-		if (! snd_mask_test(mask, k))
+		if (!snd_mask_test_format(mask, k))
 			continue;
 		bits = snd_pcm_format_physical_width(k);
 		if (bits <= 0)
 			continue; /* ignore invalid formats */
 		if ((unsigned)bits < i->min || (unsigned)bits > i->max)
-			snd_mask_reset(&m, k);
+			snd_mask_reset(&m, (__force unsigned)k);
 	}
 	return snd_mask_refine(mask, &m);
 }
···
 				       struct snd_pcm_hw_rule *rule)
 {
 	struct snd_interval t;
-	unsigned int k;
+	snd_pcm_format_t k;
+
 	t.min = UINT_MAX;
 	t.max = 0;
 	t.openmin = 0;
 	t.openmax = 0;
-	for (k = 0; k <= SNDRV_PCM_FORMAT_LAST; ++k) {
+	pcm_for_each_format(k) {
 		int bits;
-		if (! snd_mask_test(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), k))
+		if (!snd_mask_test_format(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), k))
 			continue;
 		bits = snd_pcm_format_physical_width(k);
 		if (bits <= 0)
···
 	unsigned int mask = 0;
 
         if (hw->info & SNDRV_PCM_INFO_INTERLEAVED)
-		mask |= 1 << SNDRV_PCM_ACCESS_RW_INTERLEAVED;
+		mask |= PARAM_MASK_BIT(SNDRV_PCM_ACCESS_RW_INTERLEAVED);
         if (hw->info & SNDRV_PCM_INFO_NONINTERLEAVED)
-		mask |= 1 << SNDRV_PCM_ACCESS_RW_NONINTERLEAVED;
+		mask |= PARAM_MASK_BIT(SNDRV_PCM_ACCESS_RW_NONINTERLEAVED);
 	if (hw_support_mmap(substream)) {
 		if (hw->info & SNDRV_PCM_INFO_INTERLEAVED)
-			mask |= 1 << SNDRV_PCM_ACCESS_MMAP_INTERLEAVED;
+			mask |= PARAM_MASK_BIT(SNDRV_PCM_ACCESS_MMAP_INTERLEAVED);
 		if (hw->info & SNDRV_PCM_INFO_NONINTERLEAVED)
-			mask |= 1 << SNDRV_PCM_ACCESS_MMAP_NONINTERLEAVED;
+			mask |= PARAM_MASK_BIT(SNDRV_PCM_ACCESS_MMAP_NONINTERLEAVED);
 		if (hw->info & SNDRV_PCM_INFO_COMPLEX)
-			mask |= 1 << SNDRV_PCM_ACCESS_MMAP_COMPLEX;
+			mask |= PARAM_MASK_BIT(SNDRV_PCM_ACCESS_MMAP_COMPLEX);
 	}
 	err = snd_pcm_hw_constraint_mask(runtime, SNDRV_PCM_HW_PARAM_ACCESS, mask);
 	if (err < 0)
···
 	if (err < 0)
 		return err;
 
-	err = snd_pcm_hw_constraint_mask(runtime, SNDRV_PCM_HW_PARAM_SUBFORMAT, 1 << SNDRV_PCM_SUBFORMAT_STD);
+	err = snd_pcm_hw_constraint_mask(runtime, SNDRV_PCM_HW_PARAM_SUBFORMAT,
+					 PARAM_MASK_BIT(SNDRV_PCM_SUBFORMAT_STD));
 	if (err < 0)
 		return err;
 

sound/drivers/aloop.c

···
 struct loopback_setup {
 	unsigned int notify: 1;
 	unsigned int rate_shift;
-	unsigned int format;
+	snd_pcm_format_t format;
 	unsigned int rate;
 	unsigned int channels;
 	struct snd_ctl_elem_id active_id;
···
 	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
 	uinfo->count = 1;
 	uinfo->value.integer.min = 0;
-	uinfo->value.integer.max = SNDRV_PCM_FORMAT_LAST;
+	uinfo->value.integer.max = (__force int)SNDRV_PCM_FORMAT_LAST;
 	uinfo->value.integer.step = 1;
 	return 0;
 }                                  
···
 	struct loopback *loopback = snd_kcontrol_chip(kcontrol);
 	
 	ucontrol->value.integer.value[0] =
-		loopback->setup[kcontrol->id.subdevice]
+		(__force int)loopback->setup[kcontrol->id.subdevice]
 			       [kcontrol->id.device].format;
 	return 0;
 }

sound/drivers/dummy.c

···
 static void print_formats(struct snd_dummy *dummy,
 			  struct snd_info_buffer *buffer)
 {
-	int i;
+	snd_pcm_format_t i;
 
-	for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) {
-		if (dummy->pcm_hw.formats & (1ULL << i))
+	pcm_for_each_format(i) {
+		if (dummy->pcm_hw.formats & pcm_format_to_bits(i))
 			snd_iprintf(buffer, " %s", snd_pcm_format_name(i));
 	}
 }

sound/firewire/bebob/bebob.c

···
 static struct fw_driver bebob_driver = {
 	.driver = {
 		.owner	= THIS_MODULE,
-		.name	= "snd-bebob",
+		.name	= KBUILD_MODNAME,
 		.bus	= &fw_bus_type,
 	},
 	.probe    = bebob_probe,

sound/firewire/digi00x/digi00x.c

···
 static struct fw_driver dg00x_driver = {
 	.driver = {
 		.owner = THIS_MODULE,
-		.name = "snd-firewire-digi00x",
+		.name = KBUILD_MODNAME,
 		.bus = &fw_bus_type,
 	},
 	.probe    = snd_dg00x_probe,

sound/firewire/fireface/ff.c

···
 static struct fw_driver ff_driver = {
 	.driver = {
 		.owner	= THIS_MODULE,
-		.name	= "snd-fireface",
+		.name	= KBUILD_MODNAME,
 		.bus	= &fw_bus_type,
 	},
 	.probe    = snd_ff_probe,

sound/firewire/fireworks/fireworks.c

···
 static struct fw_driver efw_driver = {
 	.driver = {
 		.owner = THIS_MODULE,
-		.name = "snd-fireworks",
+		.name = KBUILD_MODNAME,
 		.bus = &fw_bus_type,
 	},
 	.probe    = efw_probe,

sound/firewire/tascam/tascam-hwdep.c

···
 
 static long tscm_hwdep_read_locked(struct snd_tscm *tscm, char __user *buf,
 				   long count, loff_t *offset)
+	__releases(&tscm->lock)
 {
 	struct snd_firewire_event_lock_status event = {
 		.type = SNDRV_FIREWIRE_EVENT_LOCK_STATUS,
···
 
 static long tscm_hwdep_read_queue(struct snd_tscm *tscm, char __user *buf,
 				  long remained, loff_t *offset)
+	__releases(&tscm->lock)
 {
 	char __user *pos = buf;
 	unsigned int type = SNDRV_FIREWIRE_EVENT_TASCAM_CONTROL;

sound/firewire/tascam/tascam.c

···
 static struct fw_driver tscm_driver = {
 	.driver = {
 		.owner = THIS_MODULE,
-		.name = "snd-firewire-tascam",
+		.name = KBUILD_MODNAME,
 		.bus = &fw_bus_type,
 	},
 	.probe    = snd_tscm_probe,

sound/hda/hdac_device.c

···
  */
 int snd_hdac_codec_modalias(struct hdac_device *codec, char *buf, size_t size)
 {
-	return snprintf(buf, size, "hdaudio:v%08Xr%08Xa%02X\n",
+	return scnprintf(buf, size, "hdaudio:v%08Xr%08Xa%02X\n",
 			codec->vendor_id, codec->revision_id, codec->type);
 }
 EXPORT_SYMBOL_GPL(snd_hdac_codec_modalias);

sound/isa/sb/emu8000_pcm.c

···
 #define LOOP_WRITE(rec, offset, _buf, count, mode)		\
 	do {							\
 		struct snd_emu8000 *emu = (rec)->emu;		\
-		unsigned short *buf = (unsigned short *)(_buf); \
+		unsigned short *buf = (__force unsigned short *)(_buf); \
 		snd_emu8000_write_wait(emu, 1);			\
 		EMU8000_SMALW_WRITE(emu, offset);		\
 		while (count > 0) {				\
···
 #define LOOP_WRITE(rec, pos, _buf, count, mode)				\
 	do {								\
 		struct snd_emu8000 *emu = rec->emu;			\
-		unsigned short *buf = (unsigned short *)(_buf);		\
+		unsigned short *buf = (__force unsigned short *)(_buf);	\
 		snd_emu8000_write_wait(emu, 1);				\
 		EMU8000_SMALW_WRITE(emu, pos + rec->loop_start[0]);	\
 		if (rec->voices > 1)					\

sound/pci/ali5451/ali5451.c

···
 {
 	struct snd_ali *codec = snd_pcm_substream_chip(substream);
 	struct snd_pcm_substream *s;
-	unsigned int what, whati, capture_flag;
+	unsigned int what, whati;
 	struct snd_ali_voice *pvoice, *evoice;
 	unsigned int val;
 	int do_start;
···
 		return -EINVAL;
 	}
 
-	what = whati = capture_flag = 0;
+	what = whati = 0;
 	snd_pcm_group_for_each_entry(s, substream) {
 		if ((struct snd_ali *) snd_pcm_substream_chip(s) == codec) {
 			pvoice = s->runtime->private_data;
···
 					evoice->running = 0;
 			}
 			snd_pcm_trigger_done(s, substream);
-			if (pvoice->mode)
-				capture_flag = 1;
 		}
 	}
 	spin_lock(&codec->reg_lock);

sound/pci/emu10k1/emu10k1_main.c

···
 	int idx, err;
 	int is_audigy;
 	size_t page_table_size;
+	__le32 *pgtbl;
 	unsigned int silent_page;
 	const struct snd_emu_chip_details *c;
 	static const struct snd_device_ops ops = {
···
 	/* Clear silent pages and set up pointers */
 	memset(emu->silent_page.area, 0, emu->silent_page.bytes);
 	silent_page = emu->silent_page.addr << emu->address_mode;
+	pgtbl = (__le32 *)emu->ptb_pages.area;
 	for (idx = 0; idx < (emu->address_mode ? MAXPAGES1 : MAXPAGES0); idx++)
-		((u32 *)emu->ptb_pages.area)[idx] = cpu_to_le32(silent_page | idx);
+		pgtbl[idx] = cpu_to_le32(silent_page | idx);
 
 	/* set up voice indices */
 	for (idx = 0; idx < NUM_G; idx++) {

sound/pci/hda/Kconfig

···
 config SND_HDA_CODEC_CA0132_DSP
 	bool "Support new DSP code for CA0132 codec"
 	depends on SND_HDA_CODEC_CA0132
+	default y
 	select SND_HDA_DSP_LOADER
 	select FW_LOADER
 	help

sound/pci/hda/hda_codec.c

···
 	struct list_head list;
 	int len;
 	hda_nid_t nid;
-	hda_nid_t conns[0];
+	hda_nid_t conns[];
 };
 
 /* look up the cached results */

sound/pci/hda/hda_controller.c

···
 	u32 wallclk_ctr, wallclk_cycles;
 	bool direction;
 	u32 dma_select;
-	u32 timeout = 200;
+	u32 timeout;
 	u32 retry_count = 0;
 
 	runtime = substream->runtime;

sound/pci/hda/patch_ca0132.c

···
 	SND_PCI_QUIRK(0x1458, 0xA016, "Recon3Di", QUIRK_R3DI),
 	SND_PCI_QUIRK(0x1458, 0xA026, "Gigabyte G1.Sniper Z97", QUIRK_R3DI),
 	SND_PCI_QUIRK(0x1458, 0xA036, "Gigabyte GA-Z170X-Gaming 7", QUIRK_R3DI),
+	SND_PCI_QUIRK(0x3842, 0x1038, "EVGA X99 Classified", QUIRK_R3DI),
 	SND_PCI_QUIRK(0x1102, 0x0013, "Recon3D", QUIRK_R3D),
 	SND_PCI_QUIRK(0x1102, 0x0051, "Sound Blaster AE-5", QUIRK_AE5),
 	{}
···
 	u32 magic;
 	u32 chip_addr;
 	u32 count;
-	u32 data[0];
+	u32 data[];
 };
 
 static const u32 g_magic_value = 0x4c46584d;

sound/pci/hda/patch_hdmi.c

···
 	struct hda_multi_out multiout;
 	struct hda_pcm_stream pcm_playback;
 
-	bool use_jack_detect; /* jack detection enabled */
 	bool use_acomp_notifier; /* use eld_notify callback for hotplug */
 	bool acomp_registered; /* audio component registered in this driver */
 	struct drm_audio_component_audio_ops drm_audio_ops;
···
  * Unsolicited events
  */
 
-static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll);
+static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll);
 
 static void check_presence_and_report(struct hda_codec *codec, hda_nid_t nid,
 				      int dev_id)
···
 	if (pin_idx < 0)
 		return;
 	mutex_lock(&spec->pcm_lock);
-	if (hdmi_present_sense(get_pin(spec, pin_idx), 1))
-		snd_hda_jack_report_sync(codec);
+	hdmi_present_sense(get_pin(spec, pin_idx), 1);
 	mutex_unlock(&spec->pcm_lock);
 }
 
···
 	check_presence_and_report(codec, jack->nid, jack->dev_id);
 }
 
-static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res)
+static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res,
+				 struct hda_jack_tbl *jack)
 {
-	int tag = res >> AC_UNSOL_RES_TAG_SHIFT;
-	struct hda_jack_tbl *jack;
-
-	if (codec->dp_mst) {
-		int dev_entry =
-			(res & AC_UNSOL_RES_DE) >> AC_UNSOL_RES_DE_SHIFT;
-
-		jack = snd_hda_jack_tbl_get_from_tag(codec, tag, dev_entry);
-	} else {
-		jack = snd_hda_jack_tbl_get_from_tag(codec, tag, 0);
-	}
-	if (!jack)
-		return;
 	jack->jack_dirty = 1;
 
 	codec_dbg(codec,
···
 	}
 
 	if (subtag == 0)
-		hdmi_intrinsic_event(codec, res);
+		hdmi_intrinsic_event(codec, res, jack);
 	else
 		hdmi_non_intrinsic_event(codec, res);
 }
···
 	per_pin->channels = 0;
 }
 
+static struct snd_jack *pin_idx_to_pcm_jack(struct hda_codec *codec,
+					    struct hdmi_spec_per_pin *per_pin)
+{
+	struct hdmi_spec *spec = codec->spec;
+
+	if (per_pin->pcm_idx >= 0)
+		return spec->pcm_rec[per_pin->pcm_idx].jack;
+	else
+		return NULL;
+}
+
 /* update per_pin ELD from the given new ELD;
  * setup info frame and notification accordingly
+ * also notify ELD kctl and report jack status changes
  */
-static bool update_eld(struct hda_codec *codec,
+static void update_eld(struct hda_codec *codec,
 		       struct hdmi_spec_per_pin *per_pin,
-		       struct hdmi_eld *eld)
+		       struct hdmi_eld *eld,
+		       int repoll)
 {
 	struct hdmi_eld *pin_eld = &per_pin->sink_eld;
 	struct hdmi_spec *spec = codec->spec;
+	struct snd_jack *pcm_jack;
 	bool old_eld_valid = pin_eld->eld_valid;
 	bool eld_changed;
 	int pcm_idx;
 
+	if (eld->eld_valid) {
+		if (eld->eld_size <= 0 ||
+		    snd_hdmi_parse_eld(codec, &eld->info, eld->eld_buffer,
+				       eld->eld_size) < 0) {
+			eld->eld_valid = false;
+			if (repoll) {
+				schedule_delayed_work(&per_pin->work,
+						      msecs_to_jiffies(300));
+				return;
+			}
+		}
+	}
+
+	if (!eld->eld_valid || eld->eld_size <= 0) {
+		eld->eld_valid = false;
+		eld->eld_size = 0;
+	}
+
 	/* for monitor disconnection, save pcm_idx firstly */
 	pcm_idx = per_pin->pcm_idx;
+
+	/*
+	 * pcm_idx >=0 before update_eld() means it is in monitor
+	 * disconnected event. Jack must be fetched before update_eld().
+	 */
+	pcm_jack = pin_idx_to_pcm_jack(codec, per_pin);
+
 	if (spec->dyn_pcm_assign) {
 		if (eld->eld_valid) {
 			hdmi_attach_hda_pcm(spec, per_pin);
···
 	 */
 	if (pcm_idx == -1)
 		pcm_idx = per_pin->pcm_idx;
+	if (!pcm_jack)
+		pcm_jack = pin_idx_to_pcm_jack(codec, per_pin);
 
 	if (eld->eld_valid)
 		snd_hdmi_show_eld(codec, &eld->info);
···
 			       SNDRV_CTL_EVENT_MASK_VALUE |
 			       SNDRV_CTL_EVENT_MASK_INFO,
 			       &get_hdmi_pcm(spec, pcm_idx)->eld_ctl->id);
-	return eld_changed;
+
+	if (eld_changed && pcm_jack)
+		snd_jack_report(pcm_jack,
+				(eld->monitor_present && eld->eld_valid) ?
+				SND_JACK_AVOUT : 0);
 }
 
-static struct snd_jack *pin_idx_to_pcm_jack(struct hda_codec *codec,
-					    struct hdmi_spec_per_pin *per_pin)
-{
-	struct hdmi_spec *spec = codec->spec;
-	struct snd_jack *jack = NULL;
-	struct hda_jack_tbl *jack_tbl;
-
-	/* if !dyn_pcm_assign, get jack from hda_jack_tbl
-	 * in !dyn_pcm_assign case, spec->pcm_rec[].jack is not
-	 * NULL even after snd_hda_jack_tbl_clear() is called to
-	 * free snd_jack. This may cause access invalid memory
-	 * when calling snd_jack_report
-	 */
-	if (per_pin->pcm_idx >= 0 && spec->dyn_pcm_assign) {
-		jack = spec->pcm_rec[per_pin->pcm_idx].jack;
-	} else if (!spec->dyn_pcm_assign) {
-		/*
-		 * jack tbl doesn't support DP MST
-		 * DP MST will use dyn_pcm_assign,
-		 * so DP MST will never come here
-		 */
-		jack_tbl = snd_hda_jack_tbl_get_mst(codec, per_pin->pin_nid,
-						    per_pin->dev_id);
-		if (jack_tbl)
-			jack = jack_tbl->jack;
-	}
-	return jack;
-}
 /* update ELD and jack state via HD-audio verbs */
-static bool hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin,
+static void hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin,
 					 int repoll)
 {
-	struct hda_jack_tbl *jack;
 	struct hda_codec *codec = per_pin->codec;
 	struct hdmi_spec *spec = codec->spec;
 	struct hdmi_eld *eld = &spec->temp_eld;
···
 	 * the unsolicited response to avoid custom WARs.
 	 */
 	int present;
-	bool ret;
-	bool do_repoll = false;
-	struct snd_jack *pcm_jack = NULL;
+	int ret;
+
+	ret = snd_hda_power_up_pm(codec);
+	if (ret < 0 && pm_runtime_suspended(hda_codec_dev(codec)))
+		goto out;
 
 	present = snd_hda_jack_pin_sense(codec, pin_nid, dev_id);
 
···
 		if (spec->ops.pin_get_eld(codec, pin_nid, dev_id,
 					  eld->eld_buffer, &eld->eld_size) < 0)
 			eld->eld_valid = false;
-		else {
-			if (snd_hdmi_parse_eld(codec, &eld->info, eld->eld_buffer,
-						    eld->eld_size) < 0)
-				eld->eld_valid = false;
-		}
-		if (!eld->eld_valid && repoll)
-			do_repoll = true;
 	}
 
-	if (do_repoll) {
-		schedule_delayed_work(&per_pin->work, msecs_to_jiffies(300));
-	} else {
-		/*
-		 * pcm_idx >=0 before update_eld() means it is in monitor
-		 * disconnected event. Jack must be fetched before
-		 * update_eld().
-		 */
-		pcm_jack = pin_idx_to_pcm_jack(codec, per_pin);
-		update_eld(codec, per_pin, eld);
-		if (!pcm_jack)
-			pcm_jack = pin_idx_to_pcm_jack(codec, per_pin);
-	}
-
-	ret = !repoll || !eld->monitor_present || eld->eld_valid;
-
-	jack = snd_hda_jack_tbl_get_mst(codec, pin_nid, per_pin->dev_id);
-	if (jack) {
-		jack->block_report = !ret;
-		jack->pin_sense = (eld->monitor_present && eld->eld_valid) ?
-			AC_PINSENSE_PRESENCE : 0;
-
-		if (spec->dyn_pcm_assign && pcm_jack && !do_repoll) {
-			int state = 0;
-
-			if (jack->pin_sense & AC_PINSENSE_PRESENCE)
-				state = SND_JACK_AVOUT;
-			snd_jack_report(pcm_jack, state);
-		}
-
-		/*
-		 * snd_hda_jack_pin_sense() call at the beginning of this
-		 * function, updates jack->pins_sense and clears
-		 * jack->jack_dirty, therefore snd_hda_jack_report_sync() will
-		 * not override the jack->pin_sense.
-		 *
-		 * snd_hda_jack_report_sync() is superfluous for dyn_pcm_assign
-		 * case. The jack->pin_sense update was already performed, and
-		 * hda_jack->jack is NULL for dyn_pcm_assign.
-		 *
-		 * Don't call snd_hda_jack_report_sync() for
-		 * dyn_pcm_assign.
-		 */
-		ret = ret && !spec->dyn_pcm_assign;
-	}
+	update_eld(codec, per_pin, eld, repoll);
 	mutex_unlock(&per_pin->lock);
-	return ret;
+ out:
+	snd_hda_power_down_pm(codec);
 }
 
 /* update ELD and jack state via audio component */
···
 {
 	struct hdmi_spec *spec = codec->spec;
 	struct hdmi_eld *eld = &spec->temp_eld;
-	struct snd_jack *jack = NULL;
-	bool changed;
-	int size;
 
 	mutex_lock(&per_pin->lock);
 	eld->monitor_present = false;
-	size = snd_hdac_acomp_get_eld(&codec->core, per_pin->pin_nid,
+	eld->eld_size = snd_hdac_acomp_get_eld(&codec->core, per_pin->pin_nid,
 				      per_pin->dev_id, &eld->monitor_present,
 				      eld->eld_buffer, ELD_MAX_SIZE);
-	if (size > 0) {
-		size = min(size, ELD_MAX_SIZE);
-		if (snd_hdmi_parse_eld(codec, &eld->info,
-				       eld->eld_buffer, size) < 0)
-			size = -EINVAL;
-	}
-
-	if (size > 0) {
-		eld->eld_valid = true;
-		eld->eld_size = size;
-	} else {
-		eld->eld_valid = false;
-		eld->eld_size = 0;
-	}
-
-	/* pcm_idx >=0 before update_eld() means it is in monitor
-	 * disconnected event. Jack must be fetched before update_eld()
-	 */
-	jack = pin_idx_to_pcm_jack(codec, per_pin);
-	changed = update_eld(codec, per_pin, eld);
-	if (jack == NULL)
-		jack = pin_idx_to_pcm_jack(codec, per_pin);
-	if (changed && jack)
-		snd_jack_report(jack,
-				(eld->monitor_present && eld->eld_valid) ?
-				SND_JACK_AVOUT : 0);
+	eld->eld_valid = (eld->eld_size > 0);
+	update_eld(codec, per_pin, eld, 0);
 	mutex_unlock(&per_pin->lock);
 }
 
-static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll)
+static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll)
 {
 	struct hda_codec *codec = per_pin->codec;
-	int ret;
 
-	/* no temporary power up/down needed for component notifier */
-	if (!codec_has_acomp(codec)) {
-		ret = snd_hda_power_up_pm(codec);
-		if (ret < 0 && pm_runtime_suspended(hda_codec_dev(codec))) {
-			snd_hda_power_down_pm(codec);
-			return false;
-		}
-		ret = hdmi_present_sense_via_verbs(per_pin, repoll);
-		snd_hda_power_down_pm(codec);
-	} else {
+	if (!codec_has_acomp(codec))
+		hdmi_present_sense_via_verbs(per_pin, repoll);
+	else
 		sync_eld_via_acomp(codec, per_pin);
-		ret = false; /* don't call snd_hda_jack_report_sync() */
-	}
-
-	return ret;
 }
 
 static void hdmi_repoll_eld(struct work_struct *work)
···
 		per_pin->repoll_count = 0;
 
 	mutex_lock(&spec->pcm_lock);
-	if (hdmi_present_sense(per_pin, per_pin->repoll_count))
-		snd_hda_jack_report_sync(per_pin->codec);
+	hdmi_present_sense(per_pin, per_pin->repoll_count);
 	mutex_unlock(&spec->pcm_lock);
 }
 
···
 	pcm->jack = NULL;
 }
 
-static int add_hdmi_jack_kctl(struct hda_codec *codec,
-			       struct hdmi_spec *spec,
-			       int pcm_idx,
-			       const char *name)
+static int generic_hdmi_build_jack(struct hda_codec *codec, int pcm_idx)
 {
+	char hdmi_str[32] = "HDMI/DP";
+	struct hdmi_spec *spec = codec->spec;
+	struct hdmi_spec_per_pin *per_pin = get_pin(spec, pcm_idx);
 	struct snd_jack *jack;
+	int pcmdev = get_pcm_rec(spec, pcm_idx)->device;
 	int err;
 
-	err = snd_jack_new(codec->card, name, SND_JACK_AVOUT, &jack,
+	if (pcmdev > 0)
+		sprintf(hdmi_str + strlen(hdmi_str), ",pcm=%d", pcmdev);
+	if (!spec->dyn_pcm_assign &&
+	    !is_jack_detectable(codec, per_pin->pin_nid))
+		strncat(hdmi_str, " Phantom",
+			sizeof(hdmi_str) - strlen(hdmi_str) - 1);
+
+	err = snd_jack_new(codec->card, hdmi_str, SND_JACK_AVOUT, &jack,
 			   true, false);
 	if (err < 0)
 		return err;
···
 	spec->pcm_rec[pcm_idx].jack = jack;
 	jack->private_data = &spec->pcm_rec[pcm_idx];
 	jack->private_free = free_hdmi_jack_priv;
-	return 0;
-}
-
-static int generic_hdmi_build_jack(struct hda_codec *codec, int pcm_idx)
-{
-	char hdmi_str[32] = "HDMI/DP";
-	struct hdmi_spec *spec = codec->spec;
-	struct hdmi_spec_per_pin *per_pin;
-	struct hda_jack_tbl *jack;
-	int pcmdev = get_pcm_rec(spec, pcm_idx)->device;
-	bool phantom_jack;
-	int ret;
-
-	if (pcmdev > 0)
-		sprintf(hdmi_str + strlen(hdmi_str), ",pcm=%d", pcmdev);
-
-	if (spec->dyn_pcm_assign)
-		return add_hdmi_jack_kctl(codec, spec, pcm_idx, hdmi_str);
-
-	/* for !dyn_pcm_assign, we still use hda_jack for compatibility */
-	/* if !dyn_pcm_assign, it must be non-MST mode.
-	 * This means pcms and pins are statically mapped.
-	 * And pcm_idx is pin_idx.
-	 */
-	per_pin = get_pin(spec, pcm_idx);
-	phantom_jack = !is_jack_detectable(codec, per_pin->pin_nid);
-	if (phantom_jack)
-		strncat(hdmi_str, " Phantom",
-			sizeof(hdmi_str) - strlen(hdmi_str) - 1);
-	ret = snd_hda_jack_add_kctl_mst(codec, per_pin->pin_nid,
-					per_pin->dev_id, hdmi_str, phantom_jack,
-					0, NULL);
-	if (ret < 0)
-		return ret;
-	jack = snd_hda_jack_tbl_get_mst(codec, per_pin->pin_nid,
-					per_pin->dev_id);
-	if (jack == NULL)
-		return 0;
-	/* assign jack->jack to pcm_rec[].jack to
-	 * align with dyn_pcm_assign mode
-	 */
-	spec->pcm_rec[pcm_idx].jack = jack->jack;
 	return 0;
 }
 
···
 	int pin_idx;
 
 	mutex_lock(&spec->bind_lock);
-	spec->use_jack_detect = !codec->jackpoll_interval;
 	for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) {
 		struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx);
 		hda_nid_t pin_nid = per_pin->pin_nid;
···
 		hdmi_init_pin(codec, pin_nid);
 		if (codec_has_acomp(codec))
 			continue;
-		if (spec->use_jack_detect)
-			snd_hda_jack_detect_enable(codec, pin_nid, dev_id);
-		else
-			snd_hda_jack_detect_enable_callback_mst(codec, pin_nid,
-								dev_id,
-								jack_callback);
+		snd_hda_jack_detect_enable_callback_mst(codec, pin_nid, dev_id,
+							jack_callback);
 	}
 	mutex_unlock(&spec->bind_lock);
 	return 0;
···
 		unsigned int val = use_acomp ? 0 : (AC_USRSP_EN | tbl->tag);
 		snd_hda_codec_write_cache(codec, nid, 0,
 					  AC_VERB_SET_UNSOLICITED_ENABLE, val);
-	} else {
-		/* if no jack entry was defined beforehand, create a new one
-		 * at need (i.e. only when notifier is cleared)
-		 */
-		if (!use_acomp)
-			snd_hda_jack_detect_enable(codec, nid, dev_id);
 	}
 }
 
···
 	spec->use_acomp_notifier = use_acomp;
 	spec->codec->relaxed_resume = use_acomp;
 	/* reprogram each jack detection logic depending on the notifier */
-	if (spec->use_jack_detect) {
-		for (i = 0; i < spec->num_pins; i++)
-			reprogram_jack_detect(spec->codec,
-					      get_pin(spec, i)->pin_nid,
-					      get_pin(spec, i)->dev_id,
-					      use_acomp);
-	}
+	for (i = 0; i < spec->num_pins; i++)
+		reprogram_jack_detect(spec->codec,
+				      get_pin(spec, i)->pin_nid,
+				      get_pin(spec, i)->dev_id,
+				      use_acomp);
 	mutex_unlock(&spec->bind_lock);
 }
 

sound/pci/hda/patch_realtek.c

···
 	unsigned int done_hp_init:1;
 	unsigned int no_shutup_pins:1;
 	unsigned int ultra_low_power:1;
+	unsigned int has_hs_key:1;
 
 	/* for PLL fix */
 	hda_nid_t pll_nid;
···
 	case 0x10ec0215:
 	case 0x10ec0233:
 	case 0x10ec0235:
+	case 0x10ec0236:
 	case 0x10ec0255:
+	case 0x10ec0256:
 	case 0x10ec0257:
 	case 0x10ec0282:
 	case 0x10ec0283:
···
 	case 0x10ec0298:
 	case 0x10ec0289:
 	case 0x10ec0300:
-		alc_update_coef_idx(codec, 0x10, 1<<9, 0);
-		break;
-	case 0x10ec0236:
-	case 0x10ec0256:
-		alc_write_coef_idx(codec, 0x36, 0x5757);
 		alc_update_coef_idx(codec, 0x10, 1<<9, 0);
 		break;
 	case 0x10ec0275:
···
 	return alc_parse_auto_config(codec, alc269_ignore, ssids);
 }
 
+static const struct hda_jack_keymap alc_headset_btn_keymap[] = {
+	{ SND_JACK_BTN_0, KEY_PLAYPAUSE },
+	{ SND_JACK_BTN_1, KEY_VOICECOMMAND },
+	{ SND_JACK_BTN_2, KEY_VOLUMEUP },
+	{ SND_JACK_BTN_3, KEY_VOLUMEDOWN },
+	{}
+};
+
+static void alc_headset_btn_callback(struct hda_codec *codec,
+				     struct hda_jack_callback *jack)
+{
+	int report = 0;
+
+	if (jack->unsol_res & (7 << 13))
+		report |= SND_JACK_BTN_0;
+
+	if (jack->unsol_res  & (1 << 16 | 3 << 8))
+		report |= SND_JACK_BTN_1;
+
+	/* Volume up key */
+	if (jack->unsol_res & (7 << 23))
+		report |= SND_JACK_BTN_2;
+
+	/* Volume down key */
+	if (jack->unsol_res & (7 << 10))
+		report |= SND_JACK_BTN_3;
+
+	jack->jack->button_state = report;
+}
+
+static void alc_disable_headset_jack_key(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+
+	if (!spec->has_hs_key)
+		return;
+
+	switch (codec->core.vendor_id) {
+	case 0x10ec0215:
+	case 0x10ec0225:
+	case 0x10ec0285:
+	case 0x10ec0295:
+	case 0x10ec0289:
+	case 0x10ec0299:
+		alc_write_coef_idx(codec, 0x48, 0x0);
+		alc_update_coef_idx(codec, 0x49, 0x0045, 0x0);
+		alc_update_coef_idx(codec, 0x44, 0x0045 << 8, 0x0);
+		break;
+	case 0x10ec0236:
+	case 0x10ec0256:
+		alc_write_coef_idx(codec, 0x48, 0x0);
+		alc_update_coef_idx(codec, 0x49, 0x0045, 0x0);
+		break;
+	}
+}
+
+static void alc_enable_headset_jack_key(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+
+	if (!spec->has_hs_key)
+		return;
+
+	switch (codec->core.vendor_id) {
+	case 0x10ec0215:
+	case 0x10ec0225:
+	case 0x10ec0285:
+	case 0x10ec0295:
+	case 0x10ec0289:
+	case 0x10ec0299:
+		alc_write_coef_idx(codec, 0x48, 0xd011);
+		alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045);
+		alc_update_coef_idx(codec, 0x44, 0x007f << 8, 0x0045 << 8);
+		break;
+	case 0x10ec0236:
+	case 0x10ec0256:
+		alc_write_coef_idx(codec, 0x48, 0xd011);
+		alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045);
+		break;
+	}
+}
+
+static void alc_fixup_headset_jack(struct hda_codec *codec,
+				    const struct hda_fixup *fix, int action)
+{
+	struct alc_spec *spec = codec->spec;
+
+	switch (action) {
+	case HDA_FIXUP_ACT_PRE_PROBE:
+		spec->has_hs_key = 1;
+		snd_hda_jack_detect_enable_callback(codec, 0x55,
+						    alc_headset_btn_callback);
+		snd_hda_jack_add_kctl(codec, 0x55, "Headset Jack", false,
+				      SND_JACK_HEADSET, alc_headset_btn_keymap);
+		break;
+	case HDA_FIXUP_ACT_INIT:
+		alc_enable_headset_jack_key(codec);
+		break;
+	}
+}
+
 static void alc269vb_toggle_power_output(struct hda_codec *codec, int power_up)
 {
 	alc_update_coef_idx(codec, 0x04, 1 << 11, power_up ? (1 << 11) : 0);
···
 	alc_update_coefex_idx(codec, 0x57, 0x04, 0x0007, 0x4); /* Hight power */
 	alc_update_coefex_idx(codec, 0x53, 0x02, 0x8000, 1 << 15); /* Clear bit */
 	alc_update_coefex_idx(codec, 0x53, 0x02, 0x8000, 0 << 15);
-	alc_update_coef_idx(codec, 0x36, 1 << 13, 1 << 5); /* Switch pcbeep path to Line in path*/
+	/*
+	 * Expose headphone mic (or possibly Line In on some machines) instead
+	 * of PC Beep on 1Ah, and disable 1Ah loopback for all outputs. See
+	 * Documentation/sound/hd-audio/realtek-pc-beep.rst for details of
+	 * this register.
+	 */
+	alc_write_coef_idx(codec, 0x36, 0x5757);
 }
 
 static void alc256_shutup(struct hda_codec *codec)
···
 
 	if (!hp_pin)
 		hp_pin = 0x21;
+
+	alc_disable_headset_jack_key(codec);
 	/* 3k pull low control for Headset jack. */
 	alc_update_coef_idx(codec, 0x4a, 0, 3 << 10);
 
···
 		alc_update_coef_idx(codec, 0x4a, 3<<4, 2<<4);
 		msleep(30);
 	}
+
+	alc_update_coef_idx(codec, 0x4a, 3 << 10, 0);
+	alc_enable_headset_jack_key(codec);
 }
 
 static void alc_default_init(struct hda_codec *codec)
···
 				const struct hda_fixup *fix, int action)
 {
 	alc_fixup_hp_gpio_led(codec, action, 0x08, 0x10);
+}
+
+static void alc285_fixup_hp_gpio_led(struct hda_codec *codec,
+				const struct hda_fixup *fix, int action)
+{
+	alc_fixup_hp_gpio_led(codec, action, 0x04, 0x00);
 }
 
 static void alc286_fixup_hp_gpio_led(struct hda_codec *codec,
···
 	}
 }
 
-static void alc256_fixup_dell_xps_13_headphone_noise2(struct hda_codec *codec,
-						      const struct hda_fixup *fix,
-						      int action)
-{
-	if (action != HDA_FIXUP_ACT_PRE_PROBE)
-		return;
-
-	snd_hda_codec_amp_stereo(codec, 0x1a, HDA_INPUT, 0, HDA_AMP_VOLMASK, 1);
-	snd_hda_override_wcaps(codec, 0x1a, get_wcaps(codec, 0x1a) & ~AC_WCAP_IN_AMP);
-}
-
 static void alc269_fixup_limit_int_mic_boost(struct hda_codec *codec,
 					     const struct hda_fixup *fix,
 					     int action)
···
 	snd_hda_override_wcaps(codec, 0x03, 0);
 }
 
-static const struct hda_jack_keymap alc_headset_btn_keymap[] = {
-	{ SND_JACK_BTN_0, KEY_PLAYPAUSE },
-	{ SND_JACK_BTN_1, KEY_VOICECOMMAND },
-	{ SND_JACK_BTN_2, KEY_VOLUMEUP },
-	{ SND_JACK_BTN_3, KEY_VOLUMEDOWN },
-	{}
-};
-
-static void alc_headset_btn_callback(struct hda_codec *codec,
-				     struct hda_jack_callback *jack)
-{
-	int report = 0;
-
-	if (jack->unsol_res & (7 << 13))
-		report |= SND_JACK_BTN_0;
-
-	if (jack->unsol_res  & (1 << 16 | 3 << 8))
-		report |= SND_JACK_BTN_1;
-
-	/* Volume up key */
-	if (jack->unsol_res & (7 << 23))
-		report |= SND_JACK_BTN_2;
-
-	/* Volume down key */
-	if (jack->unsol_res & (7 << 10))
-		report |= SND_JACK_BTN_3;
-
-	jack->jack->button_state = report;
-}
-
-static void alc_fixup_headset_jack(struct hda_codec *codec,
-				    const struct hda_fixup *fix, int action)
-{
-
-	switch (action) {
-	case HDA_FIXUP_ACT_PRE_PROBE:
-		snd_hda_jack_detect_enable_callback(codec, 0x55,
-						    alc_headset_btn_callback);
-		snd_hda_jack_add_kctl(codec, 0x55, "Headset Jack", false,
-				      SND_JACK_HEADSET, alc_headset_btn_keymap);
-		break;
-	case HDA_FIXUP_ACT_INIT:
-		switch (codec->core.vendor_id) {
-		case 0x10ec0215:
-		case 0x10ec0225:
-		case 0x10ec0285:
-		case 0x10ec0295:
-		case 0x10ec0289:
-		case 0x10ec0299:
-			alc_write_coef_idx(codec, 0x48, 0xd011);
-			alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045);
-			alc_update_coef_idx(codec, 0x44, 0x007f << 8, 0x0045 << 8);
-			break;
-		case 0x10ec0236:
-		case 0x10ec0256:
-			alc_write_coef_idx(codec, 0x48, 0xd011);
-			alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045);
-			break;
-		}
-		break;
-	}
-}
-
 static void alc295_fixup_chromebook(struct hda_codec *codec,
 				    const struct hda_fixup *fix, int action)
 {
···
 	ALC298_FIXUP_DELL1_MIC_NO_PRESENCE,
 	ALC298_FIXUP_DELL_AIO_MIC_NO_PRESENCE,
 	ALC275_FIXUP_DELL_XPS,
-	ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE,
-	ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2,
 	ALC293_FIXUP_LENOVO_SPK_NOISE,
 	ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY,
 	ALC255_FIXUP_DELL_SPK_NOISE,
···
 	ALC294_FIXUP_ASUS_DUAL_SPK,
 	ALC285_FIXUP_THINKPAD_HEADSET_JACK,
 	ALC294_FIXUP_ASUS_HPE,
+	ALC285_FIXUP_HP_GPIO_LED,
 };
 
 static const struct hda_fixup alc269_fixups[] = {
···
 			{}
 		}
 	},
-	[ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE] = {
-		.type = HDA_FIXUP_VERBS,
-		.v.verbs = (const struct hda_verb[]) {
-			/* Disable pass-through path for FRONT 14h */
-			{0x20, AC_VERB_SET_COEF_INDEX, 0x36},
-			{0x20, AC_VERB_SET_PROC_COEF, 0x1737},
-			{}
-		},
-		.chained = true,
-		.chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE
-	},
-	[ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2] = {
-		.type = HDA_FIXUP_FUNC,
-		.v.func = alc256_fixup_dell_xps_13_headphone_noise2,
-		.chained = true,
-		.chain_id = ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE
-	},
 	[ALC293_FIXUP_LENOVO_SPK_NOISE] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc_fixup_disable_aamix,
···
 		.chained = true,
 		.chain_id = ALC294_FIXUP_ASUS_HEADSET_MIC
 	},
+	[ALC285_FIXUP_HP_GPIO_LED] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc285_fixup_hp_gpio_led,
+	},
 };
 
 static const struct snd_pci_quirk alc269_fixup_tbl[] = {
···
 	SND_PCI_QUIRK(0x1028, 0x06de, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK),
 	SND_PCI_QUIRK(0x1028, 0x06df, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK),
 	SND_PCI_QUIRK(0x1028, 0x06e0, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK),
-	SND_PCI_QUIRK(0x1028, 0x0704, "Dell XPS 13 9350", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2),
 	SND_PCI_QUIRK(0x1028, 0x0706, "Dell Inspiron 7559", ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER),
 	SND_PCI_QUIRK(0x1028, 0x0725, "Dell Inspiron 3162", ALC255_FIXUP_DELL_SPK_NOISE),
 	SND_PCI_QUIRK(0x1028, 0x0738, "Dell Precision 5820", ALC269_FIXUP_NO_SHUTUP),
-	SND_PCI_QUIRK(0x1028, 0x075b, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2),
 	SND_PCI_QUIRK(0x1028, 0x075c, "Dell XPS 27 7760", ALC298_FIXUP_SPK_VOLUME),
 	SND_PCI_QUIRK(0x1028, 0x075d, "Dell AIO", ALC298_FIXUP_SPK_VOLUME),
 	SND_PCI_QUIRK(0x1028, 0x07b0, "Dell Precision 7520", ALC295_FIXUP_DISABLE_DAC3),
 	SND_PCI_QUIRK(0x1028, 0x0798, "Dell Inspiron 17 7000 Gaming", ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER),
 	SND_PCI_QUIRK(0x1028, 0x080c, "Dell WYSE", ALC225_FIXUP_DELL_WYSE_MIC_NO_PRESENCE),
-	SND_PCI_QUIRK(0x1028, 0x082a, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2),
 	SND_PCI_QUIRK(0x1028, 0x084b, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB),
 	SND_PCI_QUIRK(0x1028, 0x084e, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB),
 	SND_PCI_QUIRK(0x1028, 0x0871, "Dell Precision 3630", ALC255_FIXUP_DELL_HEADSET_MIC),
···
 	SND_PCI_QUIRK(0x103c, 0x83b9, "HP Spectre x360", ALC269_FIXUP_HP_MUTE_LED_MIC3),
 	SND_PCI_QUIRK(0x103c, 0x8497, "HP Envy x360", ALC269_FIXUP_HP_MUTE_LED_MIC3),
 	SND_PCI_QUIRK(0x103c, 0x84e7, "HP Pavilion 15", ALC269_FIXUP_HP_MUTE_LED_MIC3),
+	SND_PCI_QUIRK(0x103c, 0x8736, "HP", ALC285_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC),
 	SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300),
 	SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
···
 	{.id = ALC298_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "alc298-dell1"},
 	{.id = ALC298_FIXUP_DELL_AIO_MIC_NO_PRESENCE, .name = "alc298-dell-aio"},
 	{.id = ALC275_FIXUP_DELL_XPS, .name = "alc275-dell-xps"},
-	{.id = ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE, .name = "alc256-dell-xps13"},
 	{.id = ALC293_FIXUP_LENOVO_SPK_NOISE, .name = "lenovo-spk-noise"},
 	{.id = ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY, .name = "lenovo-hotkey"},
 	{.id = ALC255_FIXUP_DELL_SPK_NOISE, .name = "dell-spk-noise"},

sound/pci/korg1212/korg1212.c

···
 #if K1212_DEBUG_LEVEL > 0
 #define K1212_DEBUG_PRINTK(fmt,args...)	printk(KERN_DEBUG fmt,##args)
 #else
-#define K1212_DEBUG_PRINTK(fmt,...)
+#define K1212_DEBUG_PRINTK(fmt,...)	do { } while (0)
 #endif
 #if K1212_DEBUG_LEVEL > 1
 #define K1212_DEBUG_PRINTK_VERBOSE(fmt,args...)	printk(KERN_DEBUG fmt,##args)

sound/pci/rme9652/hdsp.c

···
 				return;
 			}
 		} else {
-			int err = -EINVAL;
+			int err;
+
 			err = hdsp_request_fw_loader(hdsp);
 			if (err < 0) {
 				snd_iprintf(buffer,

sound/pci/via82xx.c

···
 {
 	unsigned int i, idx, ofs, rest;
 	struct via82xx *chip = snd_pcm_substream_chip(substream);
+	__le32 *pgtbl;
 
 	if (dev->table.area == NULL) {
 		/* the start of each lists must be aligned to 8 bytes,
···
 	/* fill the entries */
 	idx = 0;
 	ofs = 0;
+	pgtbl = (__le32 *)dev->table.area;
 	for (i = 0; i < periods; i++) {
 		rest = fragsize;
 		/* fill descriptors for a period.
···
 				return -EINVAL;
 			}
 			addr = snd_pcm_sgbuf_get_addr(substream, ofs);
-			((u32 *)dev->table.area)[idx << 1] = cpu_to_le32(addr);
+			pgtbl[idx << 1] = cpu_to_le32(addr);
 			r = snd_pcm_sgbuf_get_chunk_size(substream, ofs, rest);
 			rest -= r;
 			if (! rest) {
···
 				"tbl %d: at %d  size %d (rest %d)\n",
 				idx, ofs, r, rest);
 			*/
-			((u32 *)dev->table.area)[(idx<<1) + 1] = cpu_to_le32(r | flag);
+			pgtbl[(idx<<1) + 1] = cpu_to_le32(r | flag);
 			dev->idx_table[idx].offset = ofs;
 			dev->idx_table[idx].size = r;
 			ofs += r;

sound/pci/via82xx_modem.c

···
 {
 	unsigned int i, idx, ofs, rest;
 	struct via82xx_modem *chip = snd_pcm_substream_chip(substream);
+	__le32 *pgtbl;
 
 	if (dev->table.area == NULL) {
 		/* the start of each lists must be aligned to 8 bytes,
···
 	/* fill the entries */
 	idx = 0;
 	ofs = 0;
+	pgtbl = (__le32 *)dev->table.area;
 	for (i = 0; i < periods; i++) {
 		rest = fragsize;
 		/* fill descriptors for a period.
···
 				return -EINVAL;
 			}
 			addr = snd_pcm_sgbuf_get_addr(substream, ofs);
-			((u32 *)dev->table.area)[idx << 1] = cpu_to_le32(addr);
+			pgtbl[idx << 1] = cpu_to_le32(addr);
 			r = PAGE_SIZE - (ofs % PAGE_SIZE);
 			if (rest < r)
 				r = rest;
···
 				"tbl %d: at %d  size %d (rest %d)\n",
 				idx, ofs, r, rest);
 			*/
-			((u32 *)dev->table.area)[(idx<<1) + 1] = cpu_to_le32(r | flag);
+			pgtbl[(idx<<1) + 1] = cpu_to_le32(r | flag);
 			dev->idx_table[idx].offset = ofs;
 			dev->idx_table[idx].size = r;
 			ofs += r;

sound/ppc/keywest.c

···
 static int keywest_attach_adapter(struct i2c_adapter *adapter)
 {
 	struct i2c_board_info info;
+	struct i2c_client *client;
 
 	if (! keywest_ctx)
 		return -EINVAL;
···
 	memset(&info, 0, sizeof(struct i2c_board_info));
 	strlcpy(info.type, "keywest", I2C_NAME_SIZE);
 	info.addr = keywest_ctx->addr;
-	keywest_ctx->client = i2c_new_device(adapter, &info);
-	if (!keywest_ctx->client)
-		return -ENODEV;
+	client = i2c_new_client_device(adapter, &info);
+	if (IS_ERR(client))
+		return PTR_ERR(client);
+	keywest_ctx->client = client;
+
 	/*
 	 * We know the driver is already loaded, so the device should be
 	 * already bound. If not it means binding failed, and then there

sound/soc/amd/Kconfig

···
 	depends on X86 && PCI
 	help
 	 This option enables ACP v3.x I2S support on AMD platform
+
+config SND_SOC_AMD_RV_RT5682_MACH
+	tristate "AMD RV support for RT5682"
+	select SND_SOC_RT5682
+	select SND_SOC_MAX98357A
+	select SND_SOC_CROS_EC_CODEC
+	select I2C_CROS_EC_TUNNEL
+	depends on SND_SOC_AMD_ACP3x && I2C && CROS_EC
+	help
+	 This option enables machine driver for RT5682 and MAX9835.

sound/soc/amd/Makefile

···
 acp_audio_dma-objs := acp-pcm-dma.o
 snd-soc-acp-da7219mx98357-mach-objs := acp-da7219-max98357a.o
 snd-soc-acp-rt5645-mach-objs := acp-rt5645.o
+snd-soc-acp-rt5682-mach-objs := acp3x-rt5682-max9836.o
 
 obj-$(CONFIG_SND_SOC_AMD_ACP) += acp_audio_dma.o
 obj-$(CONFIG_SND_SOC_AMD_CZ_DA7219MX98357_MACH) += snd-soc-acp-da7219mx98357-mach.o
 obj-$(CONFIG_SND_SOC_AMD_CZ_RT5645_MACH) += snd-soc-acp-rt5645-mach.o
 obj-$(CONFIG_SND_SOC_AMD_ACP3x) += raven/
+obj-$(CONFIG_SND_SOC_AMD_RV_RT5682_MACH) += snd-soc-acp-rt5682-mach.o

sound/soc/amd/acp-da7219-max98357a.c

···
 {
 	int ret;
 	struct snd_soc_card *card = rtd->card;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	struct snd_soc_component *component = codec_dai->component;
 
 	dev_info(rtd->dev, "codec dai name = %s\n", codec_dai->name);

sound/soc/amd/acp-rt5645.c

···
 {
 	int ret = 0;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 
 	ret = snd_soc_dai_set_pll(codec_dai, 0, RT5645_PLL1_S_MCLK,
 				  CZ_PLAT_CLK, params_rate(params) * 512);
···
 	struct snd_soc_card *card;
 	struct snd_soc_component *codec;
 
-	codec = rtd->codec_dai->component;
+	codec = asoc_rtd_to_codec(rtd, 0)->component;
 	card = rtd->card;
 
 	ret = snd_soc_card_jack_new(card, "Headset Jack",

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

···
+// SPDX-License-Identifier: GPL-2.0+
+//
+// Machine driver for AMD ACP Audio engine using DA7219 & MAX98357 codec.
+//
+//Copyright 2016 Advanced Micro Devices, Inc.
+
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc-dapm.h>
+#include <sound/jack.h>
+#include <linux/clk.h>
+#include <linux/gpio.h>
+#include <linux/gpio/consumer.h>
+#include <linux/module.h>
+#include <linux/i2c.h>
+#include <linux/input.h>
+#include <linux/io.h>
+#include <linux/acpi.h>
+
+#include "raven/acp3x.h"
+#include "../codecs/rt5682.h"
+
+#define PCO_PLAT_CLK 48000000
+#define RT5682_PLL_FREQ (48000 * 512)
+#define DUAL_CHANNEL		2
+
+static struct snd_soc_jack pco_jack;
+static struct clk *rt5682_dai_wclk;
+static struct clk *rt5682_dai_bclk;
+static struct gpio_desc *dmic_sel;
+
+static int acp3x_5682_init(struct snd_soc_pcm_runtime *rtd)
+{
+	int ret;
+	struct snd_soc_card *card = rtd->card;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+	struct snd_soc_component *component = codec_dai->component;
+
+	dev_info(rtd->dev, "codec dai name = %s\n", codec_dai->name);
+
+	/* set rt5682 dai fmt */
+	ret =  snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S
+			| SND_SOC_DAIFMT_NB_NF
+			| SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0) {
+		dev_err(rtd->card->dev,
+				"Failed to set rt5682 dai fmt: %d\n", ret);
+		return ret;
+	}
+
+	/* set codec PLL */
+	ret = snd_soc_dai_set_pll(codec_dai, RT5682_PLL2, RT5682_PLL2_S_MCLK,
+				  PCO_PLAT_CLK, RT5682_PLL_FREQ);
+	if (ret < 0) {
+		dev_err(rtd->dev, "can't set rt5682 PLL: %d\n", ret);
+		return ret;
+	}
+
+	/* Set codec sysclk */
+	ret = snd_soc_dai_set_sysclk(codec_dai, RT5682_SCLK_S_PLL2,
+			RT5682_PLL_FREQ, SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		dev_err(rtd->dev,
+			"Failed to set rt5682 SYSCLK: %d\n", ret);
+		return ret;
+	}
+
+	/* Set tdm/i2s1 master bclk ratio */
+	ret = snd_soc_dai_set_bclk_ratio(codec_dai, 64);
+	if (ret < 0) {
+		dev_err(rtd->dev,
+			"Failed to set rt5682 tdm bclk ratio: %d\n", ret);
+		return ret;
+	}
+
+	rt5682_dai_wclk = clk_get(component->dev, "rt5682-dai-wclk");
+	rt5682_dai_bclk = clk_get(component->dev, "rt5682-dai-bclk");
+
+	ret = snd_soc_card_jack_new(card, "Headset Jack",
+				SND_JACK_HEADSET | SND_JACK_LINEOUT |
+				SND_JACK_BTN_0 | SND_JACK_BTN_1 |
+				SND_JACK_BTN_2 | SND_JACK_BTN_3,
+				&pco_jack, NULL, 0);
+	if (ret) {
+		dev_err(card->dev, "HP jack creation failed %d\n", ret);
+		return ret;
+	}
+
+	snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_0, KEY_PLAYPAUSE);
+	snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_1, KEY_VOLUMEUP);
+	snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN);
+	snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_3, KEY_VOICECOMMAND);
+
+	ret = snd_soc_component_set_jack(component, &pco_jack, NULL);
+	if (ret) {
+		dev_err(rtd->dev, "Headset Jack call-back failed: %d\n", ret);
+		return ret;
+	}
+
+	return ret;
+}
+
+static int rt5682_clk_enable(struct snd_pcm_substream *substream)
+{
+	int ret = 0;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+
+	/* RT5682 will support only 48K output with 48M mclk */
+	clk_set_rate(rt5682_dai_wclk, 48000);
+	clk_set_rate(rt5682_dai_bclk, 48000 * 64);
+	ret = clk_prepare_enable(rt5682_dai_wclk);
+	if (ret < 0) {
+		dev_err(rtd->dev, "can't enable wclk %d\n", ret);
+		return ret;
+	}
+
+	return ret;
+}
+
+static void rt5682_clk_disable(void)
+{
+	clk_disable_unprepare(rt5682_dai_wclk);
+}
+
+static const unsigned int channels[] = {
+	DUAL_CHANNEL,
+};
+
+static const unsigned int rates[] = {
+	48000,
+};
+
+static const struct snd_pcm_hw_constraint_list constraints_rates = {
+	.count = ARRAY_SIZE(rates),
+	.list  = rates,
+	.mask = 0,
+};
+
+static const struct snd_pcm_hw_constraint_list constraints_channels = {
+	.count = ARRAY_SIZE(channels),
+	.list = channels,
+	.mask = 0,
+};
+
+static int acp3x_5682_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_card *card = rtd->card;
+	struct acp3x_platform_info *machine = snd_soc_card_get_drvdata(card);
+
+	machine->play_i2s_instance = I2S_SP_INSTANCE;
+	machine->cap_i2s_instance = I2S_SP_INSTANCE;
+
+	runtime->hw.channels_max = DUAL_CHANNEL;
+	snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+				   &constraints_channels);
+	snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+				   &constraints_rates);
+	return rt5682_clk_enable(substream);
+}
+
+static int acp3x_max_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_card *card = rtd->card;
+	struct acp3x_platform_info *machine = snd_soc_card_get_drvdata(card);
+
+	machine->play_i2s_instance = I2S_BT_INSTANCE;
+
+	runtime->hw.channels_max = DUAL_CHANNEL;
+	snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+				   &constraints_channels);
+	snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+				   &constraints_rates);
+	return rt5682_clk_enable(substream);
+}
+
+static int acp3x_ec_dmic0_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_card *card = rtd->card;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+	struct acp3x_platform_info *machine = snd_soc_card_get_drvdata(card);
+
+	machine->cap_i2s_instance = I2S_BT_INSTANCE;
+	snd_soc_dai_set_bclk_ratio(codec_dai, 64);
+	if (dmic_sel)
+		gpiod_set_value(dmic_sel, 0);
+
+	return rt5682_clk_enable(substream);
+}
+
+static int acp3x_ec_dmic1_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_card *card = rtd->card;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+	struct acp3x_platform_info *machine = snd_soc_card_get_drvdata(card);
+
+	machine->cap_i2s_instance = I2S_BT_INSTANCE;
+	snd_soc_dai_set_bclk_ratio(codec_dai, 64);
+	if (dmic_sel)
+		gpiod_set_value(dmic_sel, 1);
+
+	return rt5682_clk_enable(substream);
+}
+
+static void rt5682_shutdown(struct snd_pcm_substream *substream)
+{
+	rt5682_clk_disable();
+}
+
+static const struct snd_soc_ops acp3x_5682_ops = {
+	.startup = acp3x_5682_startup,
+	.shutdown = rt5682_shutdown,
+};
+
+static const struct snd_soc_ops acp3x_max_play_ops = {
+	.startup = acp3x_max_startup,
+	.shutdown = rt5682_shutdown,
+};
+
+static const struct snd_soc_ops acp3x_ec_cap0_ops = {
+	.startup = acp3x_ec_dmic0_startup,
+	.shutdown = rt5682_shutdown,
+};
+
+static const struct snd_soc_ops acp3x_ec_cap1_ops = {
+	.startup = acp3x_ec_dmic1_startup,
+	.shutdown = rt5682_shutdown,
+};
+
+SND_SOC_DAILINK_DEF(acp3x_i2s,
+	DAILINK_COMP_ARRAY(COMP_CPU("acp3x_i2s_playcap.0")));
+SND_SOC_DAILINK_DEF(acp3x_bt,
+	DAILINK_COMP_ARRAY(COMP_CPU("acp3x_i2s_playcap.2")));
+
+SND_SOC_DAILINK_DEF(rt5682,
+	DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10EC5682:00", "rt5682-aif1")));
+SND_SOC_DAILINK_DEF(max,
+	DAILINK_COMP_ARRAY(COMP_CODEC("MX98357A:00", "HiFi")));
+SND_SOC_DAILINK_DEF(cros_ec,
+	DAILINK_COMP_ARRAY(COMP_CODEC("GOOG0013:00", "EC Codec I2S RX")));
+
+SND_SOC_DAILINK_DEF(platform,
+	DAILINK_COMP_ARRAY(COMP_PLATFORM("acp3x_rv_i2s_dma.0")));
+
+static struct snd_soc_dai_link acp3x_dai_5682_98357[] = {
+	{
+		.name = "acp3x-5682-play",
+		.stream_name = "Playback",
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+				| SND_SOC_DAIFMT_CBM_CFM,
+		.init = acp3x_5682_init,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+		.ops = &acp3x_5682_ops,
+		SND_SOC_DAILINK_REG(acp3x_i2s, rt5682, platform),
+	},
+	{
+		.name = "acp3x-max98357-play",
+		.stream_name = "HiFi Playback",
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+				| SND_SOC_DAIFMT_CBM_CFM,
+		.dpcm_playback = 1,
+		.ops = &acp3x_max_play_ops,
+		SND_SOC_DAILINK_REG(acp3x_bt, max, platform),
+	},
+	{
+		.name = "acp3x-ec-dmic0-capture",
+		.stream_name = "Capture DMIC0",
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+				| SND_SOC_DAIFMT_CBS_CFS,
+		.dpcm_capture = 1,
+		.ops = &acp3x_ec_cap0_ops,
+		SND_SOC_DAILINK_REG(acp3x_bt, cros_ec, platform),
+	},
+	{
+		.name = "acp3x-ec-dmic1-capture",
+		.stream_name = "Capture DMIC1",
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+				| SND_SOC_DAIFMT_CBS_CFS,
+		.dpcm_capture = 1,
+		.ops = &acp3x_ec_cap1_ops,
+		SND_SOC_DAILINK_REG(acp3x_bt, cros_ec, platform),
+	},
+};
+
+static const struct snd_soc_dapm_widget acp3x_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_SPK("Spk", NULL),
+	SND_SOC_DAPM_MIC("Headset Mic", NULL),
+};
+
+static const struct snd_soc_dapm_route acp3x_audio_route[] = {
+	{"Headphone Jack", NULL, "HPOL"},
+	{"Headphone Jack", NULL, "HPOR"},
+	{"IN1P", NULL, "Headset Mic"},
+	{"Spk", NULL, "Speaker"},
+};
+
+static const struct snd_kcontrol_new acp3x_mc_controls[] = {
+	SOC_DAPM_PIN_SWITCH("Headphone Jack"),
+	SOC_DAPM_PIN_SWITCH("Spk"),
+	SOC_DAPM_PIN_SWITCH("Headset Mic"),
+};
+
+static struct snd_soc_card acp3x_card = {
+	.name = "acp3xalc5682m98357",
+	.owner = THIS_MODULE,
+	.dai_link = acp3x_dai_5682_98357,
+	.num_links = ARRAY_SIZE(acp3x_dai_5682_98357),
+	.dapm_widgets = acp3x_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(acp3x_widgets),
+	.dapm_routes = acp3x_audio_route,
+	.num_dapm_routes = ARRAY_SIZE(acp3x_audio_route),
+	.controls = acp3x_mc_controls,
+	.num_controls = ARRAY_SIZE(acp3x_mc_controls),
+};
+
+static int acp3x_probe(struct platform_device *pdev)
+{
+	int ret;
+	struct snd_soc_card *card;
+	struct acp3x_platform_info *machine;
+
+	machine = devm_kzalloc(&pdev->dev, sizeof(*machine), GFP_KERNEL);
+	if (!machine)
+		return -ENOMEM;
+
+	card = &acp3x_card;
+	acp3x_card.dev = &pdev->dev;
+	platform_set_drvdata(pdev, card);
+	snd_soc_card_set_drvdata(card, machine);
+
+	dmic_sel = devm_gpiod_get(&pdev->dev, "dmic", GPIOD_OUT_LOW);
+	if (IS_ERR(dmic_sel)) {
+		dev_err(&pdev->dev, "DMIC gpio failed err=%ld\n",
+			PTR_ERR(dmic_sel));
+		return PTR_ERR(dmic_sel);
+	}
+
+	ret = devm_snd_soc_register_card(&pdev->dev, &acp3x_card);
+	if (ret) {
+		dev_err(&pdev->dev,
+				"devm_snd_soc_register_card(%s) failed: %d\n",
+				acp3x_card.name, ret);
+		return ret;
+	}
+	return 0;
+}
+
+static const struct acpi_device_id acp3x_audio_acpi_match[] = {
+	{ "AMDI5682", 0 },
+	{},
+};
+MODULE_DEVICE_TABLE(acpi, acp3x_audio_acpi_match);
+
+static struct platform_driver acp3x_audio = {
+	.driver = {
+		.name = "acp3x-alc5682-max98357",
+		.acpi_match_table = ACPI_PTR(acp3x_audio_acpi_match),
+		.pm = &snd_soc_pm_ops,
+	},
+	.probe = acp3x_probe,
+};
+
+module_platform_driver(acp3x_audio);
+
+MODULE_AUTHOR("[e-mail]");
+MODULE_DESCRIPTION("ALC5682 & MAX98357 audio support");
+MODULE_LICENSE("GPL v2");

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

···
 		u32 tx_mask, u32 rx_mask, int slots, int slot_width)
 {
 	struct i2s_dev_data *adata;
-	u32 val, reg_val, frmt_reg, frm_len;
+	u32 frm_len;
 	u16 slot_len;
 
 	adata = snd_soc_dai_get_drvdata(cpu_dai);
···
 	default:
 		return -EINVAL;
 	}
-
-	/* Enable I2S/BT channels TDM, respective TX/RX frame lengths.*/
-
 	frm_len = FRM_LEN | (slots << 15) | (slot_len << 18);
-	if (adata->substream_type == SNDRV_PCM_STREAM_PLAYBACK) {
-		switch (adata->i2s_instance) {
-		case I2S_BT_INSTANCE:
-			reg_val = mmACP_BTTDM_ITER;
-			frmt_reg = mmACP_BTTDM_TXFRMT;
-			break;
-		case I2S_SP_INSTANCE:
-		default:
-			reg_val = mmACP_I2STDM_ITER;
-			frmt_reg = mmACP_I2STDM_TXFRMT;
-		}
-	} else {
-		switch (adata->i2s_instance) {
-		case I2S_BT_INSTANCE:
-			reg_val = mmACP_BTTDM_IRER;
-			frmt_reg = mmACP_BTTDM_RXFRMT;
-			break;
-		case I2S_SP_INSTANCE:
-		default:
-			reg_val = mmACP_I2STDM_IRER;
-			frmt_reg = mmACP_I2STDM_RXFRMT;
-		}
-	}
-	val = rv_readl(adata->acp3x_base + reg_val);
-	rv_writel(val | 0x2, adata->acp3x_base + reg_val);
-	rv_writel(frm_len, adata->acp3x_base + frmt_reg);
 	adata->tdm_fmt = frm_len;
 	return 0;
 }
···
 	struct snd_soc_pcm_runtime *prtd;
 	struct snd_soc_card *card;
 	struct acp3x_platform_info *pinfo;
+	struct i2s_dev_data *adata;
 	u32 val;
-	u32 reg_val;
+	u32 reg_val, frmt_reg;
 
 	prtd = substream->private_data;
 	rtd = substream->runtime->private_data;
 	card = prtd->card;
+	adata = snd_soc_dai_get_drvdata(dai);
 	pinfo = snd_soc_card_get_drvdata(card);
 	if (pinfo) {
 		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
···
 		switch (rtd->i2s_instance) {
 		case I2S_BT_INSTANCE:
 			reg_val = mmACP_BTTDM_ITER;
+			frmt_reg = mmACP_BTTDM_TXFRMT;
 			break;
 		case I2S_SP_INSTANCE:
 		default:
 			reg_val = mmACP_I2STDM_ITER;
+			frmt_reg = mmACP_I2STDM_TXFRMT;
 		}
 	} else {
 		switch (rtd->i2s_instance) {
 		case I2S_BT_INSTANCE:
 			reg_val = mmACP_BTTDM_IRER;
+			frmt_reg = mmACP_BTTDM_RXFRMT;
 			break;
 		case I2S_SP_INSTANCE:
 		default:
 			reg_val = mmACP_I2STDM_IRER;
+			frmt_reg = mmACP_I2STDM_RXFRMT;
 		}
+	}
+	if (adata->tdm_mode) {
+		val = rv_readl(rtd->acp3x_base + reg_val);
+		rv_writel(val | 0x2, rtd->acp3x_base + reg_val);
+		rv_writel(adata->tdm_fmt, rtd->acp3x_base + frmt_reg);
 	}
 	val = rv_readl(rtd->acp3x_base + reg_val);
 	val = val | (rtd->xfer_resolution  << 3);

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

···
 			reg_val = mmACP_I2STDM_ITER;
 			frmt_val = mmACP_I2STDM_TXFRMT;
 		}
-	rv_writel((rtd->xfer_resolution  << 3), rtd->acp3x_base + reg_val);
+		rv_writel((rtd->xfer_resolution  << 3),
+			  rtd->acp3x_base + reg_val);
 	}
 	if (adata->capture_stream && adata->capture_stream->runtime) {
 		struct i2s_stream_instance *rtd =
···
 			reg_val = mmACP_I2STDM_IRER;
 			frmt_val = mmACP_I2STDM_RXFRMT;
 		}
-	rv_writel((rtd->xfer_resolution  << 3), rtd->acp3x_base + reg_val);
+		rv_writel((rtd->xfer_resolution  << 3),
+			  rtd->acp3x_base + reg_val);
 	}
 	if (adata->tdm_mode == TDM_ENABLE) {
 		rv_writel(adata->tdm_fmt, adata->acp3x_base + frmt_val);

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

···
 	timeout = 0;
 	while (++timeout < 500) {
 		val = rv_readl(acp3x_base + mmACP_PGFSM_STATUS);
-		if (!val)
+		if (!val) {
+			/* Set PME_EN as after ACP power On,
+			 * PME_EN gets cleared
+			 */
+			rv_writel(0x1, acp3x_base + mmACP_PME_EN);
 			return 0;
+		}
 		udelay(1);
 	}
 	return -ETIMEDOUT;

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

···
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct atmel_pcm_dma_params *prtd;
 
-	prtd = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+	prtd = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream);
 
 	if (ssc_sr & prtd->mask->ssc_error) {
 		if (snd_pcm_running(substream))
···
 	struct ssc_device *ssc;
 	int ret;
 
-	prtd = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+	prtd = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream);
 	ssc = prtd->ssc;
 
 	ret = snd_hwparams_to_dma_slave_config(substream, params, slave_config);

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

···
 	snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
 	runtime->dma_bytes = params_buffer_bytes(params);
 
-	prtd->params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+	prtd->params = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream);
 	prtd->params->dma_intr_handler = atmel_pcm_dma_irq;
 
 	prtd->dma_buffer = runtime->dma_addr;

sound/soc/atmel/atmel_wm8904.c

···
 		struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	int ret;
 
 	ret = snd_soc_dai_set_pll(codec_dai, WM8904_FLL_MCLK, WM8904_FLL_MCLK,

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

···
 	unsigned int				frame_length;
 	int					tdm_slots;
 	int					channels;
-	int					gclk_use:1;
-	int					gclk_running:1;
-	int					tx_rdy:1;
-	int					rx_rdy:1;
+	unsigned int				gclk_use:1;
+	unsigned int				gclk_running:1;
+	unsigned int				tx_rdy:1;
+	unsigned int				rx_rdy:1;
 };
 
 static irqreturn_t mchp_i2s_mcc_interrupt(int irq, void *dev_id)

sound/soc/atmel/mikroe-proto.c

···
 static int snd_proto_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_card *card = rtd->card;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 
 	/* Set proto sysclk */
 	int ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL,

sound/soc/atmel/sam9g20_wm8731.c

···
  */
 static int at91sam9g20ek_wm8731_init(struct snd_soc_pcm_runtime *rtd)
 {
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	struct device *dev = rtd->dev;
 	int ret;
 

sound/soc/atmel/sam9x5_wm8731.c

···
  */
 static int sam9x5_wm8731_init(struct snd_soc_pcm_runtime *rtd)
 {
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	struct device *dev = rtd->dev;
 	int ret;
 

sound/soc/au1x/db1200.c

···
 static int db1200_i2s_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 
 	/* WM8731 has its own 12MHz crystal */
 	snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL,

sound/soc/au1x/dbdma2.c

···
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	int stype = substream->stream, *dmaids;
 
-	dmaids = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+	dmaids = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream);
 	if (!dmaids)
 		return -ENODEV;	/* whoa, has ordering changed? */
 

sound/soc/au1x/dma.c

···
 	int *dmaids, s = substream->stream;
 	char *name;
 
-	dmaids = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+	dmaids = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream);
 	if (!dmaids)
 		return -ENODEV;	/* whoa, has ordering changed? */
 

sound/soc/au1x/psc-ac97.c

···
 static inline struct au1xpsc_audio_data *ac97_to_pscdata(struct snd_ac97 *x)
 {
 	struct snd_soc_card *c = x->bus->card->private_data;
-	return snd_soc_dai_get_drvdata(c->rtd->cpu_dai);
+	return snd_soc_dai_get_drvdata(c->asoc_rtd_to_cpu(rtd, 0));
 }
 
 #else

sound/soc/bcm/Kconfig

···
 	  Cygnus chips (bcm958300, bcm958305, bcm911360)
 
 	  If you don't know what to do here, say N.
+
+config SND_BCM63XX_I2S_WHISTLER
+	tristate "SoC Audio support for the Broadcom BCM63XX I2S module"
+	select REGMAP_MMIO
+	help
+	  Say Y if you want to add support for ASoC audio on Broadcom
+	  DSL/PON chips (bcm63158, bcm63178)
+
+	  If you don't know what to do here, say N

sound/soc/bcm/Makefile

···
 
 obj-$(CONFIG_SND_SOC_CYGNUS) += snd-soc-cygnus.o
 
+# BCM63XX Platform Support
+snd-soc-63xx-objs := bcm63xx-i2s-whistler.o bcm63xx-pcm-whistler.o
+
+obj-$(CONFIG_SND_BCM63XX_I2S_WHISTLER) += snd-soc-63xx.o

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

···
+// SPDX-License-Identifier: GPL-2.0-or-later
+// linux/sound/bcm/bcm63xx-i2s-whistler.c
+// BCM63xx whistler i2s driver
+// Copyright (c) 2020 Broadcom Corporation
+// Author: Kevin-Ke Li <[e-mail]>
+
+#include <linux/clk.h>
+#include <linux/dma-mapping.h>
+#include <linux/io.h>
+#include <linux/module.h>
+#include <linux/regmap.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include "bcm63xx-i2s.h"
+
+#define DRV_NAME "brcm-i2s"
+
+static bool brcm_i2s_wr_reg(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case I2S_TX_CFG ... I2S_TX_DESC_IFF_LEN:
+	case I2S_TX_CFG_2 ... I2S_RX_DESC_IFF_LEN:
+	case I2S_RX_CFG_2 ... I2S_REG_MAX:
+		return true;
+	default:
+		return false;
+	}
+}
+
+static bool brcm_i2s_rd_reg(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case I2S_TX_CFG ... I2S_REG_MAX:
+		return true;
+	default:
+		return false;
+	}
+}
+
+static bool brcm_i2s_volatile_reg(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case I2S_TX_CFG:
+	case I2S_TX_IRQ_CTL:
+	case I2S_TX_DESC_IFF_ADDR:
+	case I2S_TX_DESC_IFF_LEN:
+	case I2S_TX_DESC_OFF_ADDR:
+	case I2S_TX_DESC_OFF_LEN:
+	case I2S_TX_CFG_2:
+	case I2S_RX_CFG:
+	case I2S_RX_IRQ_CTL:
+	case I2S_RX_DESC_OFF_ADDR:
+	case I2S_RX_DESC_OFF_LEN:
+	case I2S_RX_DESC_IFF_LEN:
+	case I2S_RX_DESC_IFF_ADDR:
+	case I2S_RX_CFG_2:
+		return true;
+	default:
+		return false;
+	}
+}
+
+static const struct regmap_config brcm_i2s_regmap_config = {
+	.reg_bits = 32,
+	.reg_stride = 4,
+	.val_bits = 32,
+	.max_register = I2S_REG_MAX,
+	.writeable_reg = brcm_i2s_wr_reg,
+	.readable_reg = brcm_i2s_rd_reg,
+	.volatile_reg = brcm_i2s_volatile_reg,
+	.cache_type = REGCACHE_FLAT,
+};
+
+static int bcm63xx_i2s_hw_params(struct snd_pcm_substream *substream,
+				 struct snd_pcm_hw_params *params,
+				 struct snd_soc_dai *dai)
+{
+	int ret = 0;
+	struct bcm_i2s_priv *i2s_priv = snd_soc_dai_get_drvdata(dai);
+
+	ret = clk_set_rate(i2s_priv->i2s_clk, params_rate(params));
+	if (ret < 0)
+		dev_err(i2s_priv->dev,
+			"Can't set sample rate, err: %d\n", ret);
+
+	return ret;
+}
+
+static int bcm63xx_i2s_startup(struct snd_pcm_substream *substream,
+			       struct snd_soc_dai *dai)
+{
+	unsigned int slavemode;
+	struct bcm_i2s_priv *i2s_priv = snd_soc_dai_get_drvdata(dai);
+	struct regmap *regmap_i2s = i2s_priv->regmap_i2s;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		regmap_update_bits(regmap_i2s, I2S_TX_CFG,
+				   I2S_TX_OUT_R | I2S_TX_DATA_ALIGNMENT |
+				   I2S_TX_DATA_ENABLE | I2S_TX_CLOCK_ENABLE,
+				   I2S_TX_OUT_R | I2S_TX_DATA_ALIGNMENT |
+				   I2S_TX_DATA_ENABLE | I2S_TX_CLOCK_ENABLE);
+		regmap_write(regmap_i2s, I2S_TX_IRQ_CTL, 0);
+		regmap_write(regmap_i2s, I2S_TX_IRQ_IFF_THLD, 0);
+		regmap_write(regmap_i2s, I2S_TX_IRQ_OFF_THLD, 1);
+
+		/* TX and RX block each have an independent bit to indicate
+		 * if it is generating the clock for the I2S bus. The bus
+		 * clocks need to be generated from either the TX or RX block,
+		 * but not both
+		 */
+		regmap_read(regmap_i2s, I2S_RX_CFG_2, &slavemode);
+		if (slavemode & I2S_RX_SLAVE_MODE_MASK)
+			regmap_update_bits(regmap_i2s, I2S_TX_CFG_2,
+					   I2S_TX_SLAVE_MODE_MASK,
+					   I2S_TX_MASTER_MODE);
+		else
+			regmap_update_bits(regmap_i2s, I2S_TX_CFG_2,
+					   I2S_TX_SLAVE_MODE_MASK,
+					   I2S_TX_SLAVE_MODE);
+	} else {
+		regmap_update_bits(regmap_i2s, I2S_RX_CFG,
+				   I2S_RX_IN_R | I2S_RX_DATA_ALIGNMENT |
+				   I2S_RX_CLOCK_ENABLE,
+				   I2S_RX_IN_R | I2S_RX_DATA_ALIGNMENT |
+				   I2S_RX_CLOCK_ENABLE);
+		regmap_write(regmap_i2s, I2S_RX_IRQ_CTL, 0);
+		regmap_write(regmap_i2s, I2S_RX_IRQ_IFF_THLD, 0);
+		regmap_write(regmap_i2s, I2S_RX_IRQ_OFF_THLD, 1);
+
+		regmap_read(regmap_i2s, I2S_TX_CFG_2, &slavemode);
+		if (slavemode & I2S_TX_SLAVE_MODE_MASK)
+			regmap_update_bits(regmap_i2s, I2S_RX_CFG_2,
+					   I2S_RX_SLAVE_MODE_MASK, 0);
+		else
+			regmap_update_bits(regmap_i2s, I2S_RX_CFG_2,
+					   I2S_RX_SLAVE_MODE_MASK,
+					   I2S_RX_SLAVE_MODE);
+	}
+	return 0;
+}
+
+static void bcm63xx_i2s_shutdown(struct snd_pcm_substream *substream,
+				struct snd_soc_dai *dai)
+{
+	unsigned int enabled, slavemode;
+	struct bcm_i2s_priv *i2s_priv = snd_soc_dai_get_drvdata(dai);
+	struct regmap *regmap_i2s = i2s_priv->regmap_i2s;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		regmap_update_bits(regmap_i2s, I2S_TX_CFG,
+				   I2S_TX_OUT_R | I2S_TX_DATA_ALIGNMENT |
+				   I2S_TX_DATA_ENABLE | I2S_TX_CLOCK_ENABLE, 0);
+		regmap_write(regmap_i2s, I2S_TX_IRQ_CTL, 1);
+		regmap_write(regmap_i2s, I2S_TX_IRQ_IFF_THLD, 4);
+		regmap_write(regmap_i2s, I2S_TX_IRQ_OFF_THLD, 4);
+
+		regmap_read(regmap_i2s, I2S_TX_CFG_2, &slavemode);
+		slavemode = slavemode & I2S_TX_SLAVE_MODE_MASK;
+		if (!slavemode) {
+			regmap_read(regmap_i2s, I2S_RX_CFG, &enabled);
+			enabled = enabled & I2S_RX_ENABLE_MASK;
+			if (enabled)
+				regmap_update_bits(regmap_i2s, I2S_RX_CFG_2,
+						   I2S_RX_SLAVE_MODE_MASK,
+						   I2S_RX_MASTER_MODE);
+		}
+		regmap_update_bits(regmap_i2s, I2S_TX_CFG_2,
+				   I2S_TX_SLAVE_MODE_MASK,
+				   I2S_TX_SLAVE_MODE);
+	} else {
+		regmap_update_bits(regmap_i2s, I2S_RX_CFG,
+				   I2S_RX_IN_R | I2S_RX_DATA_ALIGNMENT |
+				   I2S_RX_CLOCK_ENABLE, 0);
+		regmap_write(regmap_i2s, I2S_RX_IRQ_CTL, 1);
+		regmap_write(regmap_i2s, I2S_RX_IRQ_IFF_THLD, 4);
+		regmap_write(regmap_i2s, I2S_RX_IRQ_OFF_THLD, 4);
+
+		regmap_read(regmap_i2s, I2S_RX_CFG_2, &slavemode);
+		slavemode = slavemode & I2S_RX_SLAVE_MODE_MASK;
+		if (!slavemode) {
+			regmap_read(regmap_i2s, I2S_TX_CFG, &enabled);
+			enabled = enabled & I2S_TX_ENABLE_MASK;
+			if (enabled)
+				regmap_update_bits(regmap_i2s, I2S_TX_CFG_2,
+						   I2S_TX_SLAVE_MODE_MASK,
+						   I2S_TX_MASTER_MODE);
+		}
+
+		regmap_update_bits(regmap_i2s, I2S_RX_CFG_2,
+				   I2S_RX_SLAVE_MODE_MASK, I2S_RX_SLAVE_MODE);
+	}
+}
+
+static const struct snd_soc_dai_ops bcm63xx_i2s_dai_ops = {
+	.startup = bcm63xx_i2s_startup,
+	.shutdown = bcm63xx_i2s_shutdown,
+	.hw_params = bcm63xx_i2s_hw_params,
+};
+
+static struct snd_soc_dai_driver bcm63xx_i2s_dai = {
+	.name = DRV_NAME,
+	.playback = {
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_8000_192000,
+		.formats = SNDRV_PCM_FMTBIT_S32_LE,
+	},
+	.capture = {
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_8000_192000,
+		.formats = SNDRV_PCM_FMTBIT_S32_LE,
+	},
+	.ops = &bcm63xx_i2s_dai_ops,
+	.symmetric_rates = 1,
+	.symmetric_channels = 1,
+};
+
+static const struct snd_soc_component_driver bcm63xx_i2s_component = {
+	.name = "bcm63xx",
+};
+
+static int bcm63xx_i2s_dev_probe(struct platform_device *pdev)
+{
+	int ret = 0;
+	void __iomem *regs;
+	struct resource *r_mem, *region;
+	struct bcm_i2s_priv *i2s_priv;
+	struct regmap *regmap_i2s;
+	struct clk *i2s_clk;
+
+	i2s_priv = devm_kzalloc(&pdev->dev, sizeof(*i2s_priv), GFP_KERNEL);
+	if (!i2s_priv)
+		return -ENOMEM;
+
+	i2s_clk = devm_clk_get(&pdev->dev, "i2sclk");
+	if (IS_ERR(i2s_clk)) {
+		dev_err(&pdev->dev, "%s: cannot get a brcm clock: %ld\n",
+					__func__, PTR_ERR(i2s_clk));
+		return PTR_ERR(i2s_clk);
+	}
+
+	r_mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	if (!r_mem) {
+		dev_err(&pdev->dev, "Unable to get register resource.\n");
+		return -ENODEV;
+	}
+
+	region = devm_request_mem_region(&pdev->dev, r_mem->start,
+					resource_size(r_mem), DRV_NAME);
+	if (!region) {
+		dev_err(&pdev->dev, "Memory region already claimed\n");
+		return -EBUSY;
+	}
+
+	regs = devm_ioremap_resource(&pdev->dev, r_mem);
+	if (IS_ERR(regs)) {
+		ret = PTR_ERR(regs);
+		return ret;
+	}
+
+	regmap_i2s = devm_regmap_init_mmio(&pdev->dev,
+					regs, &brcm_i2s_regmap_config);
+	if (IS_ERR(regmap_i2s))
+		return PTR_ERR(regmap_i2s);
+
+	regmap_update_bits(regmap_i2s, I2S_MISC_CFG,
+			   I2S_PAD_LVL_LOOP_DIS_MASK,
+			   I2S_PAD_LVL_LOOP_DIS_ENABLE);
+
+	ret = devm_snd_soc_register_component(&pdev->dev,
+					      &bcm63xx_i2s_component,
+					      &bcm63xx_i2s_dai, 1);
+	if (ret) {
+		dev_err(&pdev->dev, "failed to register the dai\n");
+		return ret;
+	}
+
+	i2s_priv->dev = &pdev->dev;
+	i2s_priv->i2s_clk = i2s_clk;
+	i2s_priv->regmap_i2s = regmap_i2s;
+	dev_set_drvdata(&pdev->dev, i2s_priv);
+
+	ret = bcm63xx_soc_platform_probe(pdev, i2s_priv);
+	if (ret)
+		dev_err(&pdev->dev, "failed to register the pcm\n");
+
+	return ret;
+}
+
+static int bcm63xx_i2s_dev_remove(struct platform_device *pdev)
+{
+	bcm63xx_soc_platform_remove(pdev);
+	return 0;
+}
+
+#ifdef CONFIG_OF
+static const struct of_device_id snd_soc_bcm_audio_match[] = {
+	{.compatible = "brcm,bcm63xx-i2s"},
+	{ }
+};
+#endif
+
+static struct platform_driver bcm63xx_i2s_driver = {
+	.driver = {
+		.name = DRV_NAME,
+		.of_match_table = of_match_ptr(snd_soc_bcm_audio_match),
+	},
+	.probe = bcm63xx_i2s_dev_probe,
+	.remove = bcm63xx_i2s_dev_remove,
+};
+
+module_platform_driver(bcm63xx_i2s_driver);
+
+MODULE_AUTHOR("Kevin,Li <[e-mail]>");
+MODULE_DESCRIPTION("Broadcom DSL XPON ASOC I2S Interface");
+MODULE_LICENSE("GPL v2");

sound/soc/bcm/bcm63xx-i2s.h

···
+// SPDX-License-Identifier: GPL-2.0-or-later
+// linux/sound/soc/bcm/bcm63xx-i2s.h
+// Copyright (c) 2020 Broadcom Corporation
+// Author: Kevin-Ke Li <[e-mail]>
+
+#ifndef __BCM63XX_I2S_H
+#define __BCM63XX_I2S_H
+
+#define I2S_DESC_FIFO_DEPTH		8
+#define I2S_MISC_CFG			(0x003C)
+#define I2S_PAD_LVL_LOOP_DIS_MASK	(1 << 2)
+#define I2S_PAD_LVL_LOOP_DIS_ENABLE	I2S_PAD_LVL_LOOP_DIS_MASK
+
+#define I2S_TX_ENABLE_MASK		(1 << 31)
+#define I2S_TX_ENABLE			I2S_TX_ENABLE_MASK
+#define I2S_TX_OUT_R			(1 << 19)
+#define I2S_TX_DATA_ALIGNMENT		(1 << 2)
+#define I2S_TX_DATA_ENABLE		(1 << 1)
+#define I2S_TX_CLOCK_ENABLE		(1 << 0)
+
+#define I2S_TX_DESC_OFF_LEVEL_SHIFT	12
+#define I2S_TX_DESC_OFF_LEVEL_MASK	(0x0F << I2S_TX_DESC_OFF_LEVEL_SHIFT)
+#define I2S_TX_DESC_IFF_LEVEL_SHIFT	8
+#define I2S_TX_DESC_IFF_LEVEL_MASK	(0x0F << I2S_TX_DESC_IFF_LEVEL_SHIFT)
+#define I2S_TX_DESC_OFF_INTR_EN_MSK	(1 << 1)
+#define I2S_TX_DESC_OFF_INTR_EN	I2S_TX_DESC_OFF_INTR_EN_MSK
+
+#define I2S_TX_CFG			(0x0000)
+#define I2S_TX_IRQ_CTL			(0x0004)
+#define I2S_TX_IRQ_EN			(0x0008)
+#define I2S_TX_IRQ_IFF_THLD		(0x000c)
+#define I2S_TX_IRQ_OFF_THLD		(0x0010)
+#define I2S_TX_DESC_IFF_ADDR		(0x0014)
+#define I2S_TX_DESC_IFF_LEN		(0x0018)
+#define I2S_TX_DESC_OFF_ADDR		(0x001C)
+#define I2S_TX_DESC_OFF_LEN		(0x0020)
+#define I2S_TX_CFG_2			(0x0024)
+#define I2S_TX_SLAVE_MODE_SHIFT	13
+#define I2S_TX_SLAVE_MODE_MASK		(1 << I2S_TX_SLAVE_MODE_SHIFT)
+#define I2S_TX_SLAVE_MODE		I2S_TX_SLAVE_MODE_MASK
+#define I2S_TX_MASTER_MODE		0
+#define I2S_TX_INTR_MASK		0x0F
+
+#define I2S_RX_ENABLE_MASK		(1 << 31)
+#define I2S_RX_ENABLE			I2S_RX_ENABLE_MASK
+#define I2S_RX_IN_R			(1 << 19)
+#define I2S_RX_DATA_ALIGNMENT		(1 << 2)
+#define I2S_RX_CLOCK_ENABLE		(1 << 0)
+
+#define I2S_RX_DESC_OFF_LEVEL_SHIFT	12
+#define I2S_RX_DESC_OFF_LEVEL_MASK	(0x0F << I2S_RX_DESC_OFF_LEVEL_SHIFT)
+#define I2S_RX_DESC_IFF_LEVEL_SHIFT	8
+#define I2S_RX_DESC_IFF_LEVEL_MASK	(0x0F << I2S_RX_DESC_IFF_LEVEL_SHIFT)
+#define I2S_RX_DESC_OFF_INTR_EN_MSK	(1 << 1)
+#define I2S_RX_DESC_OFF_INTR_EN	I2S_RX_DESC_OFF_INTR_EN_MSK
+
+#define I2S_RX_CFG			(0x0040) /* 20c0 */
+#define I2S_RX_IRQ_CTL			(0x0044)
+#define I2S_RX_IRQ_EN			(0x0048)
+#define I2S_RX_IRQ_IFF_THLD		(0x004C)
+#define I2S_RX_IRQ_OFF_THLD		(0x0050)
+#define I2S_RX_DESC_IFF_ADDR		(0x0054)
+#define I2S_RX_DESC_IFF_LEN		(0x0058)
+#define I2S_RX_DESC_OFF_ADDR		(0x005C)
+#define I2S_RX_DESC_OFF_LEN		(0x0060)
+#define I2S_RX_CFG_2			(0x0064)
+#define I2S_RX_SLAVE_MODE_SHIFT	13
+#define I2S_RX_SLAVE_MODE_MASK		(1 << I2S_RX_SLAVE_MODE_SHIFT)
+#define I2S_RX_SLAVE_MODE		I2S_RX_SLAVE_MODE_MASK
+#define I2S_RX_MASTER_MODE		0
+#define I2S_RX_INTR_MASK		0x0F
+
+#define I2S_REG_MAX			0x007C
+
+struct bcm_i2s_priv {
+	struct device *dev;
+	struct resource *r_irq;
+	struct regmap *regmap_i2s;
+	struct clk *i2s_clk;
+	struct snd_pcm_substream	*play_substream;
+	struct snd_pcm_substream	*capture_substream;
+	struct i2s_dma_desc *play_dma_desc;
+	struct i2s_dma_desc *capture_dma_desc;
+};
+
+extern int bcm63xx_soc_platform_probe(struct platform_device *pdev,
+				      struct bcm_i2s_priv *i2s_priv);
+extern int bcm63xx_soc_platform_remove(struct platform_device *pdev);
+
+#endif

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

···
+// SPDX-License-Identifier: GPL-2.0-or-later
+// linux/sound/bcm/bcm63xx-pcm-whistler.c
+// BCM63xx whistler pcm interface
+// Copyright (c) 2020 Broadcom Corporation
+// Author: Kevin-Ke Li <[e-mail]>
+
+#include <linux/dma-mapping.h>
+#include <linux/io.h>
+#include <linux/module.h>
+#include <sound/pcm_params.h>
+#include <linux/regmap.h>
+#include <linux/of_device.h>
+#include <sound/soc.h>
+#include "bcm63xx-i2s.h"
+
+
+struct i2s_dma_desc {
+	unsigned char *dma_area;
+	dma_addr_t dma_addr;
+	unsigned int dma_len;
+};
+
+struct bcm63xx_runtime_data {
+	int dma_len;
+	dma_addr_t dma_addr;
+	dma_addr_t dma_addr_next;
+};
+
+static const struct snd_pcm_hardware bcm63xx_pcm_hardware = {
+	.info = SNDRV_PCM_INFO_MMAP |
+		SNDRV_PCM_INFO_MMAP_VALID |
+		SNDRV_PCM_INFO_INTERLEAVED |
+		SNDRV_PCM_INFO_PAUSE |
+		SNDRV_PCM_INFO_RESUME,
+	.formats = SNDRV_PCM_FMTBIT_S32_LE, /* support S32 only */
+	.period_bytes_max = 8192 - 32,
+	.periods_min = 1,
+	.periods_max = PAGE_SIZE/sizeof(struct i2s_dma_desc),
+	.buffer_bytes_max = 128 * 1024,
+	.fifo_size = 32,
+};
+
+static int bcm63xx_pcm_hw_params(struct snd_soc_component *component,
+				 struct snd_pcm_substream *substream,
+				 struct snd_pcm_hw_params *params)
+{
+	struct i2s_dma_desc *dma_desc;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+
+	snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+	runtime->dma_bytes = params_buffer_bytes(params);
+
+	dma_desc = kzalloc(sizeof(*dma_desc), GFP_NOWAIT);
+	if (!dma_desc)
+		return -ENOMEM;
+
+	snd_soc_dai_set_dma_data(asoc_rtd_to_cpu(rtd, 0), substream, dma_desc);
+
+	return 0;
+}
+
+static int bcm63xx_pcm_hw_free(struct snd_soc_component *component,
+			struct snd_pcm_substream *substream)
+{
+	struct i2s_dma_desc	*dma_desc;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+
+	dma_desc = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream);
+	kfree(dma_desc);
+	snd_pcm_set_runtime_buffer(substream, NULL);
+
+	return 0;
+}
+
+static int bcm63xx_pcm_trigger(struct snd_soc_component *component,
+			       struct snd_pcm_substream *substream, int cmd)
+{
+	int ret = 0;
+	struct snd_soc_pcm_runtime *rtd;
+	struct bcm_i2s_priv *i2s_priv;
+	struct regmap   *regmap_i2s;
+
+	rtd = substream->private_data;
+	i2s_priv = dev_get_drvdata(asoc_rtd_to_cpu(rtd, 0)->dev);
+	regmap_i2s = i2s_priv->regmap_i2s;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		switch (cmd) {
+		case SNDRV_PCM_TRIGGER_START:
+			regmap_update_bits(regmap_i2s,
+					   I2S_TX_IRQ_EN,
+					   I2S_TX_DESC_OFF_INTR_EN,
+					   I2S_TX_DESC_OFF_INTR_EN);
+			regmap_update_bits(regmap_i2s,
+					   I2S_TX_CFG,
+					   I2S_TX_ENABLE_MASK,
+					   I2S_TX_ENABLE);
+			break;
+		case SNDRV_PCM_TRIGGER_STOP:
+		case SNDRV_PCM_TRIGGER_SUSPEND:
+		case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+			regmap_write(regmap_i2s,
+				     I2S_TX_IRQ_EN,
+				     0);
+			regmap_update_bits(regmap_i2s,
+					   I2S_TX_CFG,
+					   I2S_TX_ENABLE_MASK,
+					   0);
+			break;
+		default:
+			ret = -EINVAL;
+		}
+	} else {
+		switch (cmd) {
+		case SNDRV_PCM_TRIGGER_START:
+			regmap_update_bits(regmap_i2s,
+					   I2S_RX_IRQ_EN,
+					   I2S_RX_DESC_OFF_INTR_EN_MSK,
+					   I2S_RX_DESC_OFF_INTR_EN);
+			regmap_update_bits(regmap_i2s,
+					   I2S_RX_CFG,
+					   I2S_RX_ENABLE_MASK,
+					   I2S_RX_ENABLE);
+			break;
+		case SNDRV_PCM_TRIGGER_STOP:
+		case SNDRV_PCM_TRIGGER_SUSPEND:
+		case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+			regmap_update_bits(regmap_i2s,
+					   I2S_RX_IRQ_EN,
+					   I2S_RX_DESC_OFF_INTR_EN_MSK,
+					   0);
+			regmap_update_bits(regmap_i2s,
+					   I2S_RX_CFG,
+					   I2S_RX_ENABLE_MASK,
+					   0);
+			break;
+		default:
+			ret = -EINVAL;
+		}
+	}
+	return ret;
+}
+
+static int bcm63xx_pcm_prepare(struct snd_soc_component *component,
+			struct snd_pcm_substream *substream)
+{
+	struct i2s_dma_desc	*dma_desc;
+	struct regmap		*regmap_i2s;
+	struct bcm_i2s_priv	*i2s_priv;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	uint32_t regaddr_desclen, regaddr_descaddr;
+
+	dma_desc = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream);
+	dma_desc->dma_len  = snd_pcm_lib_period_bytes(substream);
+	dma_desc->dma_addr = runtime->dma_addr;
+	dma_desc->dma_area = runtime->dma_area;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		regaddr_desclen = I2S_TX_DESC_IFF_LEN;
+		regaddr_descaddr = I2S_TX_DESC_IFF_ADDR;
+	} else {
+		regaddr_desclen = I2S_RX_DESC_IFF_LEN;
+		regaddr_descaddr = I2S_RX_DESC_IFF_ADDR;
+	}
+
+	i2s_priv = dev_get_drvdata(asoc_rtd_to_cpu(rtd, 0)->dev);
+	regmap_i2s = i2s_priv->regmap_i2s;
+
+	regmap_write(regmap_i2s, regaddr_desclen, dma_desc->dma_len);
+	regmap_write(regmap_i2s, regaddr_descaddr, dma_desc->dma_addr);
+
+	return 0;
+}
+
+static snd_pcm_uframes_t
+bcm63xx_pcm_pointer(struct snd_soc_component *component,
+		struct snd_pcm_substream *substream)
+{
+	snd_pcm_uframes_t x;
+	struct bcm63xx_runtime_data *prtd = substream->runtime->private_data;
+
+	if ((void *)prtd->dma_addr_next == NULL)
+		prtd->dma_addr_next = substream->runtime->dma_addr;
+
+	x = bytes_to_frames(substream->runtime,
+		prtd->dma_addr_next - substream->runtime->dma_addr);
+
+	return x == substream->runtime->buffer_size ? 0 : x;
+}
+
+static int bcm63xx_pcm_mmap(struct snd_soc_component *component,
+				struct snd_pcm_substream *substream,
+				struct vm_area_struct *vma)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+
+	return  dma_mmap_wc(substream->pcm->card->dev, vma,
+			    runtime->dma_area,
+			    runtime->dma_addr,
+			    runtime->dma_bytes);
+
+}
+
+static int bcm63xx_pcm_open(struct snd_soc_component *component,
+			struct snd_pcm_substream *substream)
+{
+	int ret = 0;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct bcm63xx_runtime_data *prtd;
+
+	runtime->hw = bcm63xx_pcm_hardware;
+	ret = snd_pcm_hw_constraint_step(runtime, 0,
+					 SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 32);
+	if (ret)
+		goto out;
+
+	ret = snd_pcm_hw_constraint_step(runtime, 0,
+					 SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 32);
+	if (ret)
+		goto out;
+
+	ret = snd_pcm_hw_constraint_integer(runtime,
+					    SNDRV_PCM_HW_PARAM_PERIODS);
+	if (ret < 0)
+		goto out;
+
+	ret = -ENOMEM;
+	prtd = kzalloc(sizeof(*prtd), GFP_KERNEL);
+	if (!prtd)
+		goto out;
+
+	runtime->private_data = prtd;
+	return 0;
+out:
+	return ret;
+}
+
+static int bcm63xx_pcm_close(struct snd_soc_component *component,
+			struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct bcm63xx_runtime_data *prtd = runtime->private_data;
+
+	kfree(prtd);
+	return 0;
+}
+
+static irqreturn_t i2s_dma_isr(int irq, void *bcm_i2s_priv)
+{
+	unsigned int availdepth, ifflevel, offlevel, int_status, val_1, val_2;
+	struct bcm63xx_runtime_data *prtd;
+	struct snd_pcm_substream *substream;
+	struct snd_pcm_runtime *runtime;
+	struct regmap *regmap_i2s;
+	struct i2s_dma_desc *dma_desc;
+	struct snd_soc_pcm_runtime *rtd;
+	struct bcm_i2s_priv *i2s_priv;
+
+	i2s_priv = (struct bcm_i2s_priv *)bcm_i2s_priv;
+	regmap_i2s = i2s_priv->regmap_i2s;
+
+	/* rx */
+	regmap_read(regmap_i2s, I2S_RX_IRQ_CTL, &int_status);
+
+	if (int_status & I2S_RX_DESC_OFF_INTR_EN_MSK) {
+		substream = i2s_priv->capture_substream;
+		runtime = substream->runtime;
+		rtd = substream->private_data;
+		prtd = runtime->private_data;
+		dma_desc = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream);
+
+		offlevel = (int_status & I2S_RX_DESC_OFF_LEVEL_MASK) >>
+			   I2S_RX_DESC_OFF_LEVEL_SHIFT;
+		while (offlevel) {
+			regmap_read(regmap_i2s, I2S_RX_DESC_OFF_ADDR, &val_1);
+			regmap_read(regmap_i2s, I2S_RX_DESC_OFF_LEN, &val_2);
+			offlevel--;
+		}
+		prtd->dma_addr_next = val_1 + val_2;
+		ifflevel = (int_status & I2S_RX_DESC_IFF_LEVEL_MASK) >>
+			   I2S_RX_DESC_IFF_LEVEL_SHIFT;
+
+		availdepth = I2S_DESC_FIFO_DEPTH - ifflevel;
+		while (availdepth) {
+			dma_desc->dma_addr +=
+					snd_pcm_lib_period_bytes(substream);
+			dma_desc->dma_area +=
+					snd_pcm_lib_period_bytes(substream);
+			if (dma_desc->dma_addr - runtime->dma_addr >=
+						runtime->dma_bytes) {
+				dma_desc->dma_addr = runtime->dma_addr;
+				dma_desc->dma_area = runtime->dma_area;
+			}
+
+			prtd->dma_addr = dma_desc->dma_addr;
+			regmap_write(regmap_i2s, I2S_RX_DESC_IFF_LEN,
+				     snd_pcm_lib_period_bytes(substream));
+			regmap_write(regmap_i2s, I2S_RX_DESC_IFF_ADDR,
+				     dma_desc->dma_addr);
+			availdepth--;
+		}
+
+		snd_pcm_period_elapsed(substream);
+
+		/* Clear interrupt by writing 0 */
+		regmap_update_bits(regmap_i2s, I2S_RX_IRQ_CTL,
+				   I2S_RX_INTR_MASK, 0);
+	}
+
+	/* tx */
+	regmap_read(regmap_i2s, I2S_TX_IRQ_CTL, &int_status);
+
+	if (int_status & I2S_TX_DESC_OFF_INTR_EN_MSK) {
+		substream = i2s_priv->play_substream;
+		runtime = substream->runtime;
+		rtd = substream->private_data;
+		prtd = runtime->private_data;
+		dma_desc = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream);
+
+		offlevel = (int_status & I2S_TX_DESC_OFF_LEVEL_MASK) >>
+			   I2S_TX_DESC_OFF_LEVEL_SHIFT;
+		while (offlevel) {
+			regmap_read(regmap_i2s, I2S_TX_DESC_OFF_ADDR, &val_1);
+			regmap_read(regmap_i2s, I2S_TX_DESC_OFF_LEN,  &val_2);
+			prtd->dma_addr_next = val_1 + val_2;
+			offlevel--;
+		}
+
+		ifflevel = (int_status & I2S_TX_DESC_IFF_LEVEL_MASK) >>
+			I2S_TX_DESC_IFF_LEVEL_SHIFT;
+		availdepth = I2S_DESC_FIFO_DEPTH - ifflevel;
+
+		while (availdepth) {
+			dma_desc->dma_addr +=
+					snd_pcm_lib_period_bytes(substream);
+			dma_desc->dma_area +=
+					snd_pcm_lib_period_bytes(substream);
+
+			if (dma_desc->dma_addr - runtime->dma_addr >=
+							runtime->dma_bytes) {
+				dma_desc->dma_addr = runtime->dma_addr;
+				dma_desc->dma_area = runtime->dma_area;
+			}
+
+			prtd->dma_addr = dma_desc->dma_addr;
+			regmap_write(regmap_i2s, I2S_TX_DESC_IFF_LEN,
+				snd_pcm_lib_period_bytes(substream));
+			regmap_write(regmap_i2s, I2S_TX_DESC_IFF_ADDR,
+					dma_desc->dma_addr);
+			availdepth--;
+		}
+
+		snd_pcm_period_elapsed(substream);
+
+		/* Clear interrupt by writing 0 */
+		regmap_update_bits(regmap_i2s, I2S_TX_IRQ_CTL,
+				   I2S_TX_INTR_MASK, 0);
+	}
+
+	return IRQ_HANDLED;
+}
+
+static int bcm63xx_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 = bcm63xx_pcm_hardware.buffer_bytes_max;
+
+	buf->dev.type = SNDRV_DMA_TYPE_DEV;
+	buf->dev.dev = pcm->card->dev;
+	buf->private_data = NULL;
+
+	buf->area = dma_alloc_wc(pcm->card->dev,
+				 size, &buf->addr,
+				 GFP_KERNEL);
+	if (!buf->area)
+		return -ENOMEM;
+	buf->bytes = size;
+	return 0;
+}
+
+static int bcm63xx_soc_pcm_new(struct snd_soc_component *component,
+		struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_pcm *pcm = rtd->pcm;
+	struct bcm_i2s_priv *i2s_priv;
+	int ret;
+
+	i2s_priv = dev_get_drvdata(asoc_rtd_to_cpu(rtd, 0)->dev);
+
+	of_dma_configure(pcm->card->dev, pcm->card->dev->of_node, 1);
+
+	ret = dma_coerce_mask_and_coherent(pcm->card->dev, DMA_BIT_MASK(32));
+	if (ret)
+		goto out;
+
+	if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
+		ret = bcm63xx_pcm_preallocate_dma_buffer(pcm,
+						 SNDRV_PCM_STREAM_PLAYBACK);
+		if (ret)
+			goto out;
+
+		i2s_priv->play_substream =
+			pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
+	}
+
+	if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
+		ret = bcm63xx_pcm_preallocate_dma_buffer(pcm,
+					SNDRV_PCM_STREAM_CAPTURE);
+		if (ret)
+			goto out;
+		i2s_priv->capture_substream =
+			pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream;
+	}
+
+out:
+	return ret;
+}
+
+static void bcm63xx_pcm_free_dma_buffers(struct snd_soc_component *component,
+			 struct snd_pcm *pcm)
+{
+	int stream;
+	struct snd_dma_buffer *buf;
+	struct snd_pcm_substream *substream;
+
+	for (stream = 0; stream < 2; stream++) {
+		substream = pcm->streams[stream].substream;
+		if (!substream)
+			continue;
+		buf = &substream->dma_buffer;
+		if (!buf->area)
+			continue;
+		dma_free_wc(pcm->card->dev, buf->bytes,
+					buf->area, buf->addr);
+		buf->area = NULL;
+	}
+}
+
+static const struct snd_soc_component_driver bcm63xx_soc_platform = {
+	.open = bcm63xx_pcm_open,
+	.close = bcm63xx_pcm_close,
+	.hw_params = bcm63xx_pcm_hw_params,
+	.hw_free = bcm63xx_pcm_hw_free,
+	.prepare = bcm63xx_pcm_prepare,
+	.trigger = bcm63xx_pcm_trigger,
+	.pointer = bcm63xx_pcm_pointer,
+	.mmap = bcm63xx_pcm_mmap,
+	.pcm_construct = bcm63xx_soc_pcm_new,
+	.pcm_destruct = bcm63xx_pcm_free_dma_buffers,
+};
+
+int bcm63xx_soc_platform_probe(struct platform_device *pdev,
+			       struct bcm_i2s_priv *i2s_priv)
+{
+	int ret;
+
+	i2s_priv->r_irq = platform_get_resource(pdev, IORESOURCE_IRQ, 0);
+	if (!i2s_priv->r_irq) {
+		dev_err(&pdev->dev, "Unable to get register irq resource.\n");
+		return -ENODEV;
+	}
+
+	ret = devm_request_irq(&pdev->dev, i2s_priv->r_irq->start, i2s_dma_isr,
+			i2s_priv->r_irq->flags, "i2s_dma", (void *)i2s_priv);
+	if (ret) {
+		dev_err(&pdev->dev,
+			"i2s_init: failed to request interrupt.ret=%d\n", ret);
+		return ret;
+	}
+
+	return devm_snd_soc_register_component(&pdev->dev,
+					&bcm63xx_soc_platform, NULL, 0);
+}
+
+int bcm63xx_soc_platform_remove(struct platform_device *pdev)
+{
+	return 0;
+}
+
+MODULE_AUTHOR("Kevin,Li <[e-mail]>");
+MODULE_DESCRIPTION("Broadcom DSL XPON ASOC PCM Interface");
+MODULE_LICENSE("GPL v2");

sound/soc/bcm/cygnus-pcm.c

···
 {
 	struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
 
-	return snd_soc_dai_get_dma_data(soc_runtime->cpu_dai, substream);
+	return snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(soc_runtime, 0), substream);
 }
 
 static void ringbuf_set_initial(void __iomem *audio_io,
···
 
 	aio = cygnus_dai_get_dma_data(substream);
 
-	dev_dbg(rtd->cpu_dai->dev, "%s on port %d\n", __func__, aio->portnum);
+	dev_dbg(asoc_rtd_to_cpu(rtd, 0)->dev, "%s on port %d\n", __func__, aio->portnum);
 
 	/* The port number maps to the bit position to be set */
 	set_mask = BIT(aio->portnum);
···
 	if (!aio)
 		return -ENODEV;
 
-	dev_dbg(rtd->cpu_dai->dev, "%s port %d\n", __func__, aio->portnum);
+	dev_dbg(asoc_rtd_to_cpu(rtd, 0)->dev, "%s port %d\n", __func__, aio->portnum);
 
 	snd_soc_set_runtime_hwparams(substream, &cygnus_pcm_hw);
 
···
 
 	aio = cygnus_dai_get_dma_data(substream);
 
-	dev_dbg(rtd->cpu_dai->dev, "%s  port %d\n", __func__, aio->portnum);
+	dev_dbg(asoc_rtd_to_cpu(rtd, 0)->dev, "%s  port %d\n", __func__, aio->portnum);
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 		aio->play_stream = NULL;
···
 		aio->capture_stream = NULL;
 
 	if (!aio->play_stream && !aio->capture_stream)
-		dev_dbg(rtd->cpu_dai->dev, "freed  port %d\n", aio->portnum);
+		dev_dbg(asoc_rtd_to_cpu(rtd, 0)->dev, "freed  port %d\n", aio->portnum);
 
 	return 0;
 }
···
 	struct cygnus_aio_port *aio;
 
 	aio = cygnus_dai_get_dma_data(substream);
-	dev_dbg(rtd->cpu_dai->dev, "%s  port %d\n", __func__, aio->portnum);
+	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);
···
 	struct cygnus_aio_port *aio;
 
 	aio = cygnus_dai_get_dma_data(substream);
-	dev_dbg(rtd->cpu_dai->dev, "%s  port %d\n", __func__, aio->portnum);
+	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;
···
 	struct ringbuf_regs *p_rbuf = NULL;
 
 	aio = cygnus_dai_get_dma_data(substream);
-	dev_dbg(rtd->cpu_dai->dev, "%s port %d\n", __func__, aio->portnum);
+	dev_dbg(asoc_rtd_to_cpu(rtd, 0)->dev, "%s port %d\n", __func__, aio->portnum);
 
 	bufsize = snd_pcm_lib_buffer_bytes(substream);
 	periodsize = snd_pcm_lib_period_bytes(substream);
 
-	dev_dbg(rtd->cpu_dai->dev, "%s (buf_size %lu) (period_size %lu)\n",
+	dev_dbg(asoc_rtd_to_cpu(rtd, 0)->dev, "%s (buf_size %lu) (period_size %lu)\n",
 			__func__, bufsize, periodsize);
 
 	configure_ringbuf_regs(substream);
···
 	buf->area = dma_alloc_coherent(pcm->card->dev, size,
 			&buf->addr, GFP_KERNEL);
 
-	dev_dbg(rtd->cpu_dai->dev, "%s: size 0x%zx @ %pK\n",
+	dev_dbg(asoc_rtd_to_cpu(rtd, 0)->dev, "%s: size 0x%zx @ %pK\n",
 				__func__, size, buf->area);
 
 	if (!buf->area) {
-		dev_err(rtd->cpu_dai->dev, "%s: dma_alloc failed\n", __func__);
+		dev_err(asoc_rtd_to_cpu(rtd, 0)->dev, "%s: dma_alloc failed\n", __func__);
 		return -ENOMEM;
 	}
 	buf->bytes = size;

sound/soc/cirrus/edb93xx.c

···
 			     struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	int err;
 	unsigned int mclk_rate;
 	unsigned int rate = params_rate(params);

sound/soc/cirrus/snappercl15.c

···
 				 struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	int err;
 
 	err = snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, 

sound/soc/codecs/Kconfig

···
 config SND_SOC_ALL_CODECS
 	tristate "Build all ASoC CODEC drivers"
 	depends on COMPILE_TEST
-	select SND_SOC_88PM860X if MFD_88PM860X
-	select SND_SOC_L3
-	select SND_SOC_AB8500_CODEC if ABX500_CORE
-	select SND_SOC_AC97_CODEC
-	select SND_SOC_AD1836 if SPI_MASTER
-	select SND_SOC_AD193X_SPI if SPI_MASTER
-	select SND_SOC_AD193X_I2C if I2C
-	select SND_SOC_AD1980 if SND_SOC_AC97_BUS
-	select SND_SOC_AD73311
-	select SND_SOC_ADAU1373 if I2C
-	select SND_SOC_ADAU1761_I2C if I2C
-	select SND_SOC_ADAU1761_SPI if SPI
-	select SND_SOC_ADAU1781_I2C if I2C
-	select SND_SOC_ADAU1781_SPI if SPI
-	select SND_SOC_ADAV801 if SPI_MASTER
-	select SND_SOC_ADAV803 if I2C
-	select SND_SOC_ADAU1977_SPI if SPI_MASTER
-	select SND_SOC_ADAU1977_I2C if I2C
-	select SND_SOC_ADAU1701 if I2C
-	select SND_SOC_ADAU7002
-	select SND_SOC_ADAU7118_I2C if I2C
-	select SND_SOC_ADAU7118_HW
-	select SND_SOC_ADS117X
-	select SND_SOC_AK4104 if SPI_MASTER
-	select SND_SOC_AK4118 if I2C
-	select SND_SOC_AK4458 if I2C
-	select SND_SOC_AK4535 if I2C
-	select SND_SOC_AK4554
-	select SND_SOC_AK4613 if I2C
-	select SND_SOC_AK4641 if I2C
-	select SND_SOC_AK4642 if I2C
-	select SND_SOC_AK4671 if I2C
-	select SND_SOC_AK5386
-	select SND_SOC_AK5558 if I2C
-	select SND_SOC_ALC5623 if I2C
-	select SND_SOC_ALC5632 if I2C
-	select SND_SOC_BT_SCO
-	select SND_SOC_BD28623
-	select SND_SOC_CQ0093VC
-	select SND_SOC_CROS_EC_CODEC if CROS_EC
-	select SND_SOC_CS35L32 if I2C
-	select SND_SOC_CS35L33 if I2C
-	select SND_SOC_CS35L34 if I2C
-	select SND_SOC_CS35L35 if I2C
-	select SND_SOC_CS35L36 if I2C
-	select SND_SOC_CS42L42 if I2C
-	select SND_SOC_CS42L51_I2C if I2C
-	select SND_SOC_CS42L52 if I2C && INPUT
-	select SND_SOC_CS42L56 if I2C && INPUT
-	select SND_SOC_CS42L73 if I2C
-	select SND_SOC_CS4265 if I2C
-	select SND_SOC_CS4270 if I2C
-	select SND_SOC_CS4271_I2C if I2C
-	select SND_SOC_CS4271_SPI if SPI_MASTER
-	select SND_SOC_CS42XX8_I2C if I2C
-	select SND_SOC_CS43130 if I2C
-	select SND_SOC_CS4341 if SND_SOC_I2C_AND_SPI
-	select SND_SOC_CS4349 if I2C
-	select SND_SOC_CS47L15 if MFD_CS47L15
-	select SND_SOC_CS47L24 if MFD_CS47L24
-	select SND_SOC_CS47L35 if MFD_CS47L35
-	select SND_SOC_CS47L85 if MFD_CS47L85
-	select SND_SOC_CS47L90 if MFD_CS47L90
-	select SND_SOC_CS47L92 if MFD_CS47L92
-	select SND_SOC_CS53L30 if I2C
-	select SND_SOC_CX20442 if TTY
-	select SND_SOC_CX2072X if I2C
-	select SND_SOC_DA7210 if SND_SOC_I2C_AND_SPI
-	select SND_SOC_DA7213 if I2C
-	select SND_SOC_DA7218 if I2C
-	select SND_SOC_DA7219 if I2C
-	select SND_SOC_DA732X if I2C
-	select SND_SOC_DA9055 if I2C
-	select SND_SOC_DMIC if GPIOLIB
-	select SND_SOC_ES8316 if I2C
-	select SND_SOC_ES8328_SPI if SPI_MASTER
-	select SND_SOC_ES8328_I2C if I2C
-	select SND_SOC_ES7134
-	select SND_SOC_ES7241
-	select SND_SOC_GTM601
-	select SND_SOC_HDAC_HDMI
-	select SND_SOC_HDAC_HDA
-	select SND_SOC_ICS43432
-	select SND_SOC_INNO_RK3036
-	select SND_SOC_ISABELLE if I2C
-	select SND_SOC_JZ4740_CODEC
-	select SND_SOC_JZ4725B_CODEC
-	select SND_SOC_JZ4770_CODEC
-	select SND_SOC_LM4857 if I2C
-	select SND_SOC_LM49453 if I2C
-	select SND_SOC_LOCHNAGAR_SC if MFD_LOCHNAGAR
-	select SND_SOC_MAX98088 if I2C
-	select SND_SOC_MAX98090 if I2C
-	select SND_SOC_MAX98095 if I2C
-	select SND_SOC_MAX98357A if GPIOLIB
-	select SND_SOC_MAX98371 if I2C
-	select SND_SOC_MAX98504 if I2C
-	select SND_SOC_MAX9867 if I2C
-	select SND_SOC_MAX98925 if I2C
-	select SND_SOC_MAX98926 if I2C
-	select SND_SOC_MAX98927 if I2C
-	select SND_SOC_MAX98373 if I2C
-	select SND_SOC_MAX9850 if I2C
-	select SND_SOC_MAX9860 if I2C
-	select SND_SOC_MAX9759
-	select SND_SOC_MAX9768 if I2C
-	select SND_SOC_MAX9877 if I2C
-	select SND_SOC_MC13783 if MFD_MC13XXX
-	select SND_SOC_ML26124 if I2C
-	select SND_SOC_MT6351 if MTK_PMIC_WRAP
-	select SND_SOC_MT6358 if MTK_PMIC_WRAP
-	select SND_SOC_MT6660 if I2C
-	select SND_SOC_NAU8540 if I2C
-	select SND_SOC_NAU8810 if I2C
-	select SND_SOC_NAU8822 if I2C
-	select SND_SOC_NAU8824 if I2C
-	select SND_SOC_NAU8825 if I2C
-	select SND_SOC_HDMI_CODEC
-	select SND_SOC_PCM1681 if I2C
-	select SND_SOC_PCM1789_I2C if I2C
-	select SND_SOC_PCM179X_I2C if I2C
-	select SND_SOC_PCM179X_SPI if SPI_MASTER
-	select SND_SOC_PCM186X_I2C if I2C
-	select SND_SOC_PCM186X_SPI if SPI_MASTER
-	select SND_SOC_PCM3008
-	select SND_SOC_PCM3060_I2C if I2C
-	select SND_SOC_PCM3060_SPI if SPI_MASTER
-	select SND_SOC_PCM3168A_I2C if I2C
-	select SND_SOC_PCM3168A_SPI if SPI_MASTER
-	select SND_SOC_PCM5102A
-	select SND_SOC_PCM512x_I2C if I2C
-	select SND_SOC_PCM512x_SPI if SPI_MASTER
-	select SND_SOC_RK3328
-	select SND_SOC_RT274 if I2C
-	select SND_SOC_RT286 if I2C
-	select SND_SOC_RT298 if I2C
-	select SND_SOC_RT1011 if I2C
-	select SND_SOC_RT1015 if I2C
-	select SND_SOC_RT1305 if I2C
-	select SND_SOC_RT1308 if I2C
-	select SND_SOC_RT5514 if I2C
-	select SND_SOC_RT5616 if I2C
-	select SND_SOC_RT5631 if I2C
-	select SND_SOC_RT5640 if I2C
-	select SND_SOC_RT5645 if I2C
-	select SND_SOC_RT5651 if I2C
-	select SND_SOC_RT5659 if I2C
-	select SND_SOC_RT5660 if I2C
-	select SND_SOC_RT5663 if I2C
-	select SND_SOC_RT5665 if I2C
-	select SND_SOC_RT5668 if I2C
-	select SND_SOC_RT5670 if I2C
-	select SND_SOC_RT5677 if I2C && SPI_MASTER
-	select SND_SOC_RT5682 if I2C
-	select SND_SOC_RT700_SDW if SOUNDWIRE
-	select SND_SOC_RT711_SDW if SOUNDWIRE
-	select SND_SOC_RT715_SDW if SOUNDWIRE
-	select SND_SOC_RT1308_SDW if SOUNDWIRE
-	select SND_SOC_SGTL5000 if I2C
-	select SND_SOC_SI476X if MFD_SI476X_CORE
-	select SND_SOC_SIMPLE_AMPLIFIER
-	select SND_SOC_SIRF_AUDIO_CODEC
-	select SND_SOC_SPDIF
-	select SND_SOC_SSM2305
-	select SND_SOC_SSM2518 if I2C
-	select SND_SOC_SSM2602_SPI if SPI_MASTER
-	select SND_SOC_SSM2602_I2C if I2C
-	select SND_SOC_SSM4567 if I2C
-	select SND_SOC_STA32X if I2C
-	select SND_SOC_STA350 if I2C
-	select SND_SOC_STA529 if I2C
-	select SND_SOC_STAC9766 if SND_SOC_AC97_BUS
-	select SND_SOC_STI_SAS
-	select SND_SOC_TAS2552 if I2C
-	select SND_SOC_TAS2562 if I2C
-	select SND_SOC_TAS2770 if I2C
-	select SND_SOC_TAS5086 if I2C
-	select SND_SOC_TAS571X if I2C
-	select SND_SOC_TAS5720 if I2C
-	select SND_SOC_TAS6424 if I2C
-	select SND_SOC_TDA7419 if I2C
-	select SND_SOC_TFA9879 if I2C
-	select SND_SOC_TLV320AIC23_I2C if I2C
-	select SND_SOC_TLV320AIC23_SPI if SPI_MASTER
-	select SND_SOC_TLV320AIC26 if SPI_MASTER
-	select SND_SOC_TLV320AIC31XX if I2C
-	select SND_SOC_TLV320AIC32X4_I2C if I2C && COMMON_CLK
-	select SND_SOC_TLV320AIC32X4_SPI if SPI_MASTER && COMMON_CLK
-	select SND_SOC_TLV320AIC3X if I2C
-	select SND_SOC_TPA6130A2 if I2C
-	select SND_SOC_TLV320DAC33 if I2C
-	select SND_SOC_TSCS42XX if I2C
-	select SND_SOC_TSCS454 if I2C
-	select SND_SOC_TS3A227E if I2C
-	select SND_SOC_TWL4030 if TWL4030_CORE
-	select SND_SOC_TWL6040 if TWL6040_CORE
-	select SND_SOC_UDA1334 if GPIOLIB
-	select SND_SOC_UDA134X
-	select SND_SOC_UDA1380 if I2C
-	select SND_SOC_WCD9335 if SLIMBUS
-	select SND_SOC_WCD934X if MFD_WCD934X && COMMON_CLK
-	select SND_SOC_WL1273 if MFD_WL1273_CORE
-	select SND_SOC_WM0010 if SPI_MASTER
-	select SND_SOC_WM1250_EV1 if I2C
-	select SND_SOC_WM2000 if I2C
-	select SND_SOC_WM2200 if I2C
-	select SND_SOC_WM5100 if I2C
-	select SND_SOC_WM5102 if MFD_WM5102
-	select SND_SOC_WM5110 if MFD_WM5110
-	select SND_SOC_WM8350 if MFD_WM8350
-	select SND_SOC_WM8400 if MFD_WM8400
-	select SND_SOC_WM8510 if SND_SOC_I2C_AND_SPI
-	select SND_SOC_WM8523 if I2C
-	select SND_SOC_WM8524 if GPIOLIB
-	select SND_SOC_WM8580 if I2C
-	select SND_SOC_WM8711 if SND_SOC_I2C_AND_SPI
-	select SND_SOC_WM8727
-	select SND_SOC_WM8728 if SND_SOC_I2C_AND_SPI
-	select SND_SOC_WM8731 if SND_SOC_I2C_AND_SPI
-	select SND_SOC_WM8737 if SND_SOC_I2C_AND_SPI
-	select SND_SOC_WM8741 if SND_SOC_I2C_AND_SPI
-	select SND_SOC_WM8750 if SND_SOC_I2C_AND_SPI
-	select SND_SOC_WM8753 if SND_SOC_I2C_AND_SPI
-	select SND_SOC_WM8770 if SPI_MASTER
-	select SND_SOC_WM8776 if SND_SOC_I2C_AND_SPI
-	select SND_SOC_WM8782
-	select SND_SOC_WM8804_I2C if I2C
-	select SND_SOC_WM8804_SPI if SPI_MASTER
-	select SND_SOC_WM8900 if I2C
-	select SND_SOC_WM8903 if I2C
-	select SND_SOC_WM8904 if I2C
-	select SND_SOC_WM8940 if I2C
-	select SND_SOC_WM8955 if I2C
-	select SND_SOC_WM8960 if I2C
-	select SND_SOC_WM8961 if I2C
-	select SND_SOC_WM8962 if I2C && INPUT
-	select SND_SOC_WM8971 if I2C
-	select SND_SOC_WM8974 if I2C
-	select SND_SOC_WM8978 if I2C
-	select SND_SOC_WM8983 if SND_SOC_I2C_AND_SPI
-	select SND_SOC_WM8985 if SND_SOC_I2C_AND_SPI
-	select SND_SOC_WM8988 if SND_SOC_I2C_AND_SPI
-	select SND_SOC_WM8990 if I2C
-	select SND_SOC_WM8991 if I2C
-	select SND_SOC_WM8993 if I2C
-	select SND_SOC_WM8994 if MFD_WM8994
-	select SND_SOC_WM8995 if SND_SOC_I2C_AND_SPI
-	select SND_SOC_WM8996 if I2C
-	select SND_SOC_WM8997 if MFD_WM8997
-	select SND_SOC_WM8998 if MFD_WM8998
-	select SND_SOC_WM9081 if I2C
-	select SND_SOC_WM9090 if I2C
-	select SND_SOC_WM9705 if (SND_SOC_AC97_BUS || SND_SOC_AC97_BUS_NEW)
-	select SND_SOC_WM9712 if (SND_SOC_AC97_BUS || SND_SOC_AC97_BUS_NEW)
-	select SND_SOC_WM9713 if (SND_SOC_AC97_BUS || SND_SOC_AC97_BUS_NEW)
-	select SND_SOC_WSA881X if SOUNDWIRE
+	imply SND_SOC_88PM860X
+	imply SND_SOC_L3
+	imply SND_SOC_AB8500_CODEC
+	imply SND_SOC_AC97_CODEC
+	imply SND_SOC_AD1836
+	imply SND_SOC_AD193X_SPI
+	imply SND_SOC_AD193X_I2C
+	imply SND_SOC_AD1980
+	imply SND_SOC_AD73311
+	imply SND_SOC_ADAU1373
+	imply SND_SOC_ADAU1761_I2C
+	imply SND_SOC_ADAU1761_SPI
+	imply SND_SOC_ADAU1781_I2C
+	imply SND_SOC_ADAU1781_SPI
+	imply SND_SOC_ADAV801
+	imply SND_SOC_ADAV803
+	imply SND_SOC_ADAU1977_SPI
+	imply SND_SOC_ADAU1977_I2C
+	imply SND_SOC_ADAU1701
+	imply SND_SOC_ADAU7002
+	imply SND_SOC_ADAU7118_I2C
+	imply SND_SOC_ADAU7118_HW
+	imply SND_SOC_ADS117X
+	imply SND_SOC_AK4104
+	imply SND_SOC_AK4118
+	imply SND_SOC_AK4458
+	imply SND_SOC_AK4535
+	imply SND_SOC_AK4554
+	imply SND_SOC_AK4613
+	imply SND_SOC_AK4641
+	imply SND_SOC_AK4642
+	imply SND_SOC_AK4671
+	imply SND_SOC_AK5386
+	imply SND_SOC_AK5558
+	imply SND_SOC_ALC5623
+	imply SND_SOC_ALC5632
+	imply SND_SOC_BT_SCO
+	imply SND_SOC_BD28623
+	imply SND_SOC_CQ0093VC
+	imply SND_SOC_CROS_EC_CODEC
+	imply SND_SOC_CS35L32
+	imply SND_SOC_CS35L33
+	imply SND_SOC_CS35L34
+	imply SND_SOC_CS35L35
+	imply SND_SOC_CS35L36
+	imply SND_SOC_CS42L42
+	imply SND_SOC_CS42L51_I2C
+	imply SND_SOC_CS42L52
+	imply SND_SOC_CS42L56
+	imply SND_SOC_CS42L73
+	imply SND_SOC_CS4265
+	imply SND_SOC_CS4270
+	imply SND_SOC_CS4271_I2C
+	imply SND_SOC_CS4271_SPI
+	imply SND_SOC_CS42XX8_I2C
+	imply SND_SOC_CS43130
+	imply SND_SOC_CS4341
+	imply SND_SOC_CS4349
+	imply SND_SOC_CS47L15
+	imply SND_SOC_CS47L24
+	imply SND_SOC_CS47L35
+	imply SND_SOC_CS47L85
+	imply SND_SOC_CS47L90
+	imply SND_SOC_CS47L92
+	imply SND_SOC_CS53L30
+	imply SND_SOC_CX20442
+	imply SND_SOC_CX2072X
+	imply SND_SOC_DA7210
+	imply SND_SOC_DA7213
+	imply SND_SOC_DA7218
+	imply SND_SOC_DA7219
+	imply SND_SOC_DA732X
+	imply SND_SOC_DA9055
+	imply SND_SOC_DMIC
+	imply SND_SOC_ES8316
+	imply SND_SOC_ES8328_SPI
+	imply SND_SOC_ES8328_I2C
+	imply SND_SOC_ES7134
+	imply SND_SOC_ES7241
+	imply SND_SOC_GTM601
+	imply SND_SOC_HDAC_HDMI
+	imply SND_SOC_HDAC_HDA
+	imply SND_SOC_ICS43432
+	imply SND_SOC_INNO_RK3036
+	imply SND_SOC_ISABELLE
+	imply SND_SOC_JZ4740_CODEC
+	imply SND_SOC_JZ4725B_CODEC
+	imply SND_SOC_JZ4770_CODEC
+	imply SND_SOC_LM4857
+	imply SND_SOC_LM49453
+	imply SND_SOC_LOCHNAGAR_SC
+	imply SND_SOC_MAX98088
+	imply SND_SOC_MAX98090
+	imply SND_SOC_MAX98095
+	imply SND_SOC_MAX98357A
+	imply SND_SOC_MAX98371
+	imply SND_SOC_MAX98504
+	imply SND_SOC_MAX9867
+	imply SND_SOC_MAX98925
+	imply SND_SOC_MAX98926
+	imply SND_SOC_MAX98927
+	imply SND_SOC_MAX98373
+	imply SND_SOC_MAX9850
+	imply SND_SOC_MAX9860
+	imply SND_SOC_MAX9759
+	imply SND_SOC_MAX9768
+	imply SND_SOC_MAX9877
+	imply SND_SOC_MC13783
+	imply SND_SOC_ML26124
+	imply SND_SOC_MT6351
+	imply SND_SOC_MT6358
+	imply SND_SOC_MT6660
+	imply SND_SOC_NAU8540
+	imply SND_SOC_NAU8810
+	imply SND_SOC_NAU8822
+	imply SND_SOC_NAU8824
+	imply SND_SOC_NAU8825
+	imply SND_SOC_HDMI_CODEC
+	imply SND_SOC_PCM1681
+	imply SND_SOC_PCM1789_I2C
+	imply SND_SOC_PCM179X_I2C
+	imply SND_SOC_PCM179X_SPI
+	imply SND_SOC_PCM186X_I2C
+	imply SND_SOC_PCM186X_SPI
+	imply SND_SOC_PCM3008
+	imply SND_SOC_PCM3060_I2C
+	imply SND_SOC_PCM3060_SPI
+	imply SND_SOC_PCM3168A_I2C
+	imply SND_SOC_PCM3168A_SPI
+	imply SND_SOC_PCM5102A
+	imply SND_SOC_PCM512x_I2C
+	imply SND_SOC_PCM512x_SPI
+	imply SND_SOC_RK3328
+	imply SND_SOC_RT274
+	imply SND_SOC_RT286
+	imply SND_SOC_RT298
+	imply SND_SOC_RT1011
+	imply SND_SOC_RT1015
+	imply SND_SOC_RT1305
+	imply SND_SOC_RT1308
+	imply SND_SOC_RT5514
+	imply SND_SOC_RT5616
+	imply SND_SOC_RT5631
+	imply SND_SOC_RT5640
+	imply SND_SOC_RT5645
+	imply SND_SOC_RT5651
+	imply SND_SOC_RT5659
+	imply SND_SOC_RT5660
+	imply SND_SOC_RT5663
+	imply SND_SOC_RT5665
+	imply SND_SOC_RT5668
+	imply SND_SOC_RT5670
+	imply SND_SOC_RT5677
+	imply SND_SOC_RT5682
+	imply SND_SOC_RT5682_SDW
+	imply SND_SOC_RT700_SDW
+	imply SND_SOC_RT711_SDW
+	imply SND_SOC_RT715_SDW
+	imply SND_SOC_RT1308_SDW
+	imply SND_SOC_SGTL5000
+	imply SND_SOC_SI476X
+	imply SND_SOC_SIMPLE_AMPLIFIER
+	imply SND_SOC_SIRF_AUDIO_CODEC
+	imply SND_SOC_SPDIF
+	imply SND_SOC_SSM2305
+	imply SND_SOC_SSM2518
+	imply SND_SOC_SSM2602_SPI
+	imply SND_SOC_SSM2602_I2C
+	imply SND_SOC_SSM4567
+	imply SND_SOC_STA32X
+	imply SND_SOC_STA350
+	imply SND_SOC_STA529
+	imply SND_SOC_STAC9766
+	imply SND_SOC_STI_SAS
+	imply SND_SOC_TAS2552
+	imply SND_SOC_TAS2562
+	imply SND_SOC_TAS2770
+	imply SND_SOC_TAS5086
+	imply SND_SOC_TAS571X
+	imply SND_SOC_TAS5720
+	imply SND_SOC_TAS6424
+	imply SND_SOC_TDA7419
+	imply SND_SOC_TFA9879
+	imply SND_SOC_TLV320ADCX140
+	imply SND_SOC_TLV320AIC23_I2C
+	imply SND_SOC_TLV320AIC23_SPI
+	imply SND_SOC_TLV320AIC26
+	imply SND_SOC_TLV320AIC31XX
+	imply SND_SOC_TLV320AIC32X4_I2C
+	imply SND_SOC_TLV320AIC32X4_SPI
+	imply SND_SOC_TLV320AIC3X
+	imply SND_SOC_TPA6130A2
+	imply SND_SOC_TLV320DAC33
+	imply SND_SOC_TSCS42XX
+	imply SND_SOC_TSCS454
+	imply SND_SOC_TS3A227E
+	imply SND_SOC_TWL4030
+	imply SND_SOC_TWL6040
+	imply SND_SOC_UDA1334
+	imply SND_SOC_UDA134X
+	imply SND_SOC_UDA1380
+	imply SND_SOC_WCD9335
+	imply SND_SOC_WCD934X
+	imply SND_SOC_WL1273
+	imply SND_SOC_WM0010
+	imply SND_SOC_WM1250_EV1
+	imply SND_SOC_WM2000
+	imply SND_SOC_WM2200
+	imply SND_SOC_WM5100
+	imply SND_SOC_WM5102
+	imply SND_SOC_WM5110
+	imply SND_SOC_WM8350
+	imply SND_SOC_WM8400
+	imply SND_SOC_WM8510
+	imply SND_SOC_WM8523
+	imply SND_SOC_WM8524
+	imply SND_SOC_WM8580
+	imply SND_SOC_WM8711
+	imply SND_SOC_WM8727
+	imply SND_SOC_WM8728
+	imply SND_SOC_WM8731
+	imply SND_SOC_WM8737
+	imply SND_SOC_WM8741
+	imply SND_SOC_WM8750
+	imply SND_SOC_WM8753
+	imply SND_SOC_WM8770
+	imply SND_SOC_WM8776
+	imply SND_SOC_WM8782
+	imply SND_SOC_WM8804_I2C
+	imply SND_SOC_WM8804_SPI
+	imply SND_SOC_WM8900
+	imply SND_SOC_WM8903
+	imply SND_SOC_WM8904
+	imply SND_SOC_WM8940
+	imply SND_SOC_WM8955
+	imply SND_SOC_WM8960
+	imply SND_SOC_WM8961
+	imply SND_SOC_WM8962
+	imply SND_SOC_WM8971
+	imply SND_SOC_WM8974
+	imply SND_SOC_WM8978
+	imply SND_SOC_WM8983
+	imply SND_SOC_WM8985
+	imply SND_SOC_WM8988
+	imply SND_SOC_WM8990
+	imply SND_SOC_WM8991
+	imply SND_SOC_WM8993
+	imply SND_SOC_WM8994
+	imply SND_SOC_WM8995
+	imply SND_SOC_WM8996
+	imply SND_SOC_WM8997
+	imply SND_SOC_WM8998
+	imply SND_SOC_WM9081
+	imply SND_SOC_WM9090
+	imply SND_SOC_WM9705
+	imply SND_SOC_WM9712
+	imply SND_SOC_WM9713
+	imply SND_SOC_WSA881X
 	help
 	  Normally ASoC codec drivers are only built if a machine driver which
 	  uses them is also built since they are only usable with a machine
···
 
 config SND_SOC_88PM860X
 	tristate
+	depends on MFD_88PM860X
 
 config SND_SOC_ARIZONA
 	tristate
···
 
 config SND_SOC_AB8500_CODEC
 	tristate
+	depends on ABX500_CORE
 
 config SND_SOC_AC97_CODEC
 	tristate "Build generic ASoC AC97 CODEC driver"
···
 
 config SND_SOC_AD1836
 	tristate
+	depends on SPI_MASTER
 
 config SND_SOC_AD193X
 	tristate
 
 config SND_SOC_AD193X_SPI
 	tristate
+	depends on SPI_MASTER
 	select SND_SOC_AD193X
 
 config SND_SOC_AD193X_I2C
 	tristate
+	depends on I2C
 	select SND_SOC_AD193X
 
 config SND_SOC_AD1980
-	select REGMAP_AC97
 	tristate
+	depends on SND_SOC_AC97_BUS
+	select REGMAP_AC97
 
 config SND_SOC_AD73311
 	tristate
···
 
 config SND_SOC_ADAU1373
 	tristate
+	depends on I2C
 	select SND_SOC_ADAU_UTILS
 
 config SND_SOC_ADAU1701
···
 
 config SND_SOC_ADAU1781_I2C
 	tristate
+	depends on I2C
 	select SND_SOC_ADAU1781
 	select REGMAP_I2C
 
 config SND_SOC_ADAU1781_SPI
 	tristate
+	depends on SPI_MASTER
 	select SND_SOC_ADAU1781
 	select REGMAP_SPI
 
···
 
 config SND_SOC_ADAU1977_SPI
 	tristate
+	depends on SPI_MASTER
 	select SND_SOC_ADAU1977
 	select REGMAP_SPI
 
 config SND_SOC_ADAU1977_I2C
 	tristate
+	depends on I2C
 	select SND_SOC_ADAU1977
 	select REGMAP_I2C
 
···
 
 config SND_SOC_ADAV801
 	tristate
+	depends on SPI_MASTER
 	select SND_SOC_ADAV80X
 
 config SND_SOC_ADAV803
 	tristate
+	depends on I2C
 	select SND_SOC_ADAV80X
 
 config SND_SOC_ADS117X
···
 
 config SND_SOC_AK4535
 	tristate
+	depends on I2C
 
 config SND_SOC_AK4554
 	tristate "AKM AK4554 CODEC"
···
 
 config SND_SOC_AK4641
 	tristate
+	depends on I2C
 
 config SND_SOC_AK4642
 	tristate "AKM AK4642 CODEC"
···
 
 config SND_SOC_AK4671
 	tristate
+	depends on I2C
 
 config SND_SOC_AK5386
 	tristate "AKM AK5638 CODEC"
···
 
 config SND_SOC_ALC5632
 	tristate
+	depends on I2C
 
 config SND_SOC_BD28623
 	tristate "ROHM BD28623 CODEC"
···
 
 config SND_SOC_CS47L24
 	tristate
+	depends on MFD_CS47L24
 
 config SND_SOC_CS47L35
 	tristate
···
 
 config SND_SOC_DA7210
 	tristate
+	depends on I2C
 
 config SND_SOC_DA7213
 	tristate "Dialog DA7213 CODEC"
···
 
 config SND_SOC_DA7218
 	tristate
+	depends on I2C
 
 config SND_SOC_DA7219
 	tristate
+	depends on I2C
 
 config SND_SOC_DA732X
 	tristate
+	depends on I2C
 
 config SND_SOC_DA9055
 	tristate
+	depends on I2C
 
 config SND_SOC_DMIC
 	tristate "Generic Digital Microphone CODEC"
···
 
 config SND_SOC_ISABELLE
 	tristate
+	depends on I2C
 
 config SND_SOC_LM49453
 	tristate
+	depends on I2C
 
 config SND_SOC_LOCHNAGAR_SC
 	tristate "Lochnagar Sound Card"
···
 	depends on I2C
 
 config SND_SOC_MAX98090
-       tristate
+	tristate
+	depends on I2C
 
 config SND_SOC_MAX98095
-       tristate
+	tristate
+	depends on I2C
 
 config SND_SOC_MAX98357A
 	tristate "Maxim MAX98357A CODEC"
 	depends on GPIOLIB
 
 config SND_SOC_MAX98371
-       tristate
+	tristate
+	depends on I2C
 
 config SND_SOC_MAX98504
 	tristate "Maxim MAX98504 speaker amplifier"
···
 	depends on I2C
 
 config SND_SOC_MAX98925
-       tristate
+	tristate
+	depends on I2C
 
 config SND_SOC_MAX98926
 	tristate
+	depends on I2C
 
 config SND_SOC_MAX98927
 	tristate "Maxim Integrated MAX98927 Speaker Amplifier"
···
 
 config SND_SOC_MAX9850
 	tristate
+	depends on I2C
 
 config SND_SOC_MAX9860
 	tristate "Maxim MAX9860 Mono Audio Voice Codec"
···
 
 config SND_SOC_RT1011
 	tristate
+	depends on I2C
 
 config SND_SOC_RT1015
 	tristate
+	depends on I2C
 
 config SND_SOC_RT1305
 	tristate
+	depends on I2C
 
 config SND_SOC_RT1308
 	tristate
+	depends on I2C
 
 config SND_SOC_RT1308_SDW
 	tristate "Realtek RT1308 Codec - SDW"
-	depends on SOUNDWIRE
+	depends on I2C && SOUNDWIRE
 	select REGMAP_SOUNDWIRE
 
 config SND_SOC_RT5514
 	tristate
+	depends on I2C
 
 config SND_SOC_RT5514_SPI
 	tristate
+	depends on SPI_MASTER
 
 config SND_SOC_RT5514_SPI_BUILTIN
 	bool # force RT5514_SPI to be built-in to avoid link errors
···
 
 config SND_SOC_RT5640
 	tristate
+	depends on I2C
 
 config SND_SOC_RT5645
 	tristate
+	depends on I2C
 
 config SND_SOC_RT5651
 	tristate
+	depends on I2C
 
 config SND_SOC_RT5659
 	tristate
+	depends on I2C
 
 config SND_SOC_RT5660
 	tristate
+	depends on I2C
 
 config SND_SOC_RT5663
 	tristate
+	depends on I2C
 
 config SND_SOC_RT5665
 	tristate
+	depends on I2C
 
 config SND_SOC_RT5668
 	tristate
+	depends on I2C
 
 config SND_SOC_RT5670
 	tristate
+	depends on I2C
 
 config SND_SOC_RT5677
 	tristate
+	depends on I2C
 	select REGMAP_I2C
 	select REGMAP_IRQ
 
···
 
 config SND_SOC_RT5682
 	tristate
+	depends on I2C || SOUNDWIRE
+
+config SND_SOC_RT5682_SDW
+	tristate "Realtek RT5682 Codec - SDW"
+	depends on SOUNDWIRE
+	select SND_SOC_RT5682
+	select REGMAP_SOUNDWIRE
 
 config SND_SOC_RT700
 	tristate
···
 
 config SND_SOC_SSM2518
 	tristate
+	depends on I2C
 
 config SND_SOC_SSM2602
 	tristate
···
 
 config SND_SOC_STA529
 	tristate
+	depends on I2C
 
 config SND_SOC_STAC9766
 	tristate
+	depends on SND_SOC_AC97_BUS
 
 config SND_SOC_STI_SAS
 	tristate "codec Audio support for STI SAS codec"
···
 
 config SND_SOC_TLV320DAC33
 	tristate
+	depends on I2C
+
+config SND_SOC_TLV320ADCX140
+	tristate "Texas Instruments TLV320ADCX140 CODEC family"
+	depends on I2C
+	select REGMAP_I2C
+	help
+	  Add support for Texas Instruments tlv320adc3140, tlv320adc5140 and
+	  tlv320adc6140 quad channel ADCs.
 
 config SND_SOC_TS3A227E
 	tristate "TI Headset/Mic detect and keypress chip"
···
 	  Add support for Tempo Semiconductor's TSCS454 audio CODEC.
 
 config SND_SOC_TWL4030
-	select MFD_TWL4030_AUDIO
 	tristate
+	depends on TWL4030_CORE
+	select MFD_TWL4030_AUDIO
 
 config SND_SOC_TWL6040
 	tristate
+	depends on TWL6040_CORE
 
 config SND_SOC_UDA1334
 	tristate "NXP UDA1334 DAC"
···
 
 config SND_SOC_WM0010
 	tristate
+	depends on SPI_MASTER
 
 config SND_SOC_WM1250_EV1
 	tristate
+	depends on I2C
 
 config SND_SOC_WM2000
 	tristate
+	depends on I2C
 
 config SND_SOC_WM2200
 	tristate
+	depends on I2C
 
 config SND_SOC_WM5100
 	tristate
+	depends on I2C
 
 config SND_SOC_WM5102
 	tristate
+	depends on MFD_WM5102
 
 config SND_SOC_WM5110
 	tristate
+	depends on MFD_WM5110
 
 config SND_SOC_WM8350
 	tristate
+	depends on MFD_WM8350
 
 config SND_SOC_WM8400
 	tristate
+	# FIXME nothing selects SND_SOC_WM8400??
+	depends on MFD_WM8400
 
 config SND_SOC_WM8510
 	tristate "Wolfson Microelectronics WM8510 CODEC"
···
 
 config SND_SOC_WM8940
 	tristate
+	depends on I2C
 
 config SND_SOC_WM8955
 	tristate
+	depends on I2C
 
 config SND_SOC_WM8960
 	tristate "Wolfson Microelectronics WM8960 CODEC"
···
 
 config SND_SOC_WM8961
 	tristate
+	depends on I2C
 
 config SND_SOC_WM8962
 	tristate "Wolfson Microelectronics WM8962 CODEC"
···
 
 config SND_SOC_WM8971
 	tristate
+	depends on I2C
 
 config SND_SOC_WM8974
 	tristate "Wolfson Microelectronics WM8974 codec"
···
 
 config SND_SOC_WM8983
 	tristate
+	depends on I2C
 
 config SND_SOC_WM8985
 	tristate "Wolfson Microelectronics WM8985 and WM8758 codec driver"
···
 
 config SND_SOC_WM8990
 	tristate
+	depends on I2C
 
 config SND_SOC_WM8991
 	tristate
+	depends on I2C
 
 config SND_SOC_WM8993
 	tristate
+	depends on I2C
 
 config SND_SOC_WM8994
 	tristate
···
 
 config SND_SOC_WM8996
 	tristate
+	depends on I2C
 
 config SND_SOC_WM8997
 	tristate
+	depends on MFD_WM8997
 
 config SND_SOC_WM8998
 	tristate
+	depends on MFD_WM8998
 
 config SND_SOC_WM9081
 	tristate
···
 
 config SND_SOC_WM9090
 	tristate
+	depends on I2C
 
 config SND_SOC_WM9705
 	tristate
+	depends on SND_SOC_AC97_BUS
 	select REGMAP_AC97
 	select AC97_BUS_COMPAT if AC97_BUS_NEW
 
 config SND_SOC_WM9712
 	tristate
+	depends on SND_SOC_AC97_BUS
 	select REGMAP_AC97
 	select AC97_BUS_COMPAT if AC97_BUS_NEW
 
 config SND_SOC_WM9713
 	tristate
+	depends on SND_SOC_AC97_BUS
 	select REGMAP_AC97
 	select AC97_BUS_COMPAT if AC97_BUS_NEW
 
···
 # Amp
 config SND_SOC_LM4857
 	tristate
+	depends on I2C
 
 config SND_SOC_MAX9759
 	tristate "Maxim MAX9759 speaker Amplifier"
···
 
 config SND_SOC_MAX9768
 	tristate
+	depends on I2C
 
 config SND_SOC_MAX9877
 	tristate
+	depends on I2C
 
 config SND_SOC_MC13783
 	tristate
+	depends on MFD_MC13XXX
 
 config SND_SOC_ML26124
 	tristate
+	depends on I2C
 
 config SND_SOC_MT6351
 	tristate "MediaTek MT6351 Codec"
···
 
 config SND_SOC_NAU8825
 	tristate
+	depends on I2C
 
 config SND_SOC_TPA6130A2
 	tristate "Texas Instruments TPA6130A2 headphone amplifier"

sound/soc/codecs/Makefile

···
 snd-soc-rt5677-objs := rt5677.o
 snd-soc-rt5677-spi-objs := rt5677-spi.o
 snd-soc-rt5682-objs := rt5682.o
+snd-soc-rt5682-sdw-objs := rt5682-sdw.o
 snd-soc-rt700-objs := rt700.o rt700-sdw.o
 snd-soc-rt711-objs := rt711.o rt711-sdw.o
 snd-soc-rt715-objs := rt715.o rt715-sdw.o
···
 snd-soc-tlv320aic32x4-spi-objs := tlv320aic32x4-spi.o
 snd-soc-tlv320aic3x-objs := tlv320aic3x.o
 snd-soc-tlv320dac33-objs := tlv320dac33.o
+snd-soc-tlv320adcx140-objs := tlv320adcx140.o
 snd-soc-tscs42xx-objs := tscs42xx.o
 snd-soc-tscs454-objs := tscs454.o
 snd-soc-ts3a227e-objs := ts3a227e.o
···
 obj-$(CONFIG_SND_SOC_RT5677)	+= snd-soc-rt5677.o
 obj-$(CONFIG_SND_SOC_RT5677_SPI)	+= snd-soc-rt5677-spi.o
 obj-$(CONFIG_SND_SOC_RT5682)	+= snd-soc-rt5682.o
+obj-$(CONFIG_SND_SOC_RT5682_SDW)	+= snd-soc-rt5682-sdw.o
 obj-$(CONFIG_SND_SOC_RT700)     += snd-soc-rt700.o
 obj-$(CONFIG_SND_SOC_RT711)     += snd-soc-rt711.o
 obj-$(CONFIG_SND_SOC_RT715)     += snd-soc-rt715.o
···
 obj-$(CONFIG_SND_SOC_TLV320AIC32X4_SPI)	+= snd-soc-tlv320aic32x4-spi.o
 obj-$(CONFIG_SND_SOC_TLV320AIC3X)	+= snd-soc-tlv320aic3x.o
 obj-$(CONFIG_SND_SOC_TLV320DAC33)	+= snd-soc-tlv320dac33.o
+obj-$(CONFIG_SND_SOC_TLV320ADCX140)	+= snd-soc-tlv320adcx140.o
 obj-$(CONFIG_SND_SOC_TSCS42XX)	+= snd-soc-tscs42xx.o
 obj-$(CONFIG_SND_SOC_TSCS454)	+= snd-soc-tscs454.o
 obj-$(CONFIG_SND_SOC_TS3A227E)	+= snd-soc-ts3a227e.o

sound/soc/codecs/cros_ec_codec.c

···
 	/* DMIC */
 	atomic_t dmic_probed;
 
+	/* I2S_RX */
+	uint32_t i2s_rx_bclk_ratio;
+
 	/* WoV */
 	bool wov_enabled;
 	uint8_t *wov_audio_shm_p;
···
 		snd_soc_component_get_drvdata(component);
 	struct ec_param_ec_codec_i2s_rx p;
 	enum ec_codec_i2s_rx_sample_depth depth;
+	uint32_t bclk;
 	int ret;
 
 	if (params_rate(params) != 48000)
···
 	if (ret < 0)
 		return ret;
 
-	dev_dbg(component->dev, "set bclk to %u\n",
-		snd_soc_params_to_bclk(params));
+	if (priv->i2s_rx_bclk_ratio)
+		bclk = params_rate(params) * priv->i2s_rx_bclk_ratio;
+	else
+		bclk = snd_soc_params_to_bclk(params);
+
+	dev_dbg(component->dev, "set bclk to %u\n", bclk);
 
 	p.cmd = EC_CODEC_I2S_RX_SET_BCLK;
-	p.set_bclk_param.bclk = snd_soc_params_to_bclk(params);
+	p.set_bclk_param.bclk = bclk;
 	return send_ec_host_command(priv->ec_device, EC_CMD_EC_CODEC_I2S_RX,
 				    (uint8_t *)&p, sizeof(p), NULL, 0);
+}
+
+static int i2s_rx_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio)
+{
+	struct snd_soc_component *component = dai->component;
+	struct cros_ec_codec_priv *priv =
+		snd_soc_component_get_drvdata(component);
+
+	priv->i2s_rx_bclk_ratio = ratio;
+	return 0;
 }
 
 static int i2s_rx_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
···
 static const struct snd_soc_dai_ops i2s_rx_dai_ops = {
 	.hw_params = i2s_rx_hw_params,
 	.set_fmt = i2s_rx_set_fmt,
+	.set_bclk_ratio = i2s_rx_set_bclk_ratio,
 };
 
 static int i2s_rx_event(struct snd_soc_dapm_widget *w,

sound/soc/codecs/cs4271.c

···
 		 */
 
 		if ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
-		     !dai->capture_active) ||
+		     !dai->stream_active[SNDRV_PCM_STREAM_CAPTURE]) ||
 		    (substream->stream == SNDRV_PCM_STREAM_CAPTURE &&
-		     !dai->playback_active)) {
+		     !dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK])) {
 			ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2,
 						 CS4271_MODE2_PDN,
 						 CS4271_MODE2_PDN);

sound/soc/codecs/cs47l15.c

···
 	struct madera *madera = priv->madera;
 	int n_adsp;
 
-	if (strcmp(rtd->codec_dai->name, "cs47l15-dsp-trace") == 0) {
+	if (strcmp(asoc_rtd_to_codec(rtd, 0)->name, "cs47l15-dsp-trace") == 0) {
 		n_adsp = 0;
 	} else {
 		dev_err(madera->dev,
 			"No suitable compressed stream for DAI '%s'\n",
-			rtd->codec_dai->name);
+			asoc_rtd_to_codec(rtd, 0)->name);
 		return -EINVAL;
 	}
 

sound/soc/codecs/cs47l24.c

···
 	struct arizona *arizona = priv->core.arizona;
 	int n_adsp;
 
-	if (strcmp(rtd->codec_dai->name, "cs47l24-dsp-voicectrl") == 0) {
+	if (strcmp(asoc_rtd_to_codec(rtd, 0)->name, "cs47l24-dsp-voicectrl") == 0) {
 		n_adsp = 2;
-	} else if (strcmp(rtd->codec_dai->name, "cs47l24-dsp-trace") == 0) {
+	} else if (strcmp(asoc_rtd_to_codec(rtd, 0)->name, "cs47l24-dsp-trace") == 0) {
 		n_adsp = 1;
 	} else {
 		dev_err(arizona->dev,
 			"No suitable compressed stream for DAI '%s'\n",
-			rtd->codec_dai->name);
+			asoc_rtd_to_codec(rtd, 0)->name);
 		return -EINVAL;
 	}
 

sound/soc/codecs/cs47l35.c

···
 	struct madera *madera = priv->madera;
 	int n_adsp;
 
-	if (strcmp(rtd->codec_dai->name, "cs47l35-dsp-voicectrl") == 0) {
+	if (strcmp(asoc_rtd_to_codec(rtd, 0)->name, "cs47l35-dsp-voicectrl") == 0) {
 		n_adsp = 2;
-	} else if (strcmp(rtd->codec_dai->name, "cs47l35-dsp-trace") == 0) {
+	} else if (strcmp(asoc_rtd_to_codec(rtd, 0)->name, "cs47l35-dsp-trace") == 0) {
 		n_adsp = 0;
 	} else {
 		dev_err(madera->dev,
 			"No suitable compressed stream for DAI '%s'\n",
-			rtd->codec_dai->name);
+			asoc_rtd_to_codec(rtd, 0)->name);
 		return -EINVAL;
 	}
 

sound/soc/codecs/cs47l85.c

···
 	struct madera *madera = priv->madera;
 	int n_adsp;
 
-	if (strcmp(rtd->codec_dai->name, "cs47l85-dsp-voicectrl") == 0) {
+	if (strcmp(asoc_rtd_to_codec(rtd, 0)->name, "cs47l85-dsp-voicectrl") == 0) {
 		n_adsp = 5;
-	} else if (strcmp(rtd->codec_dai->name, "cs47l85-dsp-trace") == 0) {
+	} else if (strcmp(asoc_rtd_to_codec(rtd, 0)->name, "cs47l85-dsp-trace") == 0) {
 		n_adsp = 0;
 	} else {
 		dev_err(madera->dev,
 			"No suitable compressed stream for DAI '%s'\n",
-			rtd->codec_dai->name);
+			asoc_rtd_to_codec(rtd, 0)->name);
 		return -EINVAL;
 	}
 

sound/soc/codecs/cs47l90.c

···
 	struct madera *madera = priv->madera;
 	int n_adsp;
 
-	if (strcmp(rtd->codec_dai->name, "cs47l90-dsp-voicectrl") == 0) {
+	if (strcmp(asoc_rtd_to_codec(rtd, 0)->name, "cs47l90-dsp-voicectrl") == 0) {
 		n_adsp = 5;
-	} else if (strcmp(rtd->codec_dai->name, "cs47l90-dsp-trace") == 0) {
+	} else if (strcmp(asoc_rtd_to_codec(rtd, 0)->name, "cs47l90-dsp-trace") == 0) {
 		n_adsp = 0;
 	} else {
 		dev_err(madera->dev,
 			"No suitable compressed stream for DAI '%s'\n",
-			rtd->codec_dai->name);
+			asoc_rtd_to_codec(rtd, 0)->name);
 		return -EINVAL;
 	}
 

sound/soc/codecs/cs47l92.c

···
 	struct madera *madera = priv->madera;
 	int n_adsp;
 
-	if (strcmp(rtd->codec_dai->name, "cs47l92-dsp-trace") == 0) {
+	if (strcmp(asoc_rtd_to_codec(rtd, 0)->name, "cs47l92-dsp-trace") == 0) {
 		n_adsp = 0;
 	} else {
 		dev_err(madera->dev,
 			"No suitable compressed stream for DAI '%s'\n",
-			rtd->codec_dai->name);
+			asoc_rtd_to_codec(rtd, 0)->name);
 		return -EINVAL;
 	}
 

sound/soc/codecs/hdac_hdmi.c

···
 
 static int hdac_hdmi_dev_probe(struct hdac_device *hdev)
 {
-	struct hdac_hdmi_priv *hdmi_priv = NULL;
+	struct hdac_hdmi_priv *hdmi_priv;
 	struct snd_soc_dai_driver *hdmi_dais = NULL;
-	struct hdac_ext_link *hlink = NULL;
+	struct hdac_ext_link *hlink;
 	int num_dais = 0;
-	int ret = 0;
+	int ret;
 	struct hdac_driver *hdrv = drv_to_hdac_driver(hdev->dev.driver);
 	const struct hda_device_id *hdac_id = hdac_get_device_id(hdev, hdrv);
 

sound/soc/codecs/max98357a.c

···
  */
 
 #include <linux/acpi.h>
+#include <linux/delay.h>
 #include <linux/device.h>
 #include <linux/err.h>
 #include <linux/gpio.h>
···
 	unsigned int sdmode_delay;
 };
 
-static int max98357a_daiops_trigger(struct snd_pcm_substream *substream,
-		int cmd, struct snd_soc_dai *dai)
+static int max98357a_sdmode_event(struct snd_soc_dapm_widget *w,
+		struct snd_kcontrol *kcontrol, int event)
 {
-	struct max98357a_priv *max98357a = snd_soc_dai_get_drvdata(dai);
+	struct snd_soc_component *component =
+		snd_soc_dapm_to_component(w->dapm);
+	struct max98357a_priv *max98357a =
+		snd_soc_component_get_drvdata(component);
 
 	if (!max98357a->sdmode)
 		return 0;
 
-	switch (cmd) {
-	case SNDRV_PCM_TRIGGER_START:
-	case SNDRV_PCM_TRIGGER_RESUME:
-	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-		mdelay(max98357a->sdmode_delay);
+	if (event & SND_SOC_DAPM_POST_PMU) {
+		msleep(max98357a->sdmode_delay);
 		gpiod_set_value(max98357a->sdmode, 1);
-		break;
-	case SNDRV_PCM_TRIGGER_STOP:
-	case SNDRV_PCM_TRIGGER_SUSPEND:
-	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		dev_dbg(component->dev, "set sdmode to 1");
+	} else if (event & SND_SOC_DAPM_PRE_PMD) {
 		gpiod_set_value(max98357a->sdmode, 0);
-		break;
+		dev_dbg(component->dev, "set sdmode to 0");
 	}
 
 	return 0;
···
 
 static const struct snd_soc_dapm_widget max98357a_dapm_widgets[] = {
 	SND_SOC_DAPM_OUTPUT("Speaker"),
+	SND_SOC_DAPM_OUT_DRV_E("SD_MODE", SND_SOC_NOPM, 0, 0, NULL, 0,
+			max98357a_sdmode_event,
+			SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
 };
 
 static const struct snd_soc_dapm_route max98357a_dapm_routes[] = {
-	{"Speaker", NULL, "HiFi Playback"},
+	{"SD_MODE", NULL, "HiFi Playback"},
+	{"Speaker", NULL, "SD_MODE"},
 };
 
 static const struct snd_soc_component_driver max98357a_component_driver = {
···
 	.use_pmdown_time	= 1,
 	.endianness		= 1,
 	.non_legacy_dai_naming	= 1,
-};
-
-static const struct snd_soc_dai_ops max98357a_dai_ops = {
-	.trigger	= max98357a_daiops_trigger,
 };
 
 static struct snd_soc_dai_driver max98357a_dai_driver = {
···
 		.channels_min	= 1,
 		.channels_max	= 2,
 	},
-	.ops    = &max98357a_dai_ops,
 };
 
 static int max98357a_platform_probe(struct platform_device *pdev)
···
 #ifdef CONFIG_ACPI
 static const struct acpi_device_id max98357a_acpi_match[] = {
 	{ "MX98357A", 0 },
+	{ "MX98360A", 0 },
 	{},
 };
 MODULE_DEVICE_TABLE(acpi, max98357a_acpi_match);

sound/soc/codecs/mt6660.c

···
-// SPDX-License-Identifier: GPL-2.0 //
+// SPDX-License-Identifier: GPL-2.0
 
 // Copyright (c) 2019 MediaTek Inc.
 
 #include <linux/module.h>
 #include <linux/kernel.h>
-#include <linux/version.h>
 #include <linux/err.h>
 #include <linux/i2c.h>
 #include <linux/pm_runtime.h>
 #include <linux/delay.h>
-#include <linux/debugfs.h>
 #include <sound/soc.h>
 #include <sound/tlv.h>
 #include <sound/pcm_params.h>
···
 				 0x01, on_off ? 0x00 : 0x01);
 }
 
+struct reg_table {
+	uint32_t addr;
+	uint32_t mask;
+	uint32_t val;
+};
+
+static const struct reg_table mt6660_setting_table[] = {
+	{ 0x20, 0x80, 0x00 },
+	{ 0x30, 0x01, 0x00 },
+	{ 0x50, 0x1c, 0x04 },
+	{ 0xB1, 0x0c, 0x00 },
+	{ 0xD3, 0x03, 0x03 },
+	{ 0xE0, 0x01, 0x00 },
+	{ 0x98, 0x44, 0x04 },
+	{ 0xB9, 0xff, 0x82 },
+	{ 0xB7, 0x7777, 0x7273 },
+	{ 0xB6, 0x07, 0x03 },
+	{ 0x6B, 0xe0, 0x20 },
+	{ 0x07, 0xff, 0x70 },
+	{ 0xBB, 0xff, 0x20 },
+	{ 0x69, 0xff, 0x40 },
+	{ 0xBD, 0xffff, 0x17f8 },
+	{ 0x70, 0xff, 0x15 },
+	{ 0x7C, 0xff, 0x00 },
+	{ 0x46, 0xff, 0x1d },
+	{ 0x1A, 0xffffffff, 0x7fdb7ffe },
+	{ 0x1B, 0xffffffff, 0x7fdb7ffe },
+	{ 0x51, 0xff, 0x58 },
+	{ 0xA2, 0xff, 0xce },
+	{ 0x33, 0xffff, 0x7fff },
+	{ 0x4C, 0xffff, 0x0116 },
+	{ 0x16, 0x1800, 0x0800 },
+	{ 0x68, 0x1f, 0x07 },
+};
+
+static int mt6660_component_setting(struct snd_soc_component *component)
+{
+	struct mt6660_chip *chip = snd_soc_component_get_drvdata(component);
+	int ret = 0;
+	size_t i = 0;
+
+	ret = _mt6660_chip_power_on(chip, 1);
+	if (ret < 0) {
+		dev_err(component->dev, "%s chip power on failed\n", __func__);
+		return ret;
+	}
+
+	for (i = 0; i < ARRAY_SIZE(mt6660_setting_table); i++) {
+		ret = snd_soc_component_update_bits(component,
+				mt6660_setting_table[i].addr,
+				mt6660_setting_table[i].mask,
+				mt6660_setting_table[i].val);
+		if (ret < 0) {
+			dev_err(component->dev, "%s update 0x%02x failed\n",
+				__func__, mt6660_setting_table[i].addr);
+			return ret;
+		}
+	}
+
+	ret = _mt6660_chip_power_on(chip, 0);
+	if (ret < 0) {
+		dev_err(component->dev, "%s chip power off failed\n", __func__);
+		return ret;
+	}
+
+	return 0;
+}
+
 static int mt6660_component_probe(struct snd_soc_component *component)
 {
 	struct mt6660_chip *chip = snd_soc_component_get_drvdata(component);
+	int ret;
 
 	dev_dbg(component->dev, "%s\n", __func__);
 	snd_soc_component_init_regmap(component, chip->regmap);
 
-	return 0;
+	ret = mt6660_component_setting(component);
+	if (ret < 0)
+		dev_err(chip->dev, "mt6660 component setting failed\n");
+
+	return ret;
 }
 
 static void mt6660_component_remove(struct snd_soc_component *component)
···
 MODULE_AUTHOR("Jeff Chang <[e-mail]>");
 MODULE_DESCRIPTION("MT6660 SPKAMP Driver");
 MODULE_LICENSE("GPL");
-MODULE_VERSION("1.0.7_G");
+MODULE_VERSION("1.0.8_G");

sound/soc/codecs/rk3328_codec.c

···
 #include <linux/clk.h>
 #include <linux/delay.h>
 #include <linux/device.h>
+#include <linux/gpio/consumer.h>
 #include <linux/module.h>
 #include <linux/of.h>
 #include <linux/platform_device.h>
···
 
 struct rk3328_codec_priv {
 	struct regmap *regmap;
-	struct regmap *grf;
+	struct gpio_desc *mute;
 	struct clk *mclk;
 	struct clk *pclk;
 	unsigned int sclk;
···
 			   DAC_MODE_MASK, val);
 
 	return 0;
-}
-
-static void rk3328_analog_output(struct rk3328_codec_priv *rk3328, int mute)
-{
-	unsigned int val = BIT(17);
-
-	if (mute)
-		val |= BIT(1);
-
-	regmap_write(rk3328->grf, RK3328_GRF_SOC_CON10, val);
 }
 
 static int rk3328_digital_mute(struct snd_soc_dai *dai, int mute)
···
 	}
 
 	msleep(rk3328->spk_depop_time);
-	rk3328_analog_output(rk3328, 1);
+	gpiod_set_value(rk3328->mute, 0);
 
 	regmap_update_bits(rk3328->regmap, HPOUTL_GAIN_CTRL,
 			   HPOUTL_GAIN_MASK, OUT_VOLUME);
···
 {
 	size_t i;
 
-	rk3328_analog_output(rk3328, 0);
+	gpiod_set_value(rk3328->mute, 1);
 
 	regmap_update_bits(rk3328->regmap, HPOUTL_GAIN_CTRL,
 			   HPOUTL_GAIN_MASK, 0);
···
 		dev_err(&pdev->dev, "missing 'rockchip,grf'\n");
 		return PTR_ERR(grf);
 	}
-	rk3328->grf = grf;
 	/* enable i2s_acodec_en */
 	regmap_write(grf, RK3328_GRF_SOC_CON2,
 		     (BIT(14) << 16 | BIT(14)));
···
 		rk3328->spk_depop_time = 200;
 	}
 
-	rk3328_analog_output(rk3328, 0);
+	rk3328->mute = gpiod_get_optional(&pdev->dev, "mute", GPIOD_OUT_HIGH);
+	if (IS_ERR(rk3328->mute))
+		return PTR_ERR(rk3328->mute);
+	/*
+	 * Rock64 is the only supported platform to have widely relied on
+	 * this; if we do happen to come across an old DTB, just leave the
+	 * external mute forced off.
+	 */
+	if (!rk3328->mute && of_machine_is_compatible("pine64,rock64")) {
+		dev_warn(&pdev->dev, "assuming implicit control of GPIO_MUTE; update devicetree if possible\n");
+		regmap_write(grf, RK3328_GRF_SOC_CON10, BIT(17) | BIT(1));
+	}
 
 	rk3328->mclk = devm_clk_get(&pdev->dev, "mclk");
 	if (IS_ERR(rk3328->mclk))

sound/soc/codecs/rl6231.c

···
 static const struct pll_calc_map pll_preset_table[] = {
 	{19200000,  4096000,  23, 14, 1, false},
 	{19200000,  24576000,  3, 30, 3, false},
+	{3840000,   24576000,  3, 30, 0, true},
 };
 
 static unsigned int find_best_div(unsigned int in,

sound/soc/codecs/rl6231.h

···
 #ifndef __RL6231_H__
 #define __RL6231_H__
 
-#define RL6231_PLL_INP_MAX	40000000
+#define RL6231_PLL_INP_MAX	50000000
 #define RL6231_PLL_INP_MIN	256000
 #define RL6231_PLL_N_MAX	0x1ff
 #define RL6231_PLL_K_MAX	0x1f

sound/soc/codecs/rt1015.c

···
 	return 0;
 }
 
-static int rt5518_bypass_boost_get(struct snd_kcontrol *kcontrol,
+static int rt1015_bypass_boost_get(struct snd_kcontrol *kcontrol,
 		struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_component *component =
···
 	return 0;
 }
 
-static int rt5518_bypass_boost_put(struct snd_kcontrol *kcontrol,
+static int rt1015_bypass_boost_put(struct snd_kcontrol *kcontrol,
 		struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_component *component =
···
 		rt1015_boost_mode_get, rt1015_boost_mode_put),
 	SOC_ENUM("Mono LR Select", rt1015_mono_lr_sel),
 	SOC_SINGLE_EXT("Bypass Boost", SND_SOC_NOPM, 0, 1, 0,
-		rt5518_bypass_boost_get, rt5518_bypass_boost_put),
+		rt1015_bypass_boost_get, rt1015_bypass_boost_put),
 };
 
 static int rt1015_is_sys_clk_from_pll(struct snd_soc_dapm_widget *source,
···
 #define RT1015_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
 			SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8)
 
-struct snd_soc_dai_ops rt1015_aif_dai_ops = {
+static struct snd_soc_dai_ops rt1015_aif_dai_ops = {
 	.hw_params = rt1015_hw_params,
 	.set_fmt = rt1015_set_dai_fmt,
 };
 
-struct snd_soc_dai_driver rt1015_dai[] = {
+static struct snd_soc_dai_driver rt1015_dai[] = {
 	{
 		.name = "rt1015-aif",
 		.id = 0,

sound/soc/codecs/rt1308-sdw.c

···
 	kfree(stream);
 }
 
+static int rt1308_sdw_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 rt1308_sdw_priv *rt1308 =
+		snd_soc_component_get_drvdata(component);
+
+	if (tx_mask)
+		return -EINVAL;
+
+	if (slots > 2)
+		return -EINVAL;
+
+	rt1308->rx_mask = rx_mask;
+	rt1308->slots = slots;
+	/* slot_width is not used since it's irrelevant for SoundWire */
+
+	return 0;
+}
+
 static int rt1308_sdw_hw_params(struct snd_pcm_substream *substream,
 	struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
 {
···
 	struct sdw_port_config port_config;
 	enum sdw_data_direction direction;
 	struct sdw_stream_data *stream;
-	int retval, port, num_channels;
+	int retval, port, num_channels, ch_mask;
 
 	dev_dbg(dai->dev, "%s %s", __func__, dai->name);
 	stream = snd_soc_dai_get_dma_data(dai, substream);
···
 		return -EINVAL;
 	}
 
+	if (rt1308->slots) {
+		num_channels = rt1308->slots;
+		ch_mask = rt1308->rx_mask;
+	} else {
+		num_channels = params_channels(params);
+		ch_mask = (1 << num_channels) - 1;
+	}
+
 	stream_config.frame_rate = params_rate(params);
-	stream_config.ch_count = params_channels(params);
+	stream_config.ch_count = num_channels;
 	stream_config.bps = snd_pcm_format_width(params_format(params));
 	stream_config.direction = direction;
 
-	num_channels = params_channels(params);
-	port_config.ch_mask = (1 << (num_channels)) - 1;
+	port_config.ch_mask = ch_mask;
 	port_config.num = port;
 
 	retval = sdw_stream_add_slave(rt1308->sdw_slave, &stream_config,
···
 	.hw_free	= rt1308_sdw_pcm_hw_free,
 	.set_sdw_stream	= rt1308_set_sdw_stream,
 	.shutdown	= rt1308_sdw_shutdown,
+	.set_tdm_slot	= rt1308_sdw_set_tdm_slot,
 };
 
 #define RT1308_STEREO_RATES SNDRV_PCM_RATE_48000

sound/soc/codecs/rt1308-sdw.h

···
 	struct sdw_bus_params params;
 	bool hw_init;
 	bool first_hw_init;
+	int rx_mask;
+	int slots;
 };
 
 struct sdw_stream_data {

sound/soc/codecs/rt5659.c

···
 {
 	struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm);
 	struct rt5659_priv *rt5659 = snd_soc_component_get_drvdata(component);
-	int pd, idx = -EINVAL;
+	int pd, idx;
 
 	pd = rl6231_get_pre_div(rt5659->regmap,
 		RT5659_ADDA_CLK_1, RT5659_I2S_PD1_SFT);

sound/soc/codecs/rt5682-sdw.c

···
+// SPDX-License-Identifier: GPL-2.0-only
+//
+// rt5682-sdw.c  --  RT5682 ALSA SoC audio component driver
+//
+// Copyright 2019 Realtek Semiconductor Corp.
+// Author: Oder Chiou <[e-mail]>
+//
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/acpi.h>
+#include <linux/gpio.h>
+#include <linux/of_gpio.h>
+#include <linux/regulator/consumer.h>
+#include <linux/mutex.h>
+#include <linux/soundwire/sdw.h>
+#include <linux/soundwire/sdw_type.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/jack.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+
+#include "rt5682.h"
+#include "rt5682-sdw.h"
+
+static bool rt5682_sdw_readable_register(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case 0x00e0:
+	case 0x00f0:
+	case 0x3000:
+	case 0x3001:
+	case 0x3004:
+	case 0x3005:
+	case 0x3008:
+		return true;
+	default:
+		return false;
+	}
+}
+
+const struct regmap_config rt5682_sdw_regmap = {
+	.name = "sdw",
+	.reg_bits = 32,
+	.val_bits = 8,
+	.max_register = RT5682_I2C_MODE,
+	.readable_reg = rt5682_sdw_readable_register,
+	.cache_type = REGCACHE_NONE,
+	.use_single_read = true,
+	.use_single_write = true,
+};
+
+static int rt5682_update_status(struct sdw_slave *slave,
+					enum sdw_slave_status status)
+{
+	struct rt5682_priv *rt5682 = dev_get_drvdata(&slave->dev);
+
+	/* Update the status */
+	rt5682->status = status;
+
+	if (status == SDW_SLAVE_UNATTACHED)
+		rt5682->hw_init = false;
+
+	/*
+	 * Perform initialization only if slave status is present and
+	 * hw_init flag is false
+	 */
+	if (rt5682->hw_init || rt5682->status != SDW_SLAVE_ATTACHED)
+		return 0;
+
+	/* perform I/O transfers required for Slave initialization */
+	return rt5682_io_init(&slave->dev, slave);
+}
+
+static int rt5682_read_prop(struct sdw_slave *slave)
+{
+	struct sdw_slave_prop *prop = &slave->prop;
+	int nval, i, num_of_ports = 1;
+	u32 bit;
+	unsigned long addr;
+	struct sdw_dpn_prop *dpn;
+
+	prop->paging_support = false;
+
+	/* first we need to allocate memory for set bits in port lists */
+	prop->source_ports = 0x4;	/* BITMAP: 00000100 */
+	prop->sink_ports = 0x2;		/* BITMAP: 00000010 */
+
+	nval = hweight32(prop->source_ports);
+	num_of_ports += nval;
+	prop->src_dpn_prop = devm_kcalloc(&slave->dev, nval,
+					  sizeof(*prop->src_dpn_prop),
+					  GFP_KERNEL);
+	if (!prop->src_dpn_prop)
+		return -ENOMEM;
+
+	i = 0;
+	dpn = prop->src_dpn_prop;
+	addr = prop->source_ports;
+	for_each_set_bit(bit, &addr, 32) {
+		dpn[i].num = bit;
+		dpn[i].type = SDW_DPN_FULL;
+		dpn[i].simple_ch_prep_sm = true;
+		dpn[i].ch_prep_timeout = 10;
+		i++;
+	}
+
+	/* do this again for sink now */
+	nval = hweight32(prop->sink_ports);
+	num_of_ports += nval;
+	prop->sink_dpn_prop = devm_kcalloc(&slave->dev, nval,
+					   sizeof(*prop->sink_dpn_prop),
+					   GFP_KERNEL);
+	if (!prop->sink_dpn_prop)
+		return -ENOMEM;
+
+	i = 0;
+	dpn = prop->sink_dpn_prop;
+	addr = prop->sink_ports;
+	for_each_set_bit(bit, &addr, 32) {
+		dpn[i].num = bit;
+		dpn[i].type = SDW_DPN_FULL;
+		dpn[i].simple_ch_prep_sm = true;
+		dpn[i].ch_prep_timeout = 10;
+		i++;
+	}
+
+	/* Allocate port_ready based on num_of_ports */
+	slave->port_ready = devm_kcalloc(&slave->dev, num_of_ports,
+					 sizeof(*slave->port_ready),
+					 GFP_KERNEL);
+	if (!slave->port_ready)
+		return -ENOMEM;
+
+	/* Initialize completion */
+	for (i = 0; i < num_of_ports; i++)
+		init_completion(&slave->port_ready[i]);
+
+	/* set the timeout values */
+	prop->clk_stop_timeout = 20;
+
+	/* wake-up event */
+	prop->wake_capable = 1;
+
+	return 0;
+}
+
+/* Bus clock frequency */
+#define RT5682_CLK_FREQ_9600000HZ 9600000
+#define RT5682_CLK_FREQ_12000000HZ 12000000
+#define RT5682_CLK_FREQ_6000000HZ 6000000
+#define RT5682_CLK_FREQ_4800000HZ 4800000
+#define RT5682_CLK_FREQ_2400000HZ 2400000
+#define RT5682_CLK_FREQ_12288000HZ 12288000
+
+static int rt5682_clock_config(struct device *dev)
+{
+	struct rt5682_priv *rt5682 = dev_get_drvdata(dev);
+	unsigned int clk_freq, value;
+
+	clk_freq = (rt5682->params.curr_dr_freq >> 1);
+
+	switch (clk_freq) {
+	case RT5682_CLK_FREQ_12000000HZ:
+		value = 0x0;
+		break;
+	case RT5682_CLK_FREQ_6000000HZ:
+		value = 0x1;
+		break;
+	case RT5682_CLK_FREQ_9600000HZ:
+		value = 0x2;
+		break;
+	case RT5682_CLK_FREQ_4800000HZ:
+		value = 0x3;
+		break;
+	case RT5682_CLK_FREQ_2400000HZ:
+		value = 0x4;
+		break;
+	case RT5682_CLK_FREQ_12288000HZ:
+		value = 0x5;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	regmap_write(rt5682->sdw_regmap, 0xe0, value);
+	regmap_write(rt5682->sdw_regmap, 0xf0, value);
+
+	dev_dbg(dev, "%s complete, clk_freq=%d\n", __func__, clk_freq);
+
+	return 0;
+}
+
+static int rt5682_bus_config(struct sdw_slave *slave,
+					struct sdw_bus_params *params)
+{
+	struct rt5682_priv *rt5682 = dev_get_drvdata(&slave->dev);
+	int ret;
+
+	memcpy(&rt5682->params, params, sizeof(*params));
+
+	ret = rt5682_clock_config(&slave->dev);
+	if (ret < 0)
+		dev_err(&slave->dev, "Invalid clk config");
+
+	return ret;
+}
+
+static int rt5682_interrupt_callback(struct sdw_slave *slave,
+					struct sdw_slave_intr_status *status)
+{
+	struct rt5682_priv *rt5682 = dev_get_drvdata(&slave->dev);
+
+	dev_dbg(&slave->dev,
+		"%s control_port_stat=%x", __func__, status->control_port);
+
+	if (status->control_port & 0x4) {
+		mod_delayed_work(system_power_efficient_wq,
+			&rt5682->jack_detect_work, msecs_to_jiffies(250));
+	}
+
+	return 0;
+}
+
+static struct sdw_slave_ops rt5682_slave_ops = {
+	.read_prop = rt5682_read_prop,
+	.interrupt_callback = rt5682_interrupt_callback,
+	.update_status = rt5682_update_status,
+	.bus_config = rt5682_bus_config,
+};
+
+static int rt5682_sdw_probe(struct sdw_slave *slave,
+			   const struct sdw_device_id *id)
+{
+	struct regmap *regmap;
+
+	/* Assign ops */
+	slave->ops = &rt5682_slave_ops;
+
+	/* Regmap Initialization */
+	regmap = devm_regmap_init_sdw(slave, &rt5682_sdw_regmap);
+	if (IS_ERR(regmap))
+		return -EINVAL;
+
+	rt5682_sdw_init(&slave->dev, regmap, slave);
+
+	return 0;
+}
+
+static int rt5682_sdw_remove(struct sdw_slave *slave)
+{
+	struct rt5682_priv *rt5682 = dev_get_drvdata(&slave->dev);
+
+	if (rt5682 && rt5682->hw_init)
+		cancel_delayed_work(&rt5682->jack_detect_work);
+
+	return 0;
+}
+
+static const struct sdw_device_id rt5682_id[] = {
+	SDW_SLAVE_ENTRY(0x025d, 0x5682, 0),
+	{},
+};
+MODULE_DEVICE_TABLE(sdw, rt5682_id);
+
+static int __maybe_unused rt5682_dev_suspend(struct device *dev)
+{
+	struct rt5682_priv *rt5682 = dev_get_drvdata(dev);
+
+	if (!rt5682->hw_init)
+		return 0;
+
+	regcache_cache_only(rt5682->regmap, true);
+	regcache_mark_dirty(rt5682->regmap);
+
+	return 0;
+}
+
+static int __maybe_unused rt5682_dev_resume(struct device *dev)
+{
+	struct sdw_slave *slave = dev_to_sdw_dev(dev);
+	struct rt5682_priv *rt5682 = dev_get_drvdata(dev);
+	unsigned long time;
+
+	if (!rt5682->hw_init)
+		return 0;
+
+	if (!slave->unattach_request)
+		goto regmap_sync;
+
+	time = wait_for_completion_timeout(&slave->initialization_complete,
+				msecs_to_jiffies(RT5682_PROBE_TIMEOUT));
+	if (!time) {
+		dev_err(&slave->dev, "Initialization not complete, timed out\n");
+		return -ETIMEDOUT;
+	}
+
+regmap_sync:
+	slave->unattach_request = 0;
+	regcache_cache_only(rt5682->regmap, false);
+	regcache_sync(rt5682->regmap);
+
+	return 0;
+}
+
+static const struct dev_pm_ops rt5682_pm = {
+	SET_SYSTEM_SLEEP_PM_OPS(rt5682_dev_suspend, rt5682_dev_resume)
+	SET_RUNTIME_PM_OPS(rt5682_dev_suspend, rt5682_dev_resume, NULL)
+};
+
+static struct sdw_driver rt5682_sdw_driver = {
+	.driver = {
+		.name = "rt5682",
+		.owner = THIS_MODULE,
+		.pm = &rt5682_pm,
+	},
+	.probe = rt5682_sdw_probe,
+	.remove = rt5682_sdw_remove,
+	.ops = &rt5682_slave_ops,
+	.id_table = rt5682_id,
+};
+module_sdw_driver(rt5682_sdw_driver);
+
+MODULE_DESCRIPTION("ASoC RT5682 driver SDW");
+MODULE_AUTHOR("Oder Chiou <[e-mail]>");
+MODULE_LICENSE("GPL v2");

sound/soc/codecs/rt5682-sdw.h

···
+/* SPDX-License-Identifier: GPL-2.0-only
+ *
+ * rt5682-sdw.h  --  RT5682 SDW ALSA SoC audio driver
+ *
+ * Copyright 2019 Realtek Semiconductor Corp.
+ * Author: Oder Chiou <[e-mail]>
+ */
+
+#ifndef __RT5682_SDW_H__
+#define __RT5682_SDW_H__
+
+#define RT5682_SDW_ADDR_L			0x3000
+#define RT5682_SDW_ADDR_H			0x3001
+#define RT5682_SDW_DATA_L			0x3004
+#define RT5682_SDW_DATA_H			0x3005
+#define RT5682_SDW_CMD				0x3008
+
+#define RT5682_PROBE_TIMEOUT			2000
+
+#endif /* __RT5682_SDW_H__ */

sound/soc/codecs/rt5682.c

···
 #include <linux/init.h>
 #include <linux/delay.h>
 #include <linux/pm.h>
+#include <linux/pm_runtime.h>
 #include <linux/i2c.h>
 #include <linux/platform_device.h>
 #include <linux/spi/spi.h>
 #include <linux/acpi.h>
 #include <linux/gpio.h>
 #include <linux/of_gpio.h>
-#include <linux/regulator/consumer.h>
 #include <linux/mutex.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
···
 
 #include "rl6231.h"
 #include "rt5682.h"
-
-#define RT5682_NUM_SUPPLIES 3
+#include "rt5682-sdw.h"
 
 static const char *rt5682_supply_names[RT5682_NUM_SUPPLIES] = {
 	"AVDD",
···
 	.dmic1_clk_pin = RT5682_DMIC1_CLK_GPIO3,
 	.jd_src = RT5682_JD1,
 	.btndet_delay = 16,
-};
-
-struct rt5682_priv {
-	struct snd_soc_component *component;
-	struct rt5682_platform_data pdata;
-	struct regmap *regmap;
-	struct snd_soc_jack *hs_jack;
-	struct regulator_bulk_data supplies[RT5682_NUM_SUPPLIES];
-	struct delayed_work jack_detect_work;
-	struct delayed_work jd_check_work;
-	struct mutex calibrate_mutex;
-
-	int sysclk;
-	int sysclk_src;
-	int lrck[RT5682_AIFS];
-	int bclk[RT5682_AIFS];
-	int master[RT5682_AIFS];
-
-	int pll_src;
-	int pll_in;
-	int pll_out;
-
-	int jack_type;
+	.dai_clk_names[RT5682_DAI_WCLK_IDX] = "rt5682-dai-wclk",
+	.dai_clk_names[RT5682_DAI_BCLK_IDX] = "rt5682-dai-bclk",
 };
 
 static const struct reg_sequence patch_list[] = {
 	{RT5682_HP_IMP_SENS_CTRL_19, 0x1000},
 	{RT5682_DAC_ADC_DIG_VOL1, 0xa020},
 	{RT5682_I2C_CTRL, 0x000f},
+	{RT5682_PLL2_INTERNAL, 0x8266},
 };
 
 static const struct reg_default rt5682_reg[] = {
···
 	{0x0148, 0x0000},
 	{0x0149, 0x0000},
 	{0x0150, 0x79a1},
-	{0x0151, 0x0000},
+	{0x0156, 0xaaaa},
 	{0x0160, 0x4ec0},
 	{0x0161, 0x0080},
 	{0x0162, 0x0200},
···
 static const struct snd_kcontrol_new rt5682_if1_67_adc_swap_mux =
 	SOC_DAPM_ENUM("IF1 67 ADC Swap Mux", rt5682_if1_67_adc_enum);
 
-static void rt5682_reset(struct regmap *regmap)
+static const char * const rt5682_dac_select[] = {
+	"IF1", "SOUND"
+};
+
+static SOC_ENUM_SINGLE_DECL(rt5682_dacl_enum,
+	RT5682_AD_DA_MIXER, RT5682_DAC1_L_SEL_SFT, rt5682_dac_select);
+
+static const struct snd_kcontrol_new rt5682_dac_l_mux =
+	SOC_DAPM_ENUM("DAC L Mux", rt5682_dacl_enum);
+
+static SOC_ENUM_SINGLE_DECL(rt5682_dacr_enum,
+	RT5682_AD_DA_MIXER, RT5682_DAC1_R_SEL_SFT, rt5682_dac_select);
+
+static const struct snd_kcontrol_new rt5682_dac_r_mux =
+	SOC_DAPM_ENUM("DAC R Mux", rt5682_dacr_enum);
+
+static void rt5682_reset(struct rt5682_priv *rt5682)
 {
-	regmap_write(regmap, RT5682_RESET, 0);
-	regmap_write(regmap, RT5682_I2C_MODE, 1);
+	regmap_write(rt5682->regmap, RT5682_RESET, 0);
+	if (!rt5682->is_sdw)
+		regmap_write(rt5682->regmap, RT5682_I2C_MODE, 1);
 }
 /**
  * rt5682_sel_asrc_clk_src - select ASRC clock source for a set of filters
···
 static void rt5682_enable_push_button_irq(struct snd_soc_component *component,
 		bool enable)
 {
+	struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+
 	if (enable) {
 		snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_1,
 			RT5682_SAR_BUTT_DET_MASK, RT5682_SAR_BUTT_DET_EN);
···
 		snd_soc_component_update_bits(component, RT5682_4BTN_IL_CMD_2,
 			RT5682_4BTN_IL_MASK | RT5682_4BTN_IL_RST_MASK,
 			RT5682_4BTN_IL_EN | RT5682_4BTN_IL_NOR);
-		snd_soc_component_update_bits(component, RT5682_IRQ_CTRL_3,
-			RT5682_IL_IRQ_MASK, RT5682_IL_IRQ_EN);
+		if (rt5682->is_sdw)
+			snd_soc_component_update_bits(component,
+				RT5682_IRQ_CTRL_3,
+				RT5682_IL_IRQ_MASK | RT5682_IL_IRQ_TYPE_MASK,
+				RT5682_IL_IRQ_EN | RT5682_IL_IRQ_PUL);
+		else
+			snd_soc_component_update_bits(component,
+				RT5682_IRQ_CTRL_3, RT5682_IL_IRQ_MASK,
+				RT5682_IL_IRQ_EN);
 	} else {
 		snd_soc_component_update_bits(component, RT5682_IRQ_CTRL_3,
 			RT5682_IL_IRQ_MASK, RT5682_IL_IRQ_DIS);
···
 		int jack_insert)
 {
 	struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+	struct snd_soc_dapm_context *dapm = &component->dapm;
 	unsigned int val, count;
 
 	if (jack_insert) {
···
 			RT5682_PWR_VREF2 | RT5682_PWR_MB,
 			RT5682_PWR_VREF2 | RT5682_PWR_MB);
 		snd_soc_component_update_bits(component,
-				RT5682_PWR_ANLG_1, RT5682_PWR_FV2, 0);
+			RT5682_PWR_ANLG_1, RT5682_PWR_FV2, 0);
 		usleep_range(15000, 20000);
 		snd_soc_component_update_bits(component,
-				RT5682_PWR_ANLG_1, RT5682_PWR_FV2, RT5682_PWR_FV2);
+			RT5682_PWR_ANLG_1, RT5682_PWR_FV2, RT5682_PWR_FV2);
 		snd_soc_component_update_bits(component, RT5682_PWR_ANLG_3,
 			RT5682_PWR_CBJ, RT5682_PWR_CBJ);
 
···
 		rt5682_enable_push_button_irq(component, false);
 		snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1,
 			RT5682_TRIG_JD_MASK, RT5682_TRIG_JD_LOW);
-		snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1,
-			RT5682_PWR_VREF2 | RT5682_PWR_MB, 0);
+		if (snd_soc_dapm_get_pin_status(dapm, "MICBIAS"))
+			snd_soc_component_update_bits(component,
+				RT5682_PWR_ANLG_1, RT5682_PWR_VREF2, 0);
+		else
+			snd_soc_component_update_bits(component,
+				RT5682_PWR_ANLG_1,
+				RT5682_PWR_VREF2 | RT5682_PWR_MB, 0);
 		snd_soc_component_update_bits(component, RT5682_PWR_ANLG_3,
 			RT5682_PWR_CBJ, 0);
 
···
 
 	rt5682->hs_jack = hs_jack;
 
-	if (!hs_jack) {
-		regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2,
-				   RT5682_JD1_EN_MASK, RT5682_JD1_DIS);
-		regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL,
-				   RT5682_POW_JDH | RT5682_POW_JDL, 0);
-		cancel_delayed_work_sync(&rt5682->jack_detect_work);
-		return 0;
-	}
+	if (!rt5682->is_sdw) {
+		if (!hs_jack) {
+			regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2,
+					   RT5682_JD1_EN_MASK, RT5682_JD1_DIS);
+			regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL,
+					   RT5682_POW_JDH | RT5682_POW_JDL, 0);
+			cancel_delayed_work_sync(&rt5682->jack_detect_work);
+			return 0;
+		}
 
-	switch (rt5682->pdata.jd_src) {
-	case RT5682_JD1:
-		snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_2,
-			RT5682_EXT_JD_SRC, RT5682_EXT_JD_SRC_MANUAL);
-		snd_soc_component_write(component, RT5682_CBJ_CTRL_1, 0xd042);
-		snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_3,
-			RT5682_CBJ_IN_BUF_EN, RT5682_CBJ_IN_BUF_EN);
-		snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_1,
-			RT5682_SAR_POW_MASK, RT5682_SAR_POW_EN);
-		regmap_update_bits(rt5682->regmap, RT5682_GPIO_CTRL_1,
-			RT5682_GP1_PIN_MASK, RT5682_GP1_PIN_IRQ);
-		regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL,
+		switch (rt5682->pdata.jd_src) {
+		case RT5682_JD1:
+			snd_soc_component_update_bits(component,
+				RT5682_CBJ_CTRL_2, RT5682_EXT_JD_SRC,
+				RT5682_EXT_JD_SRC_MANUAL);
+			snd_soc_component_write(component, RT5682_CBJ_CTRL_1,
+				0xd042);
+			snd_soc_component_update_bits(component,
+				RT5682_CBJ_CTRL_3, RT5682_CBJ_IN_BUF_EN,
+				RT5682_CBJ_IN_BUF_EN);
+			snd_soc_component_update_bits(component,
+				RT5682_SAR_IL_CMD_1, RT5682_SAR_POW_MASK,
+				RT5682_SAR_POW_EN);
+			regmap_update_bits(rt5682->regmap, RT5682_GPIO_CTRL_1,
+				RT5682_GP1_PIN_MASK, RT5682_GP1_PIN_IRQ);
+			regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL,
 				RT5682_POW_IRQ | RT5682_POW_JDH |
 				RT5682_POW_ANA, RT5682_POW_IRQ |
 				RT5682_POW_JDH | RT5682_POW_ANA);
-		regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_2,
-			RT5682_PWR_JDH | RT5682_PWR_JDL,
-			RT5682_PWR_JDH | RT5682_PWR_JDL);
-		regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2,
-			RT5682_JD1_EN_MASK | RT5682_JD1_POL_MASK,
-			RT5682_JD1_EN | RT5682_JD1_POL_NOR);
-		regmap_update_bits(rt5682->regmap, RT5682_4BTN_IL_CMD_4,
-			0x7f7f, (rt5682->pdata.btndet_delay << 8 |
-			rt5682->pdata.btndet_delay));
-		regmap_update_bits(rt5682->regmap, RT5682_4BTN_IL_CMD_5,
-			0x7f7f, (rt5682->pdata.btndet_delay << 8 |
-			rt5682->pdata.btndet_delay));
-		regmap_update_bits(rt5682->regmap, RT5682_4BTN_IL_CMD_6,
-			0x7f7f, (rt5682->pdata.btndet_delay << 8 |
-			rt5682->pdata.btndet_delay));
-		regmap_update_bits(rt5682->regmap, RT5682_4BTN_IL_CMD_7,
-			0x7f7f, (rt5682->pdata.btndet_delay << 8 |
-			rt5682->pdata.btndet_delay));
-		mod_delayed_work(system_power_efficient_wq,
-			   &rt5682->jack_detect_work, msecs_to_jiffies(250));
-		break;
+			regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_2,
+				RT5682_PWR_JDH | RT5682_PWR_JDL,
+				RT5682_PWR_JDH | RT5682_PWR_JDL);
+			regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2,
+				RT5682_JD1_EN_MASK | RT5682_JD1_POL_MASK,
+				RT5682_JD1_EN | RT5682_JD1_POL_NOR);
+			regmap_update_bits(rt5682->regmap, RT5682_4BTN_IL_CMD_4,
+				0x7f7f, (rt5682->pdata.btndet_delay << 8 |
+				rt5682->pdata.btndet_delay));
+			regmap_update_bits(rt5682->regmap, RT5682_4BTN_IL_CMD_5,
+				0x7f7f, (rt5682->pdata.btndet_delay << 8 |
+				rt5682->pdata.btndet_delay));
+			regmap_update_bits(rt5682->regmap, RT5682_4BTN_IL_CMD_6,
+				0x7f7f, (rt5682->pdata.btndet_delay << 8 |
+				rt5682->pdata.btndet_delay));
+			regmap_update_bits(rt5682->regmap, RT5682_4BTN_IL_CMD_7,
+				0x7f7f, (rt5682->pdata.btndet_delay << 8 |
+				rt5682->pdata.btndet_delay));
+			mod_delayed_work(system_power_efficient_wq,
+				   &rt5682->jack_detect_work,
+					msecs_to_jiffies(250));
+			break;
 
-	case RT5682_JD_NULL:
-		regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2,
-			RT5682_JD1_EN_MASK, RT5682_JD1_DIS);
-		regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL,
-				RT5682_POW_JDH | RT5682_POW_JDL, 0);
-		break;
+		case RT5682_JD_NULL:
+			regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2,
+				RT5682_JD1_EN_MASK, RT5682_JD1_DIS);
+			regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL,
+					RT5682_POW_JDH | RT5682_POW_JDL, 0);
+			break;
 
-	default:
-		dev_warn(component->dev, "Wrong JD source\n");
-		break;
+		default:
+			dev_warn(component->dev, "Wrong JD source\n");
+			break;
+		}
 	}
 
 	return 0;
···
 			    SND_JACK_BTN_0 | SND_JACK_BTN_1 |
 			    SND_JACK_BTN_2 | SND_JACK_BTN_3);
 
-	if (rt5682->jack_type & (SND_JACK_BTN_0 | SND_JACK_BTN_1 |
-		SND_JACK_BTN_2 | SND_JACK_BTN_3))
-		schedule_delayed_work(&rt5682->jd_check_work, 0);
-	else
-		cancel_delayed_work_sync(&rt5682->jd_check_work);
+	if (!rt5682->is_sdw) {
+		if (rt5682->jack_type & (SND_JACK_BTN_0 | SND_JACK_BTN_1 |
+			SND_JACK_BTN_2 | SND_JACK_BTN_3))
+			schedule_delayed_work(&rt5682->jd_check_work, 0);
+		else
+			cancel_delayed_work_sync(&rt5682->jd_check_work);
+	}
 
 	mutex_unlock(&rt5682->calibrate_mutex);
 }
···
 static const struct snd_kcontrol_new rt5682_snd_controls[] = {
 	/* DAC Digital Volume */
 	SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5682_DAC1_DIG_VOL,
-		RT5682_L_VOL_SFT + 1, RT5682_R_VOL_SFT + 1, 86, 0, dac_vol_tlv),
+		RT5682_L_VOL_SFT + 1, RT5682_R_VOL_SFT + 1, 87, 0, dac_vol_tlv),
 
 	/* IN Boost Volume */
 	SOC_SINGLE_TLV("CBJ Boost Volume", RT5682_CBJ_BST_CTRL,
···
 	}
 
 	for (i = 0; i < size - 1; i++) {
-		pr_info("div[%d]=%d\n", i, div[i]);
+		dev_dbg(rt5682->component->dev, "div[%d]=%d\n", i, div[i]);
 		if (target * div[i] == rt5682->sysclk)
 			return i;
 		if (target * div[i + 1] > rt5682->sysclk) {
-			pr_err("can't find div for sysclk %d\n",
+			dev_dbg(rt5682->component->dev, "can't find div for sysclk %d\n",
 				rt5682->sysclk);
 			return i;
 		}
···
 	struct snd_soc_component *component =
 		snd_soc_dapm_to_component(w->dapm);
 	struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
-	int idx = -EINVAL;
+	int idx = -EINVAL, dmic_clk_rate = 3072000;
 	static const int div[] = {2, 4, 6, 8, 12, 16, 24, 32, 48, 64, 96, 128};
 
-	idx = rt5682_div_sel(rt5682, 1500000, div, ARRAY_SIZE(div));
+	if (rt5682->pdata.dmic_clk_rate)
+		dmic_clk_rate = rt5682->pdata.dmic_clk_rate;
+
+	idx = rt5682_div_sel(rt5682, dmic_clk_rate, div, ARRAY_SIZE(div));
 
 	snd_soc_component_update_bits(component, RT5682_DMIC_CTRL_1,
 		RT5682_DMIC_CLK_MASK, idx << RT5682_DMIC_CLK_SFT);
···
 	int ref, val, reg, idx = -EINVAL;
 	static const int div_f[] = {1, 2, 3, 4, 6, 8, 12, 16, 24, 32, 48};
 	static const int div_o[] = {1, 2, 4, 6, 8, 12, 16, 24, 32, 48};
+
+	if (rt5682->is_sdw)
+		return 0;
 
 	val = snd_soc_component_read32(component, RT5682_GPIO_CTRL_1) &
 		RT5682_GP4_PIN_MASK;
···
 	val = snd_soc_component_read32(component, RT5682_GLB_CLK);
 	val &= RT5682_SCLK_SRC_MASK;
 	if (val == RT5682_SCLK_SRC_PLL1)
+		return 1;
+	else
+		return 0;
+}
+
+static int is_sys_clk_from_pll2(struct snd_soc_dapm_widget *w,
+			 struct snd_soc_dapm_widget *sink)
+{
+	unsigned int val;
+	struct snd_soc_component *component =
+		snd_soc_dapm_to_component(w->dapm);
+
+	val = snd_soc_component_read32(component, RT5682_GLB_CLK);
+	val &= RT5682_SCLK_SRC_MASK;
+	if (val == RT5682_SCLK_SRC_PLL2)
 		return 1;
 	else
 		return 0;
···
 static int set_dmic_power(struct snd_soc_dapm_widget *w,
 	struct snd_kcontrol *kcontrol, int event)
 {
+	struct snd_soc_component *component =
+		snd_soc_dapm_to_component(w->dapm);
+	struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+	unsigned int delay = 50;
+
+	if (rt5682->pdata.dmic_delay)
+		delay = rt5682->pdata.dmic_delay;
+
 	switch (event) {
 	case SND_SOC_DAPM_POST_PMU:
 		/*Add delay to avoid pop noise*/
-		msleep(150);
+		msleep(delay);
 		break;
 
 	default:
···
 	return 0;
 }
 
-static int rt5655_set_verf(struct snd_soc_dapm_widget *w,
+static int rt5682_set_verf(struct snd_soc_dapm_widget *w,
 	struct snd_kcontrol *kcontrol, int event)
 {
 	struct snd_soc_component *component =
···
 	SND_SOC_DAPM_SUPPLY("PLL2B", RT5682_PWR_ANLG_3, RT5682_PWR_PLL2B_BIT,
 		0, NULL, 0),
 	SND_SOC_DAPM_SUPPLY("PLL2F", RT5682_PWR_ANLG_3, RT5682_PWR_PLL2F_BIT,
-		0, NULL, 0),
+		0, set_filter_clk, SND_SOC_DAPM_PRE_PMU),
 	SND_SOC_DAPM_SUPPLY("Vref1", RT5682_PWR_ANLG_1, RT5682_PWR_VREF1_BIT, 0,
-		rt5655_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+		rt5682_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+	SND_SOC_DAPM_SUPPLY("Vref2", RT5682_PWR_ANLG_1, RT5682_PWR_VREF2_BIT, 0,
+		NULL, 0),
+	SND_SOC_DAPM_SUPPLY("MICBIAS", SND_SOC_NOPM, 0, 0, NULL, 0),
 
 	/* ASRC */
 	SND_SOC_DAPM_SUPPLY_S("DAC STO1 ASRC", 1, RT5682_PLL_TRACK_1,
···
 	SND_SOC_DAPM_PGA("IF1 DAC1", SND_SOC_NOPM, 0, 0, NULL, 0),
 	SND_SOC_DAPM_PGA("IF1 DAC1 L", SND_SOC_NOPM, 0, 0, NULL, 0),
 	SND_SOC_DAPM_PGA("IF1 DAC1 R", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("SOUND DAC L", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("SOUND DAC R", SND_SOC_NOPM, 0, 0, NULL, 0),
 
 	/* Digital Interface Select */
 	SND_SOC_DAPM_MUX("IF1 01 ADC Swap Mux", SND_SOC_NOPM, 0, 0,
···
 	SND_SOC_DAPM_MUX("ADCDAT Mux", SND_SOC_NOPM, 0, 0,
 			&rt5682_adcdat_pin_ctrl),
 
+	SND_SOC_DAPM_MUX("DAC L Mux", SND_SOC_NOPM, 0, 0,
+			&rt5682_dac_l_mux),
+	SND_SOC_DAPM_MUX("DAC R Mux", SND_SOC_NOPM, 0, 0,
+			&rt5682_dac_r_mux),
+
 	/* Audio Interface */
 	SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0,
 		RT5682_I2S1_SDP, RT5682_SEL_ADCDAT_SFT, 1),
 	SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0,
 		RT5682_I2S2_SDP, RT5682_I2S2_PIN_CFG_SFT, 1),
 	SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("SDWRX", "SDW Playback", 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("SDWTX", "SDW Capture", 0, SND_SOC_NOPM, 0, 0),
 
 	/* Output Side */
 	/* DAC mixer before sound effect  */
···
 static const struct snd_soc_dapm_route rt5682_dapm_routes[] = {
 	/*PLL*/
 	{"ADC Stereo1 Filter", NULL, "PLL1", is_sys_clk_from_pll1},
+	{"ADC Stereo1 Filter", NULL, "PLL2B", is_sys_clk_from_pll2},
+	{"ADC Stereo1 Filter", NULL, "PLL2F", is_sys_clk_from_pll2},
 	{"DAC Stereo1 Filter", NULL, "PLL1", is_sys_clk_from_pll1},
+	{"DAC Stereo1 Filter", NULL, "PLL2B", is_sys_clk_from_pll2},
+	{"DAC Stereo1 Filter", NULL, "PLL2F", is_sys_clk_from_pll2},
 
 	/*ASRC*/
 	{"ADC Stereo1 Filter", NULL, "ADC STO1 ASRC", is_using_asrc},
···
 	{"IF1_ADC Mux", "Slot 2", "IF1 23 ADC Swap Mux"},
 	{"IF1_ADC Mux", "Slot 4", "IF1 45 ADC Swap Mux"},
 	{"IF1_ADC Mux", "Slot 6", "IF1 67 ADC Swap Mux"},
-	{"IF1_ADC Mux", NULL, "I2S1"},
 	{"ADCDAT Mux", "ADCDAT1", "IF1_ADC Mux"},
+	{"AIF1TX", NULL, "I2S1"},
 	{"AIF1TX", NULL, "ADCDAT Mux"},
 	{"IF2 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"},
 	{"IF2 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"},
···
 	{"ADCDAT Mux", "ADCDAT2", "IF2 ADC Swap Mux"},
 	{"AIF2TX", NULL, "ADCDAT Mux"},
 
+	{"SDWTX", NULL, "PLL2B"},
+	{"SDWTX", NULL, "PLL2F"},
+	{"SDWTX", NULL, "ADCDAT Mux"},
+
 	{"IF1 DAC1 L", NULL, "AIF1RX"},
 	{"IF1 DAC1 L", NULL, "I2S1"},
 	{"IF1 DAC1 L", NULL, "DAC Stereo1 Filter"},
···
 	{"IF1 DAC1 R", NULL, "I2S1"},
 	{"IF1 DAC1 R", NULL, "DAC Stereo1 Filter"},
 
+	{"SOUND DAC L", NULL, "SDWRX"},
+	{"SOUND DAC L", NULL, "DAC Stereo1 Filter"},
+	{"SOUND DAC L", NULL, "PLL2B"},
+	{"SOUND DAC L", NULL, "PLL2F"},
+	{"SOUND DAC R", NULL, "SDWRX"},
+	{"SOUND DAC R", NULL, "DAC Stereo1 Filter"},
+	{"SOUND DAC R", NULL, "PLL2B"},
+	{"SOUND DAC R", NULL, "PLL2F"},
+
+	{"DAC L Mux", "IF1", "IF1 DAC1 L"},
+	{"DAC L Mux", "SOUND", "SOUND DAC L"},
+	{"DAC R Mux", "IF1", "IF1 DAC1 R"},
+	{"DAC R Mux", "SOUND", "SOUND DAC R"},
+
 	{"DAC1 MIXL", "Stereo ADC Switch", "Stereo1 ADC MIXL"},
-	{"DAC1 MIXL", "DAC1 Switch", "IF1 DAC1 L"},
+	{"DAC1 MIXL", "DAC1 Switch", "DAC L Mux"},
 	{"DAC1 MIXR", "Stereo ADC Switch", "Stereo1 ADC MIXR"},
-	{"DAC1 MIXR", "DAC1 Switch", "IF1 DAC1 R"},
+	{"DAC1 MIXR", "DAC1 Switch", "DAC R Mux"},
 
 	{"Stereo1 DAC MIXL", "DAC L1 Switch", "DAC1 MIXL"},
 	{"Stereo1 DAC MIXL", "DAC R1 Switch", "DAC1 MIXR"},
···
 			RT5682_I2S1_DL_MASK, len_1);
 		if (rt5682->master[RT5682_AIF1]) {
 			snd_soc_component_update_bits(component,
-				RT5682_ADDA_CLK_1, RT5682_I2S_M_DIV_MASK,
-				pre_div << RT5682_I2S_M_DIV_SFT);
+				RT5682_ADDA_CLK_1, RT5682_I2S_M_DIV_MASK |
+				RT5682_I2S_CLK_SRC_MASK,
+				pre_div << RT5682_I2S_M_DIV_SFT |
+				(rt5682->sysclk_src) << RT5682_I2S_CLK_SRC_SFT);
 		}
 		if (params_channels(params) == 1) /* mono mode */
 			snd_soc_component_update_bits(component,
···
 		unsigned int freq_out)
 {
 	struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
-	struct rl6231_pll_code pll_code;
+	struct rl6231_pll_code pll_code, pll2f_code, pll2b_code;
+	unsigned int pll2_fout1;
 	int ret;
 
-	if (source == rt5682->pll_src && freq_in == rt5682->pll_in &&
-	    freq_out == rt5682->pll_out)
+	if (source == rt5682->pll_src[pll_id] &&
+	    freq_in == rt5682->pll_in[pll_id] &&
+	    freq_out == rt5682->pll_out[pll_id])
 		return 0;
 
 	if (!freq_in || !freq_out) {
 		dev_dbg(component->dev, "PLL disabled\n");
 
-		rt5682->pll_in = 0;
-		rt5682->pll_out = 0;
+		rt5682->pll_in[pll_id] = 0;
+		rt5682->pll_out[pll_id] = 0;
 		snd_soc_component_update_bits(component, RT5682_GLB_CLK,
 			RT5682_SCLK_SRC_MASK, RT5682_SCLK_SRC_MCLK);
 		return 0;
 	}
 
-	switch (source) {
-	case RT5682_PLL1_S_MCLK:
-		snd_soc_component_update_bits(component, RT5682_GLB_CLK,
-			RT5682_PLL1_SRC_MASK, RT5682_PLL1_SRC_MCLK);
-		break;
-	case RT5682_PLL1_S_BCLK1:
-		snd_soc_component_update_bits(component, RT5682_GLB_CLK,
-				RT5682_PLL1_SRC_MASK, RT5682_PLL1_SRC_BCLK1);
-		break;
-	default:
-		dev_err(component->dev, "Unknown PLL Source %d\n", source);
-		return -EINVAL;
+	if (pll_id == RT5682_PLL2) {
+		switch (source) {
+		case RT5682_PLL2_S_MCLK:
+			snd_soc_component_update_bits(component,
+				RT5682_GLB_CLK, RT5682_PLL2_SRC_MASK,
+				RT5682_PLL2_SRC_MCLK);
+			break;
+		default:
+			dev_err(component->dev, "Unknown PLL2 Source %d\n",
+				source);
+			return -EINVAL;
+		}
+
+		/**
+		 * PLL2 concatenates 2 PLL units.
+		 * We suggest the Fout of the front PLL is 3.84MHz.
+		 */
+		pll2_fout1 = 3840000;
+		ret = rl6231_pll_calc(freq_in, pll2_fout1, &pll2f_code);
+		if (ret < 0) {
+			dev_err(component->dev, "Unsupport input clock %d\n",
+				freq_in);
+			return ret;
+		}
+		dev_dbg(component->dev, "PLL2F: fin=%d fout=%d bypass=%d m=%d n=%d k=%d\n",
+			freq_in, pll2_fout1,
+			pll2f_code.m_bp,
+			(pll2f_code.m_bp ? 0 : pll2f_code.m_code),
+			pll2f_code.n_code, pll2f_code.k_code);
+
+		ret = rl6231_pll_calc(pll2_fout1, freq_out, &pll2b_code);
+		if (ret < 0) {
+			dev_err(component->dev, "Unsupport input clock %d\n",
+				pll2_fout1);
+			return ret;
+		}
+		dev_dbg(component->dev, "PLL2B: fin=%d fout=%d bypass=%d m=%d n=%d k=%d\n",
+			pll2_fout1, freq_out,
+			pll2b_code.m_bp,
+			(pll2b_code.m_bp ? 0 : pll2b_code.m_code),
+			pll2b_code.n_code, pll2b_code.k_code);
+
+		snd_soc_component_write(component, RT5682_PLL2_CTRL_1,
+			pll2f_code.k_code << RT5682_PLL2F_K_SFT |
+			pll2b_code.k_code << RT5682_PLL2B_K_SFT |
+			pll2b_code.m_code);
+		snd_soc_component_write(component, RT5682_PLL2_CTRL_2,
+			pll2f_code.m_code << RT5682_PLL2F_M_SFT |
+			pll2b_code.n_code);
+		snd_soc_component_write(component, RT5682_PLL2_CTRL_3,
+			pll2f_code.n_code << RT5682_PLL2F_N_SFT);
+		snd_soc_component_update_bits(component, RT5682_PLL2_CTRL_4,
+			RT5682_PLL2B_M_BP_MASK | RT5682_PLL2F_M_BP_MASK | 0xf,
+			(pll2b_code.m_bp ? 1 : 0) << RT5682_PLL2B_M_BP_SFT |
+			(pll2f_code.m_bp ? 1 : 0) << RT5682_PLL2F_M_BP_SFT |
+			0xf);
+	} else {
+		switch (source) {
+		case RT5682_PLL1_S_MCLK:
+			snd_soc_component_update_bits(component,
+				RT5682_GLB_CLK, RT5682_PLL1_SRC_MASK,
+				RT5682_PLL1_SRC_MCLK);
+			break;
+		case RT5682_PLL1_S_BCLK1:
+			snd_soc_component_update_bits(component,
+				RT5682_GLB_CLK, RT5682_PLL1_SRC_MASK,
+				RT5682_PLL1_SRC_BCLK1);
+			break;
+		default:
+			dev_err(component->dev, "Unknown PLL1 Source %d\n",
+				source);
+			return -EINVAL;
+		}
+
+		ret = rl6231_pll_calc(freq_in, freq_out, &pll_code);
+		if (ret < 0) {
+			dev_err(component->dev, "Unsupport input clock %d\n",
+				freq_in);
+			return ret;
+		}
+
+		dev_dbg(component->dev, "bypass=%d m=%d n=%d k=%d\n",
+			pll_code.m_bp, (pll_code.m_bp ? 0 : pll_code.m_code),
+			pll_code.n_code, pll_code.k_code);
+
+		snd_soc_component_write(component, RT5682_PLL_CTRL_1,
+			pll_code.n_code << RT5682_PLL_N_SFT | pll_code.k_code);
+		snd_soc_component_write(component, RT5682_PLL_CTRL_2,
+		    (pll_code.m_bp ? 0 : pll_code.m_code) << RT5682_PLL_M_SFT |
+		    pll_code.m_bp << RT5682_PLL_M_BP_SFT | RT5682_PLL_RST);
 	}
 
-	ret = rl6231_pll_calc(freq_in, freq_out, &pll_code);
-	if (ret < 0) {
-		dev_err(component->dev, "Unsupport input clock %d\n", freq_in);
-		return ret;
-	}
-
-	dev_dbg(component->dev, "bypass=%d m=%d n=%d k=%d\n",
-		pll_code.m_bp, (pll_code.m_bp ? 0 : pll_code.m_code),
-		pll_code.n_code, pll_code.k_code);
-
-	snd_soc_component_write(component, RT5682_PLL_CTRL_1,
-		pll_code.n_code << RT5682_PLL_N_SFT | pll_code.k_code);
-	snd_soc_component_write(component, RT5682_PLL_CTRL_2,
-		(pll_code.m_bp ? 0 : pll_code.m_code) << RT5682_PLL_M_SFT |
-		pll_code.m_bp << RT5682_PLL_M_BP_SFT | RT5682_PLL_RST);
-
-	rt5682->pll_in = freq_in;
-	rt5682->pll_out = freq_out;
-	rt5682->pll_src = source;
+	rt5682->pll_in[pll_id] = freq_in;
+	rt5682->pll_out[pll_id] = freq_out;
+	rt5682->pll_src[pll_id] = source;
 
 	return 0;
 }
 
-static int rt5682_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio)
+static int rt5682_set_bclk1_ratio(struct snd_soc_dai *dai, unsigned int ratio)
+{
+	struct snd_soc_component *component = dai->component;
+	struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+
+	rt5682->bclk[dai->id] = ratio;
+
+	switch (ratio) {
+	case 256:
+		snd_soc_component_update_bits(component, RT5682_TDM_TCON_CTRL,
+			RT5682_TDM_BCLK_MS1_MASK, RT5682_TDM_BCLK_MS1_256);
+		break;
+	case 128:
+		snd_soc_component_update_bits(component, RT5682_TDM_TCON_CTRL,
+			RT5682_TDM_BCLK_MS1_MASK, RT5682_TDM_BCLK_MS1_128);
+		break;
+	case 64:
+		snd_soc_component_update_bits(component, RT5682_TDM_TCON_CTRL,
+			RT5682_TDM_BCLK_MS1_MASK, RT5682_TDM_BCLK_MS1_64);
+		break;
+	case 32:
+		snd_soc_component_update_bits(component, RT5682_TDM_TCON_CTRL,
+			RT5682_TDM_BCLK_MS1_MASK, RT5682_TDM_BCLK_MS1_32);
+		break;
+	default:
+		dev_err(dai->dev, "Invalid bclk1 ratio %d\n", ratio);
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int rt5682_set_bclk2_ratio(struct snd_soc_dai *dai, unsigned int ratio)
 {
 	struct snd_soc_component *component = dai->component;
 	struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
···
 			RT5682_I2S2_BCLK_MS2_32);
 		break;
 	default:
-		dev_err(dai->dev, "Invalid bclk ratio %d\n", ratio);
+		dev_err(dai->dev, "Invalid bclk2 ratio %d\n", ratio);
 		return -EINVAL;
 	}
 
···
 	return 0;
 }
 
+#ifdef CONFIG_COMMON_CLK
+#define CLK_PLL2_FIN 48000000
+#define CLK_PLL2_FOUT 24576000
+#define CLK_48 48000
+
+static bool rt5682_clk_check(struct rt5682_priv *rt5682)
+{
+	if (!rt5682->master[RT5682_AIF1]) {
+		dev_err(rt5682->component->dev, "sysclk/dai not set correctly\n");
+		return false;
+	}
+	return true;
+}
+
+static int rt5682_wclk_prepare(struct clk_hw *hw)
+{
+	struct rt5682_priv *rt5682 =
+		container_of(hw, struct rt5682_priv,
+			     dai_clks_hw[RT5682_DAI_WCLK_IDX]);
+	struct snd_soc_component *component = rt5682->component;
+	struct snd_soc_dapm_context *dapm =
+			snd_soc_component_get_dapm(component);
+
+	if (!rt5682_clk_check(rt5682))
+		return -EINVAL;
+
+	snd_soc_dapm_mutex_lock(dapm);
+
+	snd_soc_dapm_force_enable_pin_unlocked(dapm, "MICBIAS");
+	snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1,
+				RT5682_PWR_MB, RT5682_PWR_MB);
+	snd_soc_dapm_force_enable_pin_unlocked(dapm, "I2S1");
+	snd_soc_dapm_force_enable_pin_unlocked(dapm, "PLL2F");
+	snd_soc_dapm_force_enable_pin_unlocked(dapm, "PLL2B");
+	snd_soc_dapm_sync_unlocked(dapm);
+
+	snd_soc_dapm_mutex_unlock(dapm);
+
+	return 0;
+}
+
+static void rt5682_wclk_unprepare(struct clk_hw *hw)
+{
+	struct rt5682_priv *rt5682 =
+		container_of(hw, struct rt5682_priv,
+			     dai_clks_hw[RT5682_DAI_WCLK_IDX]);
+	struct snd_soc_component *component = rt5682->component;
+	struct snd_soc_dapm_context *dapm =
+			snd_soc_component_get_dapm(component);
+
+	if (!rt5682_clk_check(rt5682))
+		return;
+
+	snd_soc_dapm_mutex_lock(dapm);
+
+	snd_soc_dapm_disable_pin_unlocked(dapm, "MICBIAS");
+	if (!rt5682->jack_type)
+		snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1,
+				RT5682_PWR_MB, 0);
+	snd_soc_dapm_disable_pin_unlocked(dapm, "I2S1");
+	snd_soc_dapm_disable_pin_unlocked(dapm, "PLL2F");
+	snd_soc_dapm_disable_pin_unlocked(dapm, "PLL2B");
+	snd_soc_dapm_sync_unlocked(dapm);
+
+	snd_soc_dapm_mutex_unlock(dapm);
+}
+
+static unsigned long rt5682_wclk_recalc_rate(struct clk_hw *hw,
+					     unsigned long parent_rate)
+{
+	struct rt5682_priv *rt5682 =
+		container_of(hw, struct rt5682_priv,
+			     dai_clks_hw[RT5682_DAI_WCLK_IDX]);
+
+	if (!rt5682_clk_check(rt5682))
+		return 0;
+	/*
+	 * Only accept to set wclk rate to 48kHz temporarily.
+	 */
+	return CLK_48;
+}
+
+static long rt5682_wclk_round_rate(struct clk_hw *hw, unsigned long rate,
+				   unsigned long *parent_rate)
+{
+	struct rt5682_priv *rt5682 =
+		container_of(hw, struct rt5682_priv,
+			     dai_clks_hw[RT5682_DAI_WCLK_IDX]);
+
+	if (!rt5682_clk_check(rt5682))
+		return -EINVAL;
+	/*
+	 * Only accept to set wclk rate to 48kHz temporarily.
+	 */
+	return CLK_48;
+}
+
+static int rt5682_wclk_set_rate(struct clk_hw *hw, unsigned long rate,
+				unsigned long parent_rate)
+{
+	struct rt5682_priv *rt5682 =
+		container_of(hw, struct rt5682_priv,
+			     dai_clks_hw[RT5682_DAI_WCLK_IDX]);
+	struct snd_soc_component *component = rt5682->component;
+	struct clk *parent_clk;
+	const char * const clk_name = __clk_get_name(hw->clk);
+	int pre_div;
+
+	if (!rt5682_clk_check(rt5682))
+		return -EINVAL;
+
+	/*
+	 * Whether the wclk's parent clk (mclk) exists or not, please ensure
+	 * it is fixed or set to 48MHz before setting wclk rate. It's a
+	 * temporary limitation. Only accept 48MHz clk as the clk provider.
+	 *
+	 * It will set the codec anyway by assuming mclk is 48MHz.
+	 */
+	parent_clk = clk_get_parent(hw->clk);
+	if (!parent_clk)
+		dev_warn(component->dev,
+			"Parent mclk of wclk not acquired in driver. Please ensure mclk was provided as %d Hz.\n",
+			CLK_PLL2_FIN);
+
+	if (parent_rate != CLK_PLL2_FIN)
+		dev_warn(component->dev, "clk %s only support %d Hz input\n",
+			clk_name, CLK_PLL2_FIN);
+
+	/*
+	 * It's a temporary limitation. Only accept to set wclk rate to 48kHz.
+	 * It will force wclk to 48kHz even it's not.
+	 */
+	if (rate != CLK_48) {
+		dev_warn(component->dev, "clk %s only support %d Hz output\n",
+			clk_name, CLK_48);
+		rate = CLK_48;
+	}
+
+	/*
+	 * To achieve the rate conversion from 48MHz to 48kHz, PLL2 is needed.
+	 */
+	rt5682_set_component_pll(component, RT5682_PLL2, RT5682_PLL2_S_MCLK,
+		CLK_PLL2_FIN, CLK_PLL2_FOUT);
+
+	rt5682_set_component_sysclk(component, RT5682_SCLK_S_PLL2, 0,
+		CLK_PLL2_FOUT, SND_SOC_CLOCK_IN);
+
+	pre_div = rl6231_get_clk_info(rt5682->sysclk, rate);
+
+	snd_soc_component_update_bits(component, RT5682_ADDA_CLK_1,
+		RT5682_I2S_M_DIV_MASK | RT5682_I2S_CLK_SRC_MASK,
+		pre_div << RT5682_I2S_M_DIV_SFT |
+		(rt5682->sysclk_src) << RT5682_I2S_CLK_SRC_SFT);
+
+	return 0;
+}
+
+static unsigned long rt5682_bclk_recalc_rate(struct clk_hw *hw,
+					     unsigned long parent_rate)
+{
+	struct rt5682_priv *rt5682 =
+		container_of(hw, struct rt5682_priv,
+			     dai_clks_hw[RT5682_DAI_BCLK_IDX]);
+	struct snd_soc_component *component = rt5682->component;
+	unsigned int bclks_per_wclk;
+
+	snd_soc_component_read(component, RT5682_TDM_TCON_CTRL,
+				&bclks_per_wclk);
+
+	switch (bclks_per_wclk & RT5682_TDM_BCLK_MS1_MASK) {
+	case RT5682_TDM_BCLK_MS1_256:
+		return parent_rate * 256;
+	case RT5682_TDM_BCLK_MS1_128:
+		return parent_rate * 128;
+	case RT5682_TDM_BCLK_MS1_64:
+		return parent_rate * 64;
+	case RT5682_TDM_BCLK_MS1_32:
+		return parent_rate * 32;
+	default:
+		return 0;
+	}
+}
+
+static unsigned long rt5682_bclk_get_factor(unsigned long rate,
+					    unsigned long parent_rate)
+{
+	unsigned long factor;
+
+	factor = rate / parent_rate;
+	if (factor < 64)
+		return 32;
+	else if (factor < 128)
+		return 64;
+	else if (factor < 256)
+		return 128;
+	else
+		return 256;
+}
+
+static long rt5682_bclk_round_rate(struct clk_hw *hw, unsigned long rate,
+				   unsigned long *parent_rate)
+{
+	struct rt5682_priv *rt5682 =
+		container_of(hw, struct rt5682_priv,
+			     dai_clks_hw[RT5682_DAI_BCLK_IDX]);
+	unsigned long factor;
+
+	if (!*parent_rate || !rt5682_clk_check(rt5682))
+		return -EINVAL;
+
+	/*
+	 * BCLK rates are set as a multiplier of WCLK in HW.
+	 * We don't allow changing the parent WCLK. We just do
+	 * some rounding down based on the parent WCLK rate
+	 * and find the appropriate multiplier of BCLK to
+	 * get the rounded down BCLK value.
+	 */
+	factor = rt5682_bclk_get_factor(rate, *parent_rate);
+
+	return *parent_rate * factor;
+}
+
+static int rt5682_bclk_set_rate(struct clk_hw *hw, unsigned long rate,
+				unsigned long parent_rate)
+{
+	struct rt5682_priv *rt5682 =
+		container_of(hw, struct rt5682_priv,
+			     dai_clks_hw[RT5682_DAI_BCLK_IDX]);
+	struct snd_soc_component *component = rt5682->component;
+	struct snd_soc_dai *dai = NULL;
+	unsigned long factor;
+
+	if (!rt5682_clk_check(rt5682))
+		return -EINVAL;
+
+	factor = rt5682_bclk_get_factor(rate, parent_rate);
+
+	for_each_component_dais(component, dai)
+		if (dai->id == RT5682_AIF1)
+			break;
+	if (!dai) {
+		dev_err(component->dev, "dai %d not found in component\n",
+			RT5682_AIF1);
+		return -ENODEV;
+	}
+
+	return rt5682_set_bclk1_ratio(dai, factor);
+}
+
+static const struct clk_ops rt5682_dai_clk_ops[RT5682_DAI_NUM_CLKS] = {
+	[RT5682_DAI_WCLK_IDX] = {
+		.prepare = rt5682_wclk_prepare,
+		.unprepare = rt5682_wclk_unprepare,
+		.recalc_rate = rt5682_wclk_recalc_rate,
+		.round_rate = rt5682_wclk_round_rate,
+		.set_rate = rt5682_wclk_set_rate,
+	},
+	[RT5682_DAI_BCLK_IDX] = {
+		.recalc_rate = rt5682_bclk_recalc_rate,
+		.round_rate = rt5682_bclk_round_rate,
+		.set_rate = rt5682_bclk_set_rate,
+	},
+};
+
+static int rt5682_register_dai_clks(struct snd_soc_component *component)
+{
+	struct device *dev = component->dev;
+	struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+	struct rt5682_platform_data *pdata = &rt5682->pdata;
+	struct clk_init_data init;
+	struct clk *dai_clk;
+	struct clk_lookup *dai_clk_lookup;
+	struct clk_hw *dai_clk_hw;
+	const char *parent_name;
+	int i, ret;
+
+	for (i = 0; i < RT5682_DAI_NUM_CLKS; ++i) {
+		dai_clk_hw = &rt5682->dai_clks_hw[i];
+
+		switch (i) {
+		case RT5682_DAI_WCLK_IDX:
+			/* Make MCLK the parent of WCLK */
+			if (rt5682->mclk) {
+				parent_name = __clk_get_name(rt5682->mclk);
+				init.parent_names = &parent_name;
+				init.num_parents = 1;
+			} else {
+				init.parent_names = NULL;
+				init.num_parents = 0;
+			}
+			break;
+		case RT5682_DAI_BCLK_IDX:
+			/* Make WCLK the parent of BCLK */
+			parent_name = __clk_get_name(
+				rt5682->dai_clks[RT5682_DAI_WCLK_IDX]);
+			init.parent_names = &parent_name;
+			init.num_parents = 1;
+			break;
+		default:
+			dev_err(dev, "Invalid clock index\n");
+			ret = -EINVAL;
+			goto err;
+		}
+
+		init.name = pdata->dai_clk_names[i];
+		init.ops = &rt5682_dai_clk_ops[i];
+		init.flags = CLK_GET_RATE_NOCACHE | CLK_SET_RATE_GATE;
+		dai_clk_hw->init = &init;
+
+		dai_clk = devm_clk_register(dev, dai_clk_hw);
+		if (IS_ERR(dai_clk)) {
+			dev_warn(dev, "Failed to register %s: %ld\n",
+				 init.name, PTR_ERR(dai_clk));
+			ret = PTR_ERR(dai_clk);
+			goto err;
+		}
+		rt5682->dai_clks[i] = dai_clk;
+
+		if (dev->of_node) {
+			devm_of_clk_add_hw_provider(dev, of_clk_hw_simple_get,
+						    dai_clk_hw);
+		} else {
+			dai_clk_lookup = clkdev_create(dai_clk, init.name,
+						       "%s", dev_name(dev));
+			if (!dai_clk_lookup) {
+				ret = -ENOMEM;
+				goto err;
+			} else {
+				rt5682->dai_clks_lookup[i] = dai_clk_lookup;
+			}
+		}
+	}
+
+	return 0;
+
+err:
+	do {
+		if (rt5682->dai_clks_lookup[i])
+			clkdev_drop(rt5682->dai_clks_lookup[i]);
+	} while (i-- > 0);
+
+	return ret;
+}
+#endif /* CONFIG_COMMON_CLK */
+
 static int rt5682_probe(struct snd_soc_component *component)
 {
 	struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+	struct sdw_slave *slave;
+	unsigned long time;
 
+#ifdef CONFIG_COMMON_CLK
+	int ret;
+#endif
 	rt5682->component = component;
+
+	if (rt5682->is_sdw) {
+		slave = rt5682->slave;
+		time = wait_for_completion_timeout(
+			&slave->initialization_complete,
+			msecs_to_jiffies(RT5682_PROBE_TIMEOUT));
+		if (!time) {
+			dev_err(&slave->dev, "Initialization not complete, timed out\n");
+			return -ETIMEDOUT;
+		}
+	} else {
+#ifdef CONFIG_COMMON_CLK
+		/* Check if MCLK provided */
+		rt5682->mclk = devm_clk_get(component->dev, "mclk");
+		if (IS_ERR(rt5682->mclk)) {
+			if (PTR_ERR(rt5682->mclk) != -ENOENT) {
+				ret = PTR_ERR(rt5682->mclk);
+				return ret;
+			}
+			rt5682->mclk = NULL;
+		} else {
+			/* Register CCF DAI clock control */
+			ret = rt5682_register_dai_clks(component);
+			if (ret)
+				return ret;
+		}
+		/* Initial setup for CCF */
+		rt5682->lrck[RT5682_AIF1] = CLK_48;
+#endif
+	}
 
 	return 0;
 }
···
 {
 	struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
 
-	rt5682_reset(rt5682->regmap);
+#ifdef CONFIG_COMMON_CLK
+	int i;
+
+	for (i = RT5682_DAI_NUM_CLKS - 1; i >= 0; --i) {
+		if (rt5682->dai_clks_lookup[i])
+			clkdev_drop(rt5682->dai_clks_lookup[i]);
+	}
+#endif
+
+	rt5682_reset(rt5682);
 }
 
 #ifdef CONFIG_PM
···
 	.hw_params = rt5682_hw_params,
 	.set_fmt = rt5682_set_dai_fmt,
 	.set_tdm_slot = rt5682_set_tdm_slot,
+	.set_bclk_ratio = rt5682_set_bclk1_ratio,
 };
 
 static const struct snd_soc_dai_ops rt5682_aif2_dai_ops = {
 	.hw_params = rt5682_hw_params,
 	.set_fmt = rt5682_set_dai_fmt,
-	.set_bclk_ratio = rt5682_set_bclk_ratio,
+	.set_bclk_ratio = rt5682_set_bclk2_ratio,
 };
+
+#if IS_ENABLED(CONFIG_SND_SOC_RT5682_SDW)
+struct sdw_stream_data {
+	struct sdw_stream_runtime *sdw_stream;
+};
+
+static int rt5682_set_sdw_stream(struct snd_soc_dai *dai, void *sdw_stream,
+				int direction)
+{
+	struct sdw_stream_data *stream;
+
+	stream = kzalloc(sizeof(*stream), GFP_KERNEL);
+	if (!stream)
+		return -ENOMEM;
+
+	stream->sdw_stream = (struct sdw_stream_runtime *)sdw_stream;
+
+	/* Use tx_mask or rx_mask to configure stream tag and set dma_data */
+	if (direction == SNDRV_PCM_STREAM_PLAYBACK)
+		dai->playback_dma_data = stream;
+	else
+		dai->capture_dma_data = stream;
+
+	return 0;
+}
+
+static void rt5682_sdw_shutdown(struct snd_pcm_substream *substream,
+				struct snd_soc_dai *dai)
+{
+	struct sdw_stream_data *stream;
+
+	stream = snd_soc_dai_get_dma_data(dai, substream);
+	snd_soc_dai_set_dma_data(dai, substream, NULL);
+	kfree(stream);
+}
+
+static int rt5682_sdw_hw_params(struct snd_pcm_substream *substream,
+				struct snd_pcm_hw_params *params,
+				struct snd_soc_dai *dai)
+{
+	struct snd_soc_component *component = dai->component;
+	struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+	struct sdw_stream_config stream_config;
+	struct sdw_port_config port_config;
+	enum sdw_data_direction direction;
+	struct sdw_stream_data *stream;
+	int retval, port, num_channels;
+	unsigned int val_p = 0, val_c = 0, osr_p = 0, osr_c = 0;
+
+	dev_dbg(dai->dev, "%s %s", __func__, dai->name);
+	stream = snd_soc_dai_get_dma_data(dai, substream);
+
+	if (!stream)
+		return -ENOMEM;
+
+	if (!rt5682->slave)
+		return -EINVAL;
+
+	/* SoundWire specific configuration */
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		direction = SDW_DATA_DIR_RX;
+		port = 1;
+	} else {
+		direction = SDW_DATA_DIR_TX;
+		port = 2;
+	}
+
+	stream_config.frame_rate = params_rate(params);
+	stream_config.ch_count = params_channels(params);
+	stream_config.bps = snd_pcm_format_width(params_format(params));
+	stream_config.direction = direction;
+
+	num_channels = params_channels(params);
+	port_config.ch_mask = (1 << (num_channels)) - 1;
+	port_config.num = port;
+
+	retval = sdw_stream_add_slave(rt5682->slave, &stream_config,
+				      &port_config, 1, stream->sdw_stream);
+	if (retval) {
+		dev_err(dai->dev, "Unable to configure port\n");
+		return retval;
+	}
+
+	switch (params_rate(params)) {
+	case 48000:
+		val_p = RT5682_SDW_REF_1_48K;
+		val_c = RT5682_SDW_REF_2_48K;
+		break;
+	case 96000:
+		val_p = RT5682_SDW_REF_1_96K;
+		val_c = RT5682_SDW_REF_2_96K;
+		break;
+	case 192000:
+		val_p = RT5682_SDW_REF_1_192K;
+		val_c = RT5682_SDW_REF_2_192K;
+		break;
+	case 32000:
+		val_p = RT5682_SDW_REF_1_32K;
+		val_c = RT5682_SDW_REF_2_32K;
+		break;
+	case 24000:
+		val_p = RT5682_SDW_REF_1_24K;
+		val_c = RT5682_SDW_REF_2_24K;
+		break;
+	case 16000:
+		val_p = RT5682_SDW_REF_1_16K;
+		val_c = RT5682_SDW_REF_2_16K;
+		break;
+	case 12000:
+		val_p = RT5682_SDW_REF_1_12K;
+		val_c = RT5682_SDW_REF_2_12K;
+		break;
+	case 8000:
+		val_p = RT5682_SDW_REF_1_8K;
+		val_c = RT5682_SDW_REF_2_8K;
+		break;
+	case 44100:
+		val_p = RT5682_SDW_REF_1_44K;
+		val_c = RT5682_SDW_REF_2_44K;
+		break;
+	case 88200:
+		val_p = RT5682_SDW_REF_1_88K;
+		val_c = RT5682_SDW_REF_2_88K;
+		break;
+	case 176400:
+		val_p = RT5682_SDW_REF_1_176K;
+		val_c = RT5682_SDW_REF_2_176K;
+		break;
+	case 22050:
+		val_p = RT5682_SDW_REF_1_22K;
+		val_c = RT5682_SDW_REF_2_22K;
+		break;
+	case 11025:
+		val_p = RT5682_SDW_REF_1_11K;
+		val_c = RT5682_SDW_REF_2_11K;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	if (params_rate(params) <= 48000) {
+		osr_p = RT5682_DAC_OSR_D_8;
+		osr_c = RT5682_ADC_OSR_D_8;
+	} else if (params_rate(params) <= 96000) {
+		osr_p = RT5682_DAC_OSR_D_4;
+		osr_c = RT5682_ADC_OSR_D_4;
+	} else {
+		osr_p = RT5682_DAC_OSR_D_2;
+		osr_c = RT5682_ADC_OSR_D_2;
+	}
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		regmap_update_bits(rt5682->regmap, RT5682_SDW_REF_CLK,
+			RT5682_SDW_REF_1_MASK, val_p);
+		regmap_update_bits(rt5682->regmap, RT5682_ADDA_CLK_1,
+			RT5682_DAC_OSR_MASK, osr_p);
+	} else {
+		regmap_update_bits(rt5682->regmap, RT5682_SDW_REF_CLK,
+			RT5682_SDW_REF_2_MASK, val_c);
+		regmap_update_bits(rt5682->regmap, RT5682_ADDA_CLK_1,
+			RT5682_ADC_OSR_MASK, osr_c);
+	}
+
+	return retval;
+}
+
+static int rt5682_sdw_hw_free(struct snd_pcm_substream *substream,
+				struct snd_soc_dai *dai)
+{
+	struct snd_soc_component *component = dai->component;
+	struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
+	struct sdw_stream_data *stream =
+		snd_soc_dai_get_dma_data(dai, substream);
+
+	if (!rt5682->slave)
+		return -EINVAL;
+
+	sdw_stream_remove_slave(rt5682->slave, stream->sdw_stream);
+	return 0;
+}
+
+static struct snd_soc_dai_ops rt5682_sdw_ops = {
+	.hw_params	= rt5682_sdw_hw_params,
+	.hw_free	= rt5682_sdw_hw_free,
+	.set_sdw_stream	= rt5682_set_sdw_stream,
+	.shutdown	= rt5682_sdw_shutdown,
+};
+#endif
 
 static struct snd_soc_dai_driver rt5682_dai[] = {
 	{
···
 		},
 		.ops = &rt5682_aif2_dai_ops,
 	},
+#if IS_ENABLED(CONFIG_SND_SOC_RT5682_SDW)
+	{
+		.name = "rt5682-sdw",
+		.id = RT5682_SDW,
+		.playback = {
+			.stream_name = "SDW Playback",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = RT5682_STEREO_RATES,
+			.formats = RT5682_FORMATS,
+		},
+		.capture = {
+			.stream_name = "SDW Capture",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = RT5682_STEREO_RATES,
+			.formats = RT5682_FORMATS,
+		},
+		.ops = &rt5682_sdw_ops,
+	},
+#endif
 };
 
 static const struct snd_soc_component_driver soc_component_dev_rt5682 = {
···
 		&rt5682->pdata.jd_src);
 	device_property_read_u32(dev, "realtek,btndet-delay",
 		&rt5682->pdata.btndet_delay);
+	device_property_read_u32(dev, "realtek,dmic-clk-rate-hz",
+		&rt5682->pdata.dmic_clk_rate);
+	device_property_read_u32(dev, "realtek,dmic-delay-ms",
+		&rt5682->pdata.dmic_delay);
 
 	rt5682->pdata.ldo1_en = of_get_named_gpio(dev->of_node,
 		"realtek,ldo1-en-gpios", 0);
+
+	if (device_property_read_string_array(dev, "clock-output-names",
+					      rt5682->pdata.dai_clk_names,
+					      RT5682_DAI_NUM_CLKS) < 0)
+		dev_warn(dev, "Using default DAI clk names: %s, %s\n",
+			 rt5682->pdata.dai_clk_names[RT5682_DAI_WCLK_IDX],
+			 rt5682->pdata.dai_clk_names[RT5682_DAI_BCLK_IDX]);
 
 	return 0;
 }
···
 
 	mutex_lock(&rt5682->calibrate_mutex);
 
-	rt5682_reset(rt5682->regmap);
+	rt5682_reset(rt5682);
 	regmap_write(rt5682->regmap, RT5682_I2C_CTRL, 0x000f);
 	regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xa2af);
 	usleep_range(15000, 20000);
···
 	mutex_unlock(&rt5682->calibrate_mutex);
 
 }
+
+#if IS_ENABLED(CONFIG_SND_SOC_RT5682_SDW)
+static int rt5682_sdw_read(void *context, unsigned int reg, unsigned int *val)
+{
+	struct device *dev = context;
+	struct rt5682_priv *rt5682 = dev_get_drvdata(dev);
+	unsigned int data_l, data_h;
+
+	regmap_write(rt5682->sdw_regmap, RT5682_SDW_CMD, 0);
+	regmap_write(rt5682->sdw_regmap, RT5682_SDW_ADDR_H, (reg >> 8) & 0xff);
+	regmap_write(rt5682->sdw_regmap, RT5682_SDW_ADDR_L, (reg & 0xff));
+	regmap_read(rt5682->sdw_regmap, RT5682_SDW_DATA_H, &data_h);
+	regmap_read(rt5682->sdw_regmap, RT5682_SDW_DATA_L, &data_l);
+
+	*val = (data_h << 8) | data_l;
+
+	dev_vdbg(dev, "[%s] %04x => %04x\n", __func__, reg, *val);
+
+	return 0;
+}
+
+static int rt5682_sdw_write(void *context, unsigned int reg, unsigned int val)
+{
+	struct device *dev = context;
+	struct rt5682_priv *rt5682 = dev_get_drvdata(dev);
+
+	regmap_write(rt5682->sdw_regmap, RT5682_SDW_CMD, 1);
+	regmap_write(rt5682->sdw_regmap, RT5682_SDW_ADDR_H, (reg >> 8) & 0xff);
+	regmap_write(rt5682->sdw_regmap, RT5682_SDW_ADDR_L, (reg & 0xff));
+	regmap_write(rt5682->sdw_regmap, RT5682_SDW_DATA_H, (val >> 8) & 0xff);
+	regmap_write(rt5682->sdw_regmap, RT5682_SDW_DATA_L, (val & 0xff));
+
+	dev_vdbg(dev, "[%s] %04x <= %04x\n", __func__, reg, val);
+
+	return 0;
+}
+
+static const struct regmap_config rt5682_sdw_regmap = {
+	.reg_bits = 16,
+	.val_bits = 16,
+	.max_register = RT5682_I2C_MODE,
+	.volatile_reg = rt5682_volatile_register,
+	.readable_reg = rt5682_readable_register,
+	.cache_type = REGCACHE_RBTREE,
+	.reg_defaults = rt5682_reg,
+	.num_reg_defaults = ARRAY_SIZE(rt5682_reg),
+	.use_single_read = true,
+	.use_single_write = true,
+	.reg_read = rt5682_sdw_read,
+	.reg_write = rt5682_sdw_write,
+};
+
+int rt5682_sdw_init(struct device *dev, struct regmap *regmap,
+	struct sdw_slave *slave)
+{
+	struct rt5682_priv *rt5682;
+	int ret;
+
+	rt5682 = devm_kzalloc(dev, sizeof(*rt5682), GFP_KERNEL);
+	if (!rt5682)
+		return -ENOMEM;
+
+	dev_set_drvdata(dev, rt5682);
+	rt5682->slave = slave;
+	rt5682->sdw_regmap = regmap;
+	rt5682->is_sdw = true;
+
+	rt5682->regmap = devm_regmap_init(dev, NULL, dev, &rt5682_sdw_regmap);
+	if (IS_ERR(rt5682->regmap)) {
+		ret = PTR_ERR(rt5682->regmap);
+		dev_err(dev, "Failed to allocate register map: %d\n",
+			ret);
+		return ret;
+	}
+
+	/*
+	 * Mark hw_init to false
+	 * HW init will be performed when device reports present
+	 */
+	rt5682->hw_init = false;
+	rt5682->first_hw_init = false;
+
+	mutex_init(&rt5682->calibrate_mutex);
+	INIT_DELAYED_WORK(&rt5682->jack_detect_work,
+		rt5682_jack_detect_handler);
+
+	ret = devm_snd_soc_register_component(dev, &soc_component_dev_rt5682,
+		rt5682_dai, ARRAY_SIZE(rt5682_dai));
+
+	dev_dbg(&slave->dev, "%s\n", __func__);
+
+	return ret;
+}
+EXPORT_SYMBOL_GPL(rt5682_sdw_init);
+
+int rt5682_io_init(struct device *dev, struct sdw_slave *slave)
+{
+	struct rt5682_priv *rt5682 = dev_get_drvdata(dev);
+	int ret = 0;
+	unsigned int val;
+
+	if (rt5682->hw_init)
+		return 0;
+
+	regmap_read(rt5682->regmap, RT5682_DEVICE_ID, &val);
+	if (val != DEVICE_ID) {
+		pr_err("Device with ID register %x is not rt5682\n", val);
+		return -ENODEV;
+	}
+
+	/*
+	 * PM runtime is only enabled when a Slave reports as Attached
+	 */
+	if (!rt5682->first_hw_init) {
+		/* set autosuspend parameters */
+		pm_runtime_set_autosuspend_delay(&slave->dev, 3000);
+		pm_runtime_use_autosuspend(&slave->dev);
+
+		/* update count of parent 'active' children */
+		pm_runtime_set_active(&slave->dev);
+
+		/* make sure the device does not suspend immediately */
+		pm_runtime_mark_last_busy(&slave->dev);
+
+		pm_runtime_enable(&slave->dev);
+	}
+
+	pm_runtime_get_noresume(&slave->dev);
+
+	rt5682_reset(rt5682);
+
+	if (rt5682->first_hw_init) {
+		regcache_cache_only(rt5682->regmap, false);
+		regcache_cache_bypass(rt5682->regmap, true);
+	}
+
+	rt5682_calibrate(rt5682);
+
+	if (rt5682->first_hw_init) {
+		regcache_cache_bypass(rt5682->regmap, false);
+		regcache_mark_dirty(rt5682->regmap);
+		regcache_sync(rt5682->regmap);
+
+		/* volatile registers */
+		regmap_update_bits(rt5682->regmap, RT5682_CBJ_CTRL_2,
+			RT5682_EXT_JD_SRC, RT5682_EXT_JD_SRC_MANUAL);
+
+		goto reinit;
+	}
+
+	ret = regmap_multi_reg_write(rt5682->regmap, patch_list,
+				    ARRAY_SIZE(patch_list));
+	if (ret != 0)
+		dev_warn(dev, "Failed to apply regmap patch: %d\n", ret);
+
+	regmap_write(rt5682->regmap, RT5682_DEPOP_1, 0x0000);
+
+	regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_1,
+			RT5682_LDO1_DVO_MASK | RT5682_HP_DRIVER_MASK,
+			RT5682_LDO1_DVO_12 | RT5682_HP_DRIVER_5X);
+	regmap_write(rt5682->regmap, RT5682_MICBIAS_2, 0x0380);
+	regmap_write(rt5682->regmap, RT5682_TEST_MODE_CTRL_1, 0x0000);
+	regmap_update_bits(rt5682->regmap, RT5682_BIAS_CUR_CTRL_8,
+			RT5682_HPA_CP_BIAS_CTRL_MASK, RT5682_HPA_CP_BIAS_3UA);
+	regmap_update_bits(rt5682->regmap, RT5682_CHARGE_PUMP_1,
+			RT5682_CP_CLK_HP_MASK, RT5682_CP_CLK_HP_300KHZ);
+	regmap_update_bits(rt5682->regmap, RT5682_HP_CHARGE_PUMP_1,
+			RT5682_PM_HP_MASK, RT5682_PM_HP_HV);
+
+	/* Soundwire */
+	regmap_write(rt5682->regmap, RT5682_PLL2_INTERNAL, 0xa266);
+	regmap_write(rt5682->regmap, RT5682_PLL2_CTRL_1, 0x1700);
+	regmap_write(rt5682->regmap, RT5682_PLL2_CTRL_2, 0x0006);
+	regmap_write(rt5682->regmap, RT5682_PLL2_CTRL_3, 0x2600);
+	regmap_write(rt5682->regmap, RT5682_PLL2_CTRL_4, 0x0c8f);
+	regmap_write(rt5682->regmap, RT5682_PLL_TRACK_2, 0x3000);
+	regmap_write(rt5682->regmap, RT5682_PLL_TRACK_3, 0x4000);
+	regmap_update_bits(rt5682->regmap, RT5682_GLB_CLK,
+		RT5682_SCLK_SRC_MASK | RT5682_PLL2_SRC_MASK,
+		RT5682_SCLK_SRC_PLL2 | RT5682_PLL2_SRC_SDW);
+
+	regmap_update_bits(rt5682->regmap, RT5682_CBJ_CTRL_2,
+		RT5682_EXT_JD_SRC, RT5682_EXT_JD_SRC_MANUAL);
+	regmap_write(rt5682->regmap, RT5682_CBJ_CTRL_1, 0xd042);
+	regmap_update_bits(rt5682->regmap, RT5682_CBJ_CTRL_3,
+		RT5682_CBJ_IN_BUF_EN, RT5682_CBJ_IN_BUF_EN);
+	regmap_update_bits(rt5682->regmap, RT5682_SAR_IL_CMD_1,
+		RT5682_SAR_POW_MASK, RT5682_SAR_POW_EN);
+	regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL,
+			RT5682_POW_IRQ | RT5682_POW_JDH |
+			RT5682_POW_ANA, RT5682_POW_IRQ |
+			RT5682_POW_JDH | RT5682_POW_ANA);
+	regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_2,
+		RT5682_PWR_JDH, RT5682_PWR_JDH);
+	regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2,
+		RT5682_JD1_EN_MASK | RT5682_JD1_IRQ_MASK,
+		RT5682_JD1_EN | RT5682_JD1_IRQ_PUL);
+
+reinit:
+	mod_delayed_work(system_power_efficient_wq,
+		   &rt5682->jack_detect_work, msecs_to_jiffies(250));
+
+	/* Mark Slave initialization complete */
+	rt5682->hw_init = true;
+	rt5682->first_hw_init = true;
+
+	pm_runtime_mark_last_busy(&slave->dev);
+	pm_runtime_put_autosuspend(&slave->dev);
+
+	dev_dbg(&slave->dev, "%s hw_init complete\n", __func__);
+
+	return ret;
+}
+EXPORT_SYMBOL_GPL(rt5682_io_init);
+#endif
 
 static int rt5682_i2c_probe(struct i2c_client *i2c,
 		    const struct i2c_device_id *id)
···
 		return -ENODEV;
 	}
 
-	rt5682_reset(rt5682->regmap);
+	rt5682_reset(rt5682);
 
 	mutex_init(&rt5682->calibrate_mutex);
 	rt5682_calibrate(rt5682);
···
 			RT5682_CP_CLK_HP_MASK, RT5682_CP_CLK_HP_300KHZ);
 	regmap_update_bits(rt5682->regmap, RT5682_HP_CHARGE_PUMP_1,
 			RT5682_PM_HP_MASK, RT5682_PM_HP_HV);
+	regmap_update_bits(rt5682->regmap, RT5682_DMIC_CTRL_1,
+			RT5682_FIFO_CLK_DIV_MASK, RT5682_FIFO_CLK_DIV_2);
 
 	INIT_DELAYED_WORK(&rt5682->jack_detect_work,
 				rt5682_jack_detect_handler);
···
 {
 	struct rt5682_priv *rt5682 = i2c_get_clientdata(client);
 
-	rt5682_reset(rt5682->regmap);
+	rt5682_reset(rt5682);
 }
 
 #ifdef CONFIG_OF

sound/soc/codecs/rt5682.h

···
 #define __RT5682_H__
 
 #include <sound/rt5682.h>
+#include <linux/regulator/consumer.h>
+#include <linux/clk.h>
+#include <linux/clkdev.h>
+#include <linux/clk-provider.h>
+#include <linux/soundwire/sdw.h>
+#include <linux/soundwire/sdw_type.h>
 
 #define DEVICE_ID 0x6530
 
···
 #define RT5682_TEST_MODE_CTRL_4			0x0148
 #define RT5682_TEST_MODE_CTRL_5			0x0149
 #define RT5682_PLL1_INTERNAL			0x0150
-#define RT5682_PLL2_INTERNAL			0x0151
+#define RT5682_PLL2_INTERNAL			0x0156
 #define RT5682_STO_NG2_CTRL_1			0x0160
 #define RT5682_STO_NG2_CTRL_2			0x0161
 #define RT5682_STO_NG2_CTRL_3			0x0162
···
 #define RT5682_DMIC_1_EN_SFT			15
 #define RT5682_DMIC_1_DIS			(0x0 << 15)
 #define RT5682_DMIC_1_EN			(0x1 << 15)
+#define RT5682_FIFO_CLK_DIV_MASK		(0x7 << 12)
+#define RT5682_FIFO_CLK_DIV_2			(0x1 << 12)
 #define RT5682_DMIC_1_DP_MASK			(0x3 << 4)
 #define RT5682_DMIC_1_DP_SFT			4
 #define RT5682_DMIC_1_DP_GPIO2			(0x0 << 4)
···
 #define RT5682_ADC_OSR_D_24			(0x7 << 12)
 #define RT5682_ADC_OSR_D_32			(0x8 << 12)
 #define RT5682_ADC_OSR_D_48			(0x9 << 12)
-#define RT5682_I2S_M_DIV_MASK			(0xf << 12)
+#define RT5682_I2S_M_DIV_MASK			(0xf << 8)
 #define RT5682_I2S_M_DIV_SFT			8
 #define RT5682_I2S_M_D_1			(0x0 << 8)
 #define RT5682_I2S_M_D_2			(0x1 << 8)
···
 #define RT5682_TDM_DF_PCM_B			(0x3 << 11)
 #define RT5682_TDM_DF_PCM_A_N			(0x6 << 11)
 #define RT5682_TDM_DF_PCM_B_N			(0x7 << 11)
+#define RT5682_TDM_BCLK_MS1_MASK		(0x3 << 9)
+#define RT5682_TDM_BCLK_MS1_SFT			9
+#define RT5682_TDM_BCLK_MS1_32			(0x0 << 9)
+#define RT5682_TDM_BCLK_MS1_64			(0x1 << 9)
+#define RT5682_TDM_BCLK_MS1_128			(0x2 << 9)
+#define RT5682_TDM_BCLK_MS1_256			(0x3 << 9)
 #define RT5682_TDM_CL_MASK			(0x3 << 4)
 #define RT5682_TDM_CL_16			(0x0 << 4)
 #define RT5682_TDM_CL_20			(0x1 << 4)
···
 #define RT5682_TDM_M_LP_INV			(0x1 << 1)
 #define RT5682_TDM_MS_MASK			(0x1 << 0)
 #define RT5682_TDM_MS_SFT			0
-#define RT5682_TDM_MS_M				(0x0 << 0)
-#define RT5682_TDM_MS_S				(0x1 << 0)
+#define RT5682_TDM_MS_S				(0x0 << 0)
+#define RT5682_TDM_MS_M				(0x1 << 0)
 
 /* Global Clock Control (0x0080) */
 #define RT5682_SCLK_SRC_MASK			(0x7 << 13)
···
 #define RT5682_PWR_CLK1M_PD			(0x0 << 8)
 #define RT5682_PWR_CLK1M_PU			(0x1 << 8)
 
+/* PLL2 M/N/K Code Control 1 (0x009b) */
+#define RT5682_PLL2F_K_MASK			(0x1f << 8)
+#define RT5682_PLL2F_K_SFT			8
+#define RT5682_PLL2B_K_MASK			(0xf << 4)
+#define RT5682_PLL2B_K_SFT			4
+#define RT5682_PLL2B_M_MASK			(0xf << 0)
+
+/* PLL2 M/N/K Code Control 2 (0x009c) */
+#define RT5682_PLL2F_M_MASK			(0x3f << 8)
+#define RT5682_PLL2F_M_SFT			8
+#define RT5682_PLL2B_N_MASK			(0x3f << 0)
+
+/* PLL2 M/N/K Code Control 2 (0x009d) */
+#define RT5682_PLL2F_N_MASK			(0x7f << 8)
+#define RT5682_PLL2F_N_SFT			8
+
+/* PLL2 M/N/K Code Control 2 (0x009e) */
+#define RT5682_PLL2B_M_BP_MASK			(0x1 << 11)
+#define RT5682_PLL2B_M_BP_SFT			11
+#define RT5682_PLL2F_M_BP_MASK			(0x1 << 7)
+#define RT5682_PLL2F_M_BP_SFT			7
+
 /* RC Clock Control (0x009f) */
 #define RT5682_POW_IRQ				(0x1 << 15)
 #define RT5682_POW_JDH				(0x1 << 14)
···
 #define RT5682_JD1_POL_MASK			(0x1 << 13)
 #define RT5682_JD1_POL_NOR			(0x0 << 13)
 #define RT5682_JD1_POL_INV			(0x1 << 13)
+#define RT5682_JD1_IRQ_MASK			(0x1 << 10)
+#define RT5682_JD1_IRQ_LEV			(0x0 << 10)
+#define RT5682_JD1_IRQ_PUL			(0x1 << 10)
 
 /* IRQ Control 3 (0x00b8) */
 #define RT5682_IL_IRQ_MASK			(0x1 << 7)
 #define RT5682_IL_IRQ_DIS			(0x0 << 7)
 #define RT5682_IL_IRQ_EN			(0x1 << 7)
+#define RT5682_IL_IRQ_TYPE_MASK			(0x1 << 4)
+#define RT5682_IL_IRQ_LEV			(0x0 << 4)
+#define RT5682_IL_IRQ_PUL			(0x1 << 4)
 
 /* GPIO Control 1 (0x00c0) */
 #define RT5682_GP1_PIN_MASK			(0x3 << 14)
···
 	RT5682_PLL1_S_MCLK,
 	RT5682_PLL1_S_BCLK1,
 	RT5682_PLL1_S_RCCLK,
+	RT5682_PLL2_S_MCLK,
+};
+
+enum {
+	RT5682_PLL1,
+	RT5682_PLL2,
+	RT5682_PLLS,
 };
 
 enum {
 	RT5682_AIF1,
 	RT5682_AIF2,
+	RT5682_SDW,
 	RT5682_AIFS
 };
 
···
 	RT5682_CLK_SEL_I2S2_ASRC,
 };
 
+#define RT5682_NUM_SUPPLIES 3
+
+struct rt5682_priv {
+	struct snd_soc_component *component;
+	struct rt5682_platform_data pdata;
+	struct regmap *regmap;
+	struct regmap *sdw_regmap;
+	struct snd_soc_jack *hs_jack;
+	struct regulator_bulk_data supplies[RT5682_NUM_SUPPLIES];
+	struct delayed_work jack_detect_work;
+	struct delayed_work jd_check_work;
+	struct mutex calibrate_mutex;
+	struct sdw_slave *slave;
+	enum sdw_slave_status status;
+	struct sdw_bus_params params;
+	bool hw_init;
+	bool first_hw_init;
+	bool is_sdw;
+
+#ifdef CONFIG_COMMON_CLK
+	struct clk_hw dai_clks_hw[RT5682_DAI_NUM_CLKS];
+	struct clk_lookup *dai_clks_lookup[RT5682_DAI_NUM_CLKS];
+	struct clk *dai_clks[RT5682_DAI_NUM_CLKS];
+	struct clk *mclk;
+#endif
+
+	int sysclk;
+	int sysclk_src;
+	int lrck[RT5682_AIFS];
+	int bclk[RT5682_AIFS];
+	int master[RT5682_AIFS];
+
+	int pll_src[RT5682_PLLS];
+	int pll_in[RT5682_PLLS];
+	int pll_out[RT5682_PLLS];
+
+	int jack_type;
+};
+
 int rt5682_sel_asrc_clk_src(struct snd_soc_component *component,
 		unsigned int filter_mask, unsigned int clk_src);
+int rt5682_sdw_init(struct device *dev, struct regmap *regmap,
+	       struct sdw_slave *slave);
+int rt5682_io_init(struct device *dev, struct sdw_slave *slave);
 
 #endif /* __RT5682_H__ */

sound/soc/codecs/tas2562.c

···
 #define TAS2562_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |\
 			 SNDRV_PCM_FORMAT_S32_LE)
 
+/* DVC equation involves floating point math
+ * round(10^(volume in dB/20)*2^30)
+ * so create a lookup table for 2dB step
+ */
+static const unsigned int float_vol_db_lookup[] = {
+0x00000d43, 0x000010b2, 0x00001505, 0x00001a67, 0x00002151,
+0x000029f1, 0x000034cd, 0x00004279, 0x000053af, 0x0000695b,
+0x0000695b, 0x0000a6fa, 0x0000d236, 0x000108a4, 0x00014d2a,
+0x0001a36e, 0x00021008, 0x000298c0, 0x000344df, 0x00041d8f,
+0x00052e5a, 0x000685c8, 0x00083621, 0x000a566d, 0x000d03a7,
+0x0010624d, 0x0014a050, 0x0019f786, 0x0020b0bc, 0x0029279d,
+0x0033cf8d, 0x004139d3, 0x00521d50, 0x00676044, 0x0082248a,
+0x00a3d70a, 0x00ce4328, 0x0103ab3d, 0x0146e75d, 0x019b8c27,
+0x02061b89, 0x028c423f, 0x03352529, 0x0409c2b0, 0x05156d68,
+0x080e9f96, 0x0a24b062, 0x0cc509ab, 0x10137987, 0x143d1362,
+0x197a967f, 0x2013739e, 0x28619ae9, 0x32d64617, 0x40000000
+};
+
 struct tas2562_data {
 	struct snd_soc_component *component;
 	struct gpio_desc *sdz_gpio;
···
 	struct i2c_client *client;
 	int v_sense_slot;
 	int i_sense_slot;
+	int volume_lvl;
+};
+
+enum tas256x_model {
+	TAS2562,
+	TAS2563,
 };
 
 static int tas2562_set_bias_level(struct snd_soc_component *component,
···
 	struct snd_soc_component *component =
 					snd_soc_dapm_to_component(w->dapm);
 	struct tas2562_data *tas2562 = snd_soc_component_get_drvdata(component);
+	int ret;
 
 	switch (event) {
 	case SND_SOC_DAPM_POST_PMU:
-		dev_info(tas2562->dev, "SND_SOC_DAPM_POST_PMU\n");
+		ret = snd_soc_component_update_bits(component,
+			TAS2562_PWR_CTRL,
+			TAS2562_MODE_MASK,
+			TAS2562_MUTE);
+		if (ret)
+			goto end;
 		break;
 	case SND_SOC_DAPM_PRE_PMD:
-		dev_info(tas2562->dev, "SND_SOC_DAPM_PRE_PMD\n");
+		ret = snd_soc_component_update_bits(component,
+			TAS2562_PWR_CTRL,
+			TAS2562_MODE_MASK,
+			TAS2562_SHUTDOWN);
+		if (ret)
+			goto end;
 		break;
 	default:
-		break;
+		dev_err(tas2562->dev, "Not supported evevt\n");
+		return -EINVAL;
 	}
+
+end:
+	if (ret < 0)
+		return ret;
 
 	return 0;
 }
+
+static int tas2562_volume_control_get(struct snd_kcontrol *kcontrol,
+				      struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol);
+	struct tas2562_data *tas2562 = snd_soc_component_get_drvdata(component);
+
+	ucontrol->value.integer.value[0] = tas2562->volume_lvl;
+	return 0;
+}
+
+static int tas2562_volume_control_put(struct snd_kcontrol *kcontrol,
+				      struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol);
+	struct tas2562_data *tas2562 = snd_soc_component_get_drvdata(component);
+	int ret;
+	u32 reg_val;
+
+	reg_val = float_vol_db_lookup[ucontrol->value.integer.value[0]/2];
+	ret = snd_soc_component_write(component, TAS2562_DVC_CFG4,
+				      (reg_val & 0xff));
+	if (ret)
+		return ret;
+	ret = snd_soc_component_write(component, TAS2562_DVC_CFG3,
+				      ((reg_val >> 8) & 0xff));
+	if (ret)
+		return ret;
+	ret = snd_soc_component_write(component, TAS2562_DVC_CFG2,
+				      ((reg_val >> 16) & 0xff));
+	if (ret)
+		return ret;
+	ret = snd_soc_component_write(component, TAS2562_DVC_CFG1,
+				      ((reg_val >> 24) & 0xff));
+	if (ret)
+		return ret;
+
+	tas2562->volume_lvl = ucontrol->value.integer.value[0];
+
+	return ret;
+}
+
+/* Digital Volume Control. From 0 dB to -110 dB in 1 dB steps */
+static const DECLARE_TLV_DB_SCALE(dvc_tlv, -11000, 100, 0);
 
 static DECLARE_TLV_DB_SCALE(tas2562_dac_tlv, 850, 50, 0);
 
···
 			1, 1);
 
 static const struct snd_kcontrol_new tas2562_snd_controls[] = {
-	SOC_SINGLE_TLV("Amp Gain Volume", TAS2562_PB_CFG1, 0, 0x1c, 0,
+	SOC_SINGLE_TLV("Amp Gain Volume", TAS2562_PB_CFG1, 1, 0x1c, 0,
 		       tas2562_dac_tlv),
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Digital Volume Control",
+		.index = 0,
+		.tlv.p = dvc_tlv,
+		.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | SNDRV_CTL_ELEM_ACCESS_READWRITE,
+		.info = snd_soc_info_volsw,
+		.get = tas2562_volume_control_get,
+		.put = tas2562_volume_control_put,
+		.private_value = SOC_SINGLE_VALUE(TAS2562_DVC_CFG1, 0, 110, 0, 0) ,
+	},
 };
 
 static const struct snd_soc_dapm_widget tas2562_dapm_widgets[] = {
 	SND_SOC_DAPM_AIF_IN("ASI1", "ASI1 Playback", 0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_MUX("ASI1 Sel", SND_SOC_NOPM, 0, 0, &tas2562_asi1_mux),
-	SND_SOC_DAPM_AIF_IN("DAC IN", "Playback", 0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_DAC_E("DAC", NULL, SND_SOC_NOPM, 0, 0, tas2562_dac_event,
 			   SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
 	SND_SOC_DAPM_SWITCH("ISENSE", TAS2562_PWR_CTRL, 3, 1, &isense_switch),
···
 	{"ASI1 Sel", "Left", "ASI1"},
 	{"ASI1 Sel", "Right", "ASI1"},
 	{"ASI1 Sel", "LeftRightDiv2", "ASI1"},
-	{ "DAC", NULL, "DAC IN" },
+	{ "DAC", NULL, "ASI1 Sel" },
 	{ "OUT", NULL, "DAC" },
 	{"ISENSE", "Switch", "IMON"},
 	{"VSENSE", "Switch", "VMON"},
···
 			.rates      = SNDRV_PCM_RATE_8000_192000,
 			.formats    = TAS2562_FORMATS,
 		},
+		.capture = {
+			.stream_name    = "ASI1 Capture",
+			.channels_min   = 0,
+			.channels_max   = 2,
+			.rates		= SNDRV_PCM_RATE_8000_192000,
+			.formats	= TAS2562_FORMATS,
+		},
 		.ops = &tas2562_speaker_dai_ops,
 	},
 };
···
 	{ TAS2562_PB_CFG1, 0x20 },
 	{ TAS2562_TDM_CFG0, 0x09 },
 	{ TAS2562_TDM_CFG1, 0x02 },
+	{ TAS2562_DVC_CFG1, 0x40 },
+	{ TAS2562_DVC_CFG2, 0x40 },
+	{ TAS2562_DVC_CFG3, 0x00 },
+	{ TAS2562_DVC_CFG4, 0x00 },
 };
 
 static const struct regmap_config tas2562_regmap_config = {
···
 }
 
 static const struct i2c_device_id tas2562_id[] = {
-	{ "tas2562", 0 },
+	{ "tas2562", TAS2562 },
+	{ "tas2563", TAS2563 },
 	{ }
 };
 MODULE_DEVICE_TABLE(i2c, tas2562_id);
 
 static const struct of_device_id tas2562_of_match[] = {
 	{ .compatible = "ti,tas2562", },
+	{ .compatible = "ti,tas2563", },
 	{ },
 };
 MODULE_DEVICE_TABLE(of, tas2562_of_match);

sound/soc/codecs/tas2562.h

···
 #define TAS2562_REV_ID		TAS2562_REG(0, 0x7d)
 
 /* Page 2 */
-#define TAS2562_DVC_CFG1	TAS2562_REG(2, 0x01)
-#define TAS2562_DVC_CFG2	TAS2562_REG(2, 0x02)
+#define TAS2562_DVC_CFG1	TAS2562_REG(2, 0x0c)
+#define TAS2562_DVC_CFG2	TAS2562_REG(2, 0x0d)
+#define TAS2562_DVC_CFG3	TAS2562_REG(2, 0x0e)
+#define TAS2562_DVC_CFG4	TAS2562_REG(2, 0x0f)
 
 #define TAS2562_RESET	BIT(0)
 
-#define TAS2562_MODE_MASK	0x3
+#define TAS2562_MODE_MASK	GENMASK(1,0)
 #define TAS2562_ACTIVE		0x0
 #define TAS2562_MUTE		0x1
 #define TAS2562_SHUTDOWN	0x2
···
 #define TAS2562_TDM_CFG2_RXWLEN_24B	BIT(3)
 #define TAS2562_TDM_CFG2_RXWLEN_32B	(BIT(2) | BIT(3))
 
-#define TAS2562_VSENSE_POWER_EN		BIT(2)
-#define TAS2562_ISENSE_POWER_EN		BIT(3)
+#define TAS2562_VSENSE_POWER_EN		2
+#define TAS2562_ISENSE_POWER_EN		3
 
 #define TAS2562_TDM_CFG5_VSNS_EN	BIT(6)
 #define TAS2562_TDM_CFG5_VSNS_SLOT_MASK	GENMASK(5, 0)

sound/soc/codecs/tlv320adcx140.c

···
+// SPDX-License-Identifier: GPL-2.0
+// TLV320ADCX140 Sound driver
+// Copyright (C) 2020 Texas Instruments Incorporated - http://www.ti.com/
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/gpio/consumer.h>
+#include <linux/regulator/consumer.h>
+#include <linux/acpi.h>
+#include <linux/of.h>
+#include <linux/of_gpio.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+
+#include "tlv320adcx140.h"
+
+struct adcx140_priv {
+	struct snd_soc_component *component;
+	struct regulator *supply_areg;
+	struct gpio_desc *gpio_reset;
+	struct regmap *regmap;
+	struct device *dev;
+
+	int micbias_vg;
+
+	unsigned int dai_fmt;
+	unsigned int tdm_delay;
+	unsigned int slot_width;
+};
+
+static const struct reg_default adcx140_reg_defaults[] = {
+	{ ADCX140_PAGE_SELECT, 0x00 },
+	{ ADCX140_SW_RESET, 0x00 },
+	{ ADCX140_SLEEP_CFG, 0x00 },
+	{ ADCX140_SHDN_CFG, 0x05 },
+	{ ADCX140_ASI_CFG0, 0x30 },
+	{ ADCX140_ASI_CFG1, 0x00 },
+	{ ADCX140_ASI_CFG2, 0x00 },
+	{ ADCX140_ASI_CH1, 0x00 },
+	{ ADCX140_ASI_CH2, 0x01 },
+	{ ADCX140_ASI_CH3, 0x02 },
+	{ ADCX140_ASI_CH4, 0x03 },
+	{ ADCX140_ASI_CH5, 0x04 },
+	{ ADCX140_ASI_CH6, 0x05 },
+	{ ADCX140_ASI_CH7, 0x06 },
+	{ ADCX140_ASI_CH8, 0x07 },
+	{ ADCX140_MST_CFG0, 0x02 },
+	{ ADCX140_MST_CFG1, 0x48 },
+	{ ADCX140_ASI_STS, 0xff },
+	{ ADCX140_CLK_SRC, 0x10 },
+	{ ADCX140_PDMCLK_CFG, 0x40 },
+	{ ADCX140_PDM_CFG, 0x00 },
+	{ ADCX140_GPIO_CFG0, 0x22 },
+	{ ADCX140_GPO_CFG1, 0x00 },
+	{ ADCX140_GPO_CFG2, 0x00 },
+	{ ADCX140_GPO_CFG3, 0x00 },
+	{ ADCX140_GPO_CFG4, 0x00 },
+	{ ADCX140_GPO_VAL, 0x00 },
+	{ ADCX140_GPIO_MON, 0x00 },
+	{ ADCX140_GPI_CFG0, 0x00 },
+	{ ADCX140_GPI_CFG1, 0x00 },
+	{ ADCX140_GPI_MON, 0x00 },
+	{ ADCX140_INT_CFG, 0x00 },
+	{ ADCX140_INT_MASK0, 0xff },
+	{ ADCX140_INT_LTCH0, 0x00 },
+	{ ADCX140_BIAS_CFG, 0x00 },
+	{ ADCX140_CH1_CFG0, 0x00 },
+	{ ADCX140_CH1_CFG1, 0x00 },
+	{ ADCX140_CH1_CFG2, 0xc9 },
+	{ ADCX140_CH1_CFG3, 0x80 },
+	{ ADCX140_CH1_CFG4, 0x00 },
+	{ ADCX140_CH2_CFG0, 0x00 },
+	{ ADCX140_CH2_CFG1, 0x00 },
+	{ ADCX140_CH2_CFG2, 0xc9 },
+	{ ADCX140_CH2_CFG3, 0x80 },
+	{ ADCX140_CH2_CFG4, 0x00 },
+	{ ADCX140_CH3_CFG0, 0x00 },
+	{ ADCX140_CH3_CFG1, 0x00 },
+	{ ADCX140_CH3_CFG2, 0xc9 },
+	{ ADCX140_CH3_CFG3, 0x80 },
+	{ ADCX140_CH3_CFG4, 0x00 },
+	{ ADCX140_CH4_CFG0, 0x00 },
+	{ ADCX140_CH4_CFG1, 0x00 },
+	{ ADCX140_CH4_CFG2, 0xc9 },
+	{ ADCX140_CH4_CFG3, 0x80 },
+	{ ADCX140_CH4_CFG4, 0x00 },
+	{ ADCX140_CH5_CFG2, 0xc9 },
+	{ ADCX140_CH5_CFG3, 0x80 },
+	{ ADCX140_CH5_CFG4, 0x00 },
+	{ ADCX140_CH6_CFG2, 0xc9 },
+	{ ADCX140_CH6_CFG3, 0x80 },
+	{ ADCX140_CH6_CFG4, 0x00 },
+	{ ADCX140_CH7_CFG2, 0xc9 },
+	{ ADCX140_CH7_CFG3, 0x80 },
+	{ ADCX140_CH7_CFG4, 0x00 },
+	{ ADCX140_CH8_CFG2, 0xc9 },
+	{ ADCX140_CH8_CFG3, 0x80 },
+	{ ADCX140_CH8_CFG4, 0x00 },
+	{ ADCX140_DSP_CFG0, 0x01 },
+	{ ADCX140_DSP_CFG1, 0x40 },
+	{ ADCX140_DRE_CFG0, 0x7b },
+	{ ADCX140_AGC_CFG0, 0xe7 },
+	{ ADCX140_IN_CH_EN, 0xf0 },
+	{ ADCX140_ASI_OUT_CH_EN, 0x00 },
+	{ ADCX140_PWR_CFG, 0x00 },
+	{ ADCX140_DEV_STS0, 0x00 },
+	{ ADCX140_DEV_STS1, 0x80 },
+};
+
+static const struct regmap_range_cfg adcx140_ranges[] = {
+	{
+		.range_min = 0,
+		.range_max = 12 * 128,
+		.selector_reg = ADCX140_PAGE_SELECT,
+		.selector_mask = 0xff,
+		.selector_shift = 0,
+		.window_start = 0,
+		.window_len = 128,
+	},
+};
+
+static bool adcx140_volatile(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case ADCX140_SW_RESET:
+	case ADCX140_DEV_STS0:
+	case ADCX140_DEV_STS1:
+	case ADCX140_ASI_STS:
+		return true;
+	default:
+		return false;
+	}
+}
+
+static const struct regmap_config adcx140_i2c_regmap = {
+	.reg_bits = 8,
+	.val_bits = 8,
+	.reg_defaults = adcx140_reg_defaults,
+	.num_reg_defaults = ARRAY_SIZE(adcx140_reg_defaults),
+	.cache_type = REGCACHE_FLAT,
+	.ranges = adcx140_ranges,
+	.num_ranges = ARRAY_SIZE(adcx140_ranges),
+	.max_register = 12 * 128,
+	.volatile_reg = adcx140_volatile,
+};
+
+/* Digital Volume control. From -100 to 27 dB in 0.5 dB steps */
+static DECLARE_TLV_DB_SCALE(dig_vol_tlv, -10000, 50, 0);
+
+/* ADC gain. From 0 to 42 dB in 1 dB steps */
+static DECLARE_TLV_DB_SCALE(adc_tlv, 0, 100, 0);
+
+/* DRE Level. From -12 dB to -66 dB in 1 dB steps */
+static DECLARE_TLV_DB_SCALE(dre_thresh_tlv, -6600, 100, 0);
+/* DRE Max Gain. From 2 dB to 26 dB in 2 dB steps */
+static DECLARE_TLV_DB_SCALE(dre_gain_tlv, 200, 200, 0);
+
+/* AGC Level. From -6 dB to -36 dB in 2 dB steps */
+static DECLARE_TLV_DB_SCALE(agc_thresh_tlv, -3600, 200, 0);
+/* AGC Max Gain. From 3 dB to 42 dB in 3 dB steps */
+static DECLARE_TLV_DB_SCALE(agc_gain_tlv, 300, 300, 0);
+
+static const char * const decimation_filter_text[] = {
+	"Linear Phase", "Low Latency", "Ultra-low Latency"
+};
+
+static SOC_ENUM_SINGLE_DECL(decimation_filter_enum, ADCX140_DSP_CFG0, 4,
+			    decimation_filter_text);
+
+static const struct snd_kcontrol_new decimation_filter_controls[] = {
+	SOC_DAPM_ENUM("Decimation Filter", decimation_filter_enum),
+};
+
+static const char * const resistor_text[] = {
+	"2.5 kOhm", "10 kOhm", "20 kOhm"
+};
+
+static SOC_ENUM_SINGLE_DECL(in1_resistor_enum, ADCX140_CH1_CFG0, 2,
+			    resistor_text);
+static SOC_ENUM_SINGLE_DECL(in2_resistor_enum, ADCX140_CH2_CFG0, 2,
+			    resistor_text);
+static SOC_ENUM_SINGLE_DECL(in3_resistor_enum, ADCX140_CH3_CFG0, 2,
+			    resistor_text);
+static SOC_ENUM_SINGLE_DECL(in4_resistor_enum, ADCX140_CH4_CFG0, 2,
+			    resistor_text);
+
+static const struct snd_kcontrol_new in1_resistor_controls[] = {
+	SOC_DAPM_ENUM("CH1 Resistor Select", in1_resistor_enum),
+};
+static const struct snd_kcontrol_new in2_resistor_controls[] = {
+	SOC_DAPM_ENUM("CH2 Resistor Select", in2_resistor_enum),
+};
+static const struct snd_kcontrol_new in3_resistor_controls[] = {
+	SOC_DAPM_ENUM("CH3 Resistor Select", in3_resistor_enum),
+};
+static const struct snd_kcontrol_new in4_resistor_controls[] = {
+	SOC_DAPM_ENUM("CH4 Resistor Select", in4_resistor_enum),
+};
+
+/* Analog/Digital Selection */
+static const char *adcx140_mic_sel_text[] = {"Analog", "Line In", "Digital"};
+static const char *adcx140_analog_sel_text[] = {"Analog", "Line In"};
+
+static SOC_ENUM_SINGLE_DECL(adcx140_mic1p_enum,
+			    ADCX140_CH1_CFG0, 5,
+			    adcx140_mic_sel_text);
+
+static const struct snd_kcontrol_new adcx140_dapm_mic1p_control =
+SOC_DAPM_ENUM("MIC1P MUX", adcx140_mic1p_enum);
+
+static SOC_ENUM_SINGLE_DECL(adcx140_mic1_analog_enum,
+			    ADCX140_CH1_CFG0, 7,
+			    adcx140_analog_sel_text);
+
+static const struct snd_kcontrol_new adcx140_dapm_mic1_analog_control =
+SOC_DAPM_ENUM("MIC1 Analog MUX", adcx140_mic1_analog_enum);
+
+static SOC_ENUM_SINGLE_DECL(adcx140_mic1m_enum,
+			    ADCX140_CH1_CFG0, 5,
+			    adcx140_mic_sel_text);
+
+static const struct snd_kcontrol_new adcx140_dapm_mic1m_control =
+SOC_DAPM_ENUM("MIC1M MUX", adcx140_mic1m_enum);
+
+static SOC_ENUM_SINGLE_DECL(adcx140_mic2p_enum,
+			    ADCX140_CH2_CFG0, 5,
+			    adcx140_mic_sel_text);
+
+static const struct snd_kcontrol_new adcx140_dapm_mic2p_control =
+SOC_DAPM_ENUM("MIC2P MUX", adcx140_mic2p_enum);
+
+static SOC_ENUM_SINGLE_DECL(adcx140_mic2_analog_enum,
+			    ADCX140_CH2_CFG0, 7,
+			    adcx140_analog_sel_text);
+
+static const struct snd_kcontrol_new adcx140_dapm_mic2_analog_control =
+SOC_DAPM_ENUM("MIC2 Analog MUX", adcx140_mic2_analog_enum);
+
+static SOC_ENUM_SINGLE_DECL(adcx140_mic2m_enum,
+			    ADCX140_CH2_CFG0, 5,
+			    adcx140_mic_sel_text);
+
+static const struct snd_kcontrol_new adcx140_dapm_mic2m_control =
+SOC_DAPM_ENUM("MIC2M MUX", adcx140_mic2m_enum);
+
+static SOC_ENUM_SINGLE_DECL(adcx140_mic3p_enum,
+			    ADCX140_CH3_CFG0, 5,
+			    adcx140_mic_sel_text);
+
+static const struct snd_kcontrol_new adcx140_dapm_mic3p_control =
+SOC_DAPM_ENUM("MIC3P MUX", adcx140_mic3p_enum);
+
+static SOC_ENUM_SINGLE_DECL(adcx140_mic3_analog_enum,
+			    ADCX140_CH3_CFG0, 7,
+			    adcx140_analog_sel_text);
+
+static const struct snd_kcontrol_new adcx140_dapm_mic3_analog_control =
+SOC_DAPM_ENUM("MIC3 Analog MUX", adcx140_mic3_analog_enum);
+
+static SOC_ENUM_SINGLE_DECL(adcx140_mic3m_enum,
+			    ADCX140_CH3_CFG0, 5,
+			    adcx140_mic_sel_text);
+
+static const struct snd_kcontrol_new adcx140_dapm_mic3m_control =
+SOC_DAPM_ENUM("MIC3M MUX", adcx140_mic3m_enum);
+
+static SOC_ENUM_SINGLE_DECL(adcx140_mic4p_enum,
+			    ADCX140_CH4_CFG0, 5,
+			    adcx140_mic_sel_text);
+
+static const struct snd_kcontrol_new adcx140_dapm_mic4p_control =
+SOC_DAPM_ENUM("MIC4P MUX", adcx140_mic4p_enum);
+
+static SOC_ENUM_SINGLE_DECL(adcx140_mic4_analog_enum,
+			    ADCX140_CH4_CFG0, 7,
+			    adcx140_analog_sel_text);
+
+static const struct snd_kcontrol_new adcx140_dapm_mic4_analog_control =
+SOC_DAPM_ENUM("MIC4 Analog MUX", adcx140_mic4_analog_enum);
+
+static SOC_ENUM_SINGLE_DECL(adcx140_mic4m_enum,
+			    ADCX140_CH4_CFG0, 5,
+			    adcx140_mic_sel_text);
+
+static const struct snd_kcontrol_new adcx140_dapm_mic4m_control =
+SOC_DAPM_ENUM("MIC4M MUX", adcx140_mic4m_enum);
+
+static const struct snd_kcontrol_new adcx140_dapm_ch1_en_switch =
+	SOC_DAPM_SINGLE("Switch", ADCX140_ASI_OUT_CH_EN, 7, 1, 0);
+static const struct snd_kcontrol_new adcx140_dapm_ch2_en_switch =
+	SOC_DAPM_SINGLE("Switch", ADCX140_ASI_OUT_CH_EN, 6, 1, 0);
+static const struct snd_kcontrol_new adcx140_dapm_ch3_en_switch =
+	SOC_DAPM_SINGLE("Switch", ADCX140_ASI_OUT_CH_EN, 5, 1, 0);
+static const struct snd_kcontrol_new adcx140_dapm_ch4_en_switch =
+	SOC_DAPM_SINGLE("Switch", ADCX140_ASI_OUT_CH_EN, 4, 1, 0);
+
+static const struct snd_kcontrol_new adcx140_dapm_ch1_dre_en_switch =
+	SOC_DAPM_SINGLE("Switch", ADCX140_CH1_CFG0, 0, 1, 0);
+static const struct snd_kcontrol_new adcx140_dapm_ch2_dre_en_switch =
+	SOC_DAPM_SINGLE("Switch", ADCX140_CH2_CFG0, 0, 1, 0);
+static const struct snd_kcontrol_new adcx140_dapm_ch3_dre_en_switch =
+	SOC_DAPM_SINGLE("Switch", ADCX140_CH3_CFG0, 0, 1, 0);
+static const struct snd_kcontrol_new adcx140_dapm_ch4_dre_en_switch =
+	SOC_DAPM_SINGLE("Switch", ADCX140_CH4_CFG0, 0, 1, 0);
+
+static const struct snd_kcontrol_new adcx140_dapm_dre_en_switch =
+	SOC_DAPM_SINGLE("Switch", ADCX140_DSP_CFG1, 3, 1, 0);
+
+/* Output Mixer */
+static const struct snd_kcontrol_new adcx140_output_mixer_controls[] = {
+	SOC_DAPM_SINGLE("Digital CH1 Switch", 0, 0, 0, 0),
+	SOC_DAPM_SINGLE("Digital CH2 Switch", 0, 0, 0, 0),
+	SOC_DAPM_SINGLE("Digital CH3 Switch", 0, 0, 0, 0),
+	SOC_DAPM_SINGLE("Digital CH4 Switch", 0, 0, 0, 0),
+};
+
+static const struct snd_soc_dapm_widget adcx140_dapm_widgets[] = {
+	/* Analog Differential Inputs */
+	SND_SOC_DAPM_INPUT("MIC1P"),
+	SND_SOC_DAPM_INPUT("MIC1M"),
+	SND_SOC_DAPM_INPUT("MIC2P"),
+	SND_SOC_DAPM_INPUT("MIC2M"),
+	SND_SOC_DAPM_INPUT("MIC3P"),
+	SND_SOC_DAPM_INPUT("MIC3M"),
+	SND_SOC_DAPM_INPUT("MIC4P"),
+	SND_SOC_DAPM_INPUT("MIC4M"),
+
+	SND_SOC_DAPM_OUTPUT("CH1_OUT"),
+	SND_SOC_DAPM_OUTPUT("CH2_OUT"),
+	SND_SOC_DAPM_OUTPUT("CH3_OUT"),
+	SND_SOC_DAPM_OUTPUT("CH4_OUT"),
+	SND_SOC_DAPM_OUTPUT("CH5_OUT"),
+	SND_SOC_DAPM_OUTPUT("CH6_OUT"),
+	SND_SOC_DAPM_OUTPUT("CH7_OUT"),
+	SND_SOC_DAPM_OUTPUT("CH8_OUT"),
+
+	SND_SOC_DAPM_MIXER("Output Mixer", SND_SOC_NOPM, 0, 0,
+		&adcx140_output_mixer_controls[0],
+		ARRAY_SIZE(adcx140_output_mixer_controls)),
+
+	/* Input Selection to MIC_PGA */
+	SND_SOC_DAPM_MUX("MIC1P Input Mux", SND_SOC_NOPM, 0, 0,
+			 &adcx140_dapm_mic1p_control),
+	SND_SOC_DAPM_MUX("MIC2P Input Mux", SND_SOC_NOPM, 0, 0,
+			 &adcx140_dapm_mic2p_control),
+	SND_SOC_DAPM_MUX("MIC3P Input Mux", SND_SOC_NOPM, 0, 0,
+			 &adcx140_dapm_mic3p_control),
+	SND_SOC_DAPM_MUX("MIC4P Input Mux", SND_SOC_NOPM, 0, 0,
+			 &adcx140_dapm_mic4p_control),
+
+	/* Input Selection to MIC_PGA */
+	SND_SOC_DAPM_MUX("MIC1 Analog Mux", SND_SOC_NOPM, 0, 0,
+			 &adcx140_dapm_mic1_analog_control),
+	SND_SOC_DAPM_MUX("MIC2 Analog Mux", SND_SOC_NOPM, 0, 0,
+			 &adcx140_dapm_mic2_analog_control),
+	SND_SOC_DAPM_MUX("MIC3 Analog Mux", SND_SOC_NOPM, 0, 0,
+			 &adcx140_dapm_mic3_analog_control),
+	SND_SOC_DAPM_MUX("MIC4 Analog Mux", SND_SOC_NOPM, 0, 0,
+			 &adcx140_dapm_mic4_analog_control),
+
+	SND_SOC_DAPM_MUX("MIC1M Input Mux", SND_SOC_NOPM, 0, 0,
+			 &adcx140_dapm_mic1m_control),
+	SND_SOC_DAPM_MUX("MIC2M Input Mux", SND_SOC_NOPM, 0, 0,
+			 &adcx140_dapm_mic2m_control),
+	SND_SOC_DAPM_MUX("MIC3M Input Mux", SND_SOC_NOPM, 0, 0,
+			 &adcx140_dapm_mic3m_control),
+	SND_SOC_DAPM_MUX("MIC4M Input Mux", SND_SOC_NOPM, 0, 0,
+			 &adcx140_dapm_mic4m_control),
+
+	SND_SOC_DAPM_PGA("MIC_GAIN_CTL_CH1", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("MIC_GAIN_CTL_CH2", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("MIC_GAIN_CTL_CH3", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("MIC_GAIN_CTL_CH4", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+	SND_SOC_DAPM_ADC("CH1_ADC", "CH1 Capture", ADCX140_IN_CH_EN, 7, 0),
+	SND_SOC_DAPM_ADC("CH2_ADC", "CH2 Capture", ADCX140_IN_CH_EN, 6, 0),
+	SND_SOC_DAPM_ADC("CH3_ADC", "CH3 Capture", ADCX140_IN_CH_EN, 5, 0),
+	SND_SOC_DAPM_ADC("CH4_ADC", "CH4 Capture", ADCX140_IN_CH_EN, 4, 0),
+
+	SND_SOC_DAPM_SWITCH("CH1_ASI_EN", SND_SOC_NOPM, 0, 0,
+			    &adcx140_dapm_ch1_en_switch),
+	SND_SOC_DAPM_SWITCH("CH2_ASI_EN", SND_SOC_NOPM, 0, 0,
+			    &adcx140_dapm_ch2_en_switch),
+	SND_SOC_DAPM_SWITCH("CH3_ASI_EN", SND_SOC_NOPM, 0, 0,
+			    &adcx140_dapm_ch3_en_switch),
+	SND_SOC_DAPM_SWITCH("CH4_ASI_EN", SND_SOC_NOPM, 0, 0,
+			    &adcx140_dapm_ch4_en_switch),
+
+	SND_SOC_DAPM_SWITCH("DRE_ENABLE", SND_SOC_NOPM, 0, 0,
+			    &adcx140_dapm_dre_en_switch),
+
+	SND_SOC_DAPM_SWITCH("CH1_DRE_EN", SND_SOC_NOPM, 0, 0,
+			    &adcx140_dapm_ch1_dre_en_switch),
+	SND_SOC_DAPM_SWITCH("CH2_DRE_EN", SND_SOC_NOPM, 0, 0,
+			    &adcx140_dapm_ch2_dre_en_switch),
+	SND_SOC_DAPM_SWITCH("CH3_DRE_EN", SND_SOC_NOPM, 0, 0,
+			    &adcx140_dapm_ch3_dre_en_switch),
+	SND_SOC_DAPM_SWITCH("CH4_DRE_EN", SND_SOC_NOPM, 0, 0,
+			    &adcx140_dapm_ch4_dre_en_switch),
+
+	SND_SOC_DAPM_MUX("IN1 Analog Mic Resistor", SND_SOC_NOPM, 0, 0,
+			in1_resistor_controls),
+	SND_SOC_DAPM_MUX("IN2 Analog Mic Resistor", SND_SOC_NOPM, 0, 0,
+			in2_resistor_controls),
+	SND_SOC_DAPM_MUX("IN3 Analog Mic Resistor", SND_SOC_NOPM, 0, 0,
+			in3_resistor_controls),
+	SND_SOC_DAPM_MUX("IN4 Analog Mic Resistor", SND_SOC_NOPM, 0, 0,
+			in4_resistor_controls),
+
+	SND_SOC_DAPM_MUX("Decimation Filter", SND_SOC_NOPM, 0, 0,
+			decimation_filter_controls),
+};
+
+static const struct snd_soc_dapm_route adcx140_audio_map[] = {
+	/* Outputs */
+	{"CH1_OUT", NULL, "Output Mixer"},
+	{"CH2_OUT", NULL, "Output Mixer"},
+	{"CH3_OUT", NULL, "Output Mixer"},
+	{"CH4_OUT", NULL, "Output Mixer"},
+
+	{"CH1_ASI_EN", "Switch", "CH1_ADC"},
+	{"CH2_ASI_EN", "Switch", "CH2_ADC"},
+	{"CH3_ASI_EN", "Switch", "CH3_ADC"},
+	{"CH4_ASI_EN", "Switch", "CH4_ADC"},
+
+	{"Decimation Filter", "Linear Phase", "DRE_ENABLE"},
+	{"Decimation Filter", "Low Latency", "DRE_ENABLE"},
+	{"Decimation Filter", "Ultra-low Latency", "DRE_ENABLE"},
+
+	{"DRE_ENABLE", "Switch", "CH1_DRE_EN"},
+	{"DRE_ENABLE", "Switch", "CH2_DRE_EN"},
+	{"DRE_ENABLE", "Switch", "CH3_DRE_EN"},
+	{"DRE_ENABLE", "Switch", "CH4_DRE_EN"},
+
+	{"CH1_DRE_EN", "Switch", "CH1_ADC"},
+	{"CH2_DRE_EN", "Switch", "CH2_ADC"},
+	{"CH3_DRE_EN", "Switch", "CH3_ADC"},
+	{"CH4_DRE_EN", "Switch", "CH4_ADC"},
+
+	/* Mic input */
+	{"CH1_ADC", NULL, "MIC_GAIN_CTL_CH1"},
+	{"CH2_ADC", NULL, "MIC_GAIN_CTL_CH2"},
+	{"CH3_ADC", NULL, "MIC_GAIN_CTL_CH3"},
+	{"CH4_ADC", NULL, "MIC_GAIN_CTL_CH4"},
+
+	{"MIC_GAIN_CTL_CH1", NULL, "IN1 Analog Mic Resistor"},
+	{"MIC_GAIN_CTL_CH1", NULL, "IN1 Analog Mic Resistor"},
+	{"MIC_GAIN_CTL_CH2", NULL, "IN2 Analog Mic Resistor"},
+	{"MIC_GAIN_CTL_CH2", NULL, "IN2 Analog Mic Resistor"},
+	{"MIC_GAIN_CTL_CH3", NULL, "IN3 Analog Mic Resistor"},
+	{"MIC_GAIN_CTL_CH3", NULL, "IN3 Analog Mic Resistor"},
+	{"MIC_GAIN_CTL_CH4", NULL, "IN4 Analog Mic Resistor"},
+	{"MIC_GAIN_CTL_CH4", NULL, "IN4 Analog Mic Resistor"},
+
+	{"IN1 Analog Mic Resistor", "2.5 kOhm", "MIC1P Input Mux"},
+	{"IN1 Analog Mic Resistor", "10 kOhm", "MIC1P Input Mux"},
+	{"IN1 Analog Mic Resistor", "20 kOhm", "MIC1P Input Mux"},
+
+	{"IN1 Analog Mic Resistor", "2.5 kOhm", "MIC1M Input Mux"},
+	{"IN1 Analog Mic Resistor", "10 kOhm", "MIC1M Input Mux"},
+	{"IN1 Analog Mic Resistor", "20 kOhm", "MIC1M Input Mux"},
+
+	{"IN2 Analog Mic Resistor", "2.5 kOhm", "MIC2P Input Mux"},
+	{"IN2 Analog Mic Resistor", "10 kOhm", "MIC2P Input Mux"},
+	{"IN2 Analog Mic Resistor", "20 kOhm", "MIC2P Input Mux"},
+
+	{"IN2 Analog Mic Resistor", "2.5 kOhm", "MIC2M Input Mux"},
+	{"IN2 Analog Mic Resistor", "10 kOhm", "MIC2M Input Mux"},
+	{"IN2 Analog Mic Resistor", "20 kOhm", "MIC2M Input Mux"},
+
+	{"IN3 Analog Mic Resistor", "2.5 kOhm", "MIC3P Input Mux"},
+	{"IN3 Analog Mic Resistor", "10 kOhm", "MIC3P Input Mux"},
+	{"IN3 Analog Mic Resistor", "20 kOhm", "MIC3P Input Mux"},
+
+	{"IN3 Analog Mic Resistor", "2.5 kOhm", "MIC3M Input Mux"},
+	{"IN3 Analog Mic Resistor", "10 kOhm", "MIC3M Input Mux"},
+	{"IN3 Analog Mic Resistor", "20 kOhm", "MIC3M Input Mux"},
+
+	{"IN4 Analog Mic Resistor", "2.5 kOhm", "MIC4P Input Mux"},
+	{"IN4 Analog Mic Resistor", "10 kOhm", "MIC4P Input Mux"},
+	{"IN4 Analog Mic Resistor", "20 kOhm", "MIC4P Input Mux"},
+
+	{"IN4 Analog Mic Resistor", "2.5 kOhm", "MIC4M Input Mux"},
+	{"IN4 Analog Mic Resistor", "10 kOhm", "MIC4M Input Mux"},
+	{"IN4 Analog Mic Resistor", "20 kOhm", "MIC4M Input Mux"},
+
+	{"MIC1 Analog Mux", "Line In", "MIC1P"},
+	{"MIC2 Analog Mux", "Line In", "MIC2P"},
+	{"MIC3 Analog Mux", "Line In", "MIC3P"},
+	{"MIC4 Analog Mux", "Line In", "MIC4P"},
+
+	{"MIC1P Input Mux", "Analog", "MIC1P"},
+	{"MIC1M Input Mux", "Analog", "MIC1M"},
+	{"MIC2P Input Mux", "Analog", "MIC2P"},
+	{"MIC2M Input Mux", "Analog", "MIC2M"},
+	{"MIC3P Input Mux", "Analog", "MIC3P"},
+	{"MIC3M Input Mux", "Analog", "MIC3M"},
+	{"MIC4P Input Mux", "Analog", "MIC4P"},
+	{"MIC4M Input Mux", "Analog", "MIC4M"},
+};
+
+static const struct snd_kcontrol_new adcx140_snd_controls[] = {
+	SOC_SINGLE_TLV("Analog CH1 Mic Gain Volume", ADCX140_CH1_CFG1, 2, 42, 0,
+			adc_tlv),
+	SOC_SINGLE_TLV("Analog CH2 Mic Gain Volume", ADCX140_CH1_CFG2, 2, 42, 0,
+			adc_tlv),
+	SOC_SINGLE_TLV("Analog CH3 Mic Gain Volume", ADCX140_CH1_CFG3, 2, 42, 0,
+			adc_tlv),
+	SOC_SINGLE_TLV("Analog CH4 Mic Gain Volume", ADCX140_CH1_CFG4, 2, 42, 0,
+			adc_tlv),
+
+	SOC_SINGLE_TLV("DRE Threshold", ADCX140_DRE_CFG0, 4, 9, 0,
+		       dre_thresh_tlv),
+	SOC_SINGLE_TLV("DRE Max Gain", ADCX140_DRE_CFG0, 0, 12, 0,
+		       dre_gain_tlv),
+
+	SOC_SINGLE_TLV("AGC Threshold", ADCX140_AGC_CFG0, 4, 15, 0,
+		       agc_thresh_tlv),
+	SOC_SINGLE_TLV("AGC Max Gain", ADCX140_AGC_CFG0, 0, 13, 0,
+		       agc_gain_tlv),
+
+	SOC_SINGLE_TLV("Digital CH1 Out Volume", ADCX140_CH1_CFG2,
+			0, 0xff, 0, dig_vol_tlv),
+	SOC_SINGLE_TLV("Digital CH2 Out Volume", ADCX140_CH2_CFG2,
+			0, 0xff, 0, dig_vol_tlv),
+	SOC_SINGLE_TLV("Digital CH3 Out Volume", ADCX140_CH3_CFG2,
+			0, 0xff, 0, dig_vol_tlv),
+	SOC_SINGLE_TLV("Digital CH4 Out Volume", ADCX140_CH4_CFG2,
+			0, 0xff, 0, dig_vol_tlv),
+	SOC_SINGLE_TLV("Digital CH5 Out Volume", ADCX140_CH5_CFG2,
+			0, 0xff, 0, dig_vol_tlv),
+	SOC_SINGLE_TLV("Digital CH6 Out Volume", ADCX140_CH6_CFG2,
+			0, 0xff, 0, dig_vol_tlv),
+	SOC_SINGLE_TLV("Digital CH7 Out Volume", ADCX140_CH7_CFG2,
+			0, 0xff, 0, dig_vol_tlv),
+	SOC_SINGLE_TLV("Digital CH8 Out Volume", ADCX140_CH8_CFG2,
+			0, 0xff, 0, dig_vol_tlv),
+};
+
+static int adcx140_reset(struct adcx140_priv *adcx140)
+{
+	int ret = 0;
+
+	if (adcx140->gpio_reset) {
+		gpiod_direction_output(adcx140->gpio_reset, 0);
+		/* 8.4.1: wait for hw shutdown (25ms) + >= 1ms */
+		usleep_range(30000, 100000);
+		gpiod_direction_output(adcx140->gpio_reset, 1);
+	} else {
+		ret = regmap_write(adcx140->regmap, ADCX140_SW_RESET,
+		          ADCX140_RESET);
+	}
+
+	/* 8.4.2: wait >= 10 ms after entering sleep mode. */
+	usleep_range(10000, 100000);
+
+	return 0;
+}
+
+static int adcx140_hw_params(struct snd_pcm_substream *substream,
+			     struct snd_pcm_hw_params *params,
+			     struct snd_soc_dai *dai)
+{
+	struct snd_soc_component *component = dai->component;
+	u8 data = 0;
+
+	switch (params_width(params)) {
+	case 16:
+		data = ADCX140_16_BIT_WORD;
+		break;
+	case 20:
+		data = ADCX140_20_BIT_WORD;
+		break;
+	case 24:
+		data = ADCX140_24_BIT_WORD;
+		break;
+	case 32:
+		data = ADCX140_32_BIT_WORD;
+		break;
+	default:
+		dev_err(component->dev, "%s: Unsupported width %d\n",
+			__func__, params_width(params));
+		return -EINVAL;
+	}
+
+	snd_soc_component_update_bits(component, ADCX140_ASI_CFG0,
+			    ADCX140_WORD_LEN_MSK, data);
+
+	return 0;
+}
+
+static int adcx140_set_dai_fmt(struct snd_soc_dai *codec_dai,
+			       unsigned int fmt)
+{
+	struct snd_soc_component *component = codec_dai->component;
+	struct adcx140_priv *adcx140 = snd_soc_component_get_drvdata(component);
+	u8 iface_reg1 = 0;
+	u8 iface_reg2 = 0;
+
+	/* set master/slave audio interface */
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		iface_reg2 |= ADCX140_BCLK_FSYNC_MASTER;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		break;
+	case SND_SOC_DAIFMT_CBS_CFM:
+	case SND_SOC_DAIFMT_CBM_CFS:
+	default:
+		dev_err(component->dev, "Invalid DAI master/slave interface\n");
+		return -EINVAL;
+	}
+
+	/* signal polarity */
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_IF:
+		iface_reg1 |= ADCX140_FSYNCINV_BIT;
+		break;
+	case SND_SOC_DAIFMT_IB_IF:
+		iface_reg1 |= ADCX140_BCLKINV_BIT | ADCX140_FSYNCINV_BIT;
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		iface_reg1 |= ADCX140_BCLKINV_BIT;
+		break;
+	case SND_SOC_DAIFMT_NB_NF:
+		break;
+	default:
+		dev_err(component->dev, "Invalid DAI clock signal polarity\n");
+		return -EINVAL;
+	}
+
+	/* interface format */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		iface_reg1 |= ADCX140_I2S_MODE_BIT;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		iface_reg1 |= ADCX140_LEFT_JUST_BIT;
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+	case SND_SOC_DAIFMT_DSP_B:
+		break;
+	default:
+		dev_err(component->dev, "Invalid DAI interface format\n");
+		return -EINVAL;
+	}
+
+	adcx140->dai_fmt = fmt & SND_SOC_DAIFMT_FORMAT_MASK;
+
+	snd_soc_component_update_bits(component, ADCX140_ASI_CFG0,
+				      ADCX140_FSYNCINV_BIT |
+				      ADCX140_BCLKINV_BIT |
+				      ADCX140_ASI_FORMAT_MSK,
+				      iface_reg1);
+	snd_soc_component_update_bits(component, ADCX140_MST_CFG0,
+				      ADCX140_BCLK_FSYNC_MASTER, iface_reg2);
+
+	return 0;
+}
+
+static int adcx140_set_dai_tdm_slot(struct snd_soc_dai *codec_dai,
+				  unsigned int tx_mask, unsigned int rx_mask,
+				  int slots, int slot_width)
+{
+	struct snd_soc_component *component = codec_dai->component;
+	struct adcx140_priv *adcx140 = snd_soc_component_get_drvdata(component);
+	unsigned int lsb;
+
+	if (tx_mask != rx_mask) {
+		dev_err(component->dev, "tx and rx masks must be symmetric\n");
+		return -EINVAL;
+	}
+
+	/* TDM based on DSP mode requires slots to be adjacent */
+	lsb = __ffs(tx_mask);
+	if ((lsb + 1) != __fls(tx_mask)) {
+		dev_err(component->dev, "Invalid mask, slots must be adjacent\n");
+		return -EINVAL;
+	}
+
+	switch (slot_width) {
+	case 16:
+	case 20:
+	case 24:
+	case 32:
+		break;
+	default:
+		dev_err(component->dev, "Unsupported slot width %d\n", slot_width);
+		return -EINVAL;
+	}
+
+	adcx140->tdm_delay = lsb;
+	adcx140->slot_width = slot_width;
+
+	return 0;
+}
+
+static int adcx140_prepare(struct snd_pcm_substream *substream,
+			 struct snd_soc_dai *dai)
+{
+	struct snd_soc_component *component = dai->component;
+	struct adcx140_priv *adcx140 = snd_soc_component_get_drvdata(component);
+	int offset = 0;
+	int width = adcx140->slot_width;
+
+	if (!width)
+		width = substream->runtime->sample_bits;
+
+	/* TDM slot selection only valid in DSP_A/_B mode */
+	if (adcx140->dai_fmt == SND_SOC_DAIFMT_DSP_A)
+		offset += (adcx140->tdm_delay * width + 1);
+	else if (adcx140->dai_fmt == SND_SOC_DAIFMT_DSP_B)
+		offset += adcx140->tdm_delay * width;
+
+	/* Configure data offset */
+	snd_soc_component_update_bits(component, ADCX140_ASI_CFG1,
+				      ADCX140_TX_OFFSET_MASK, offset);
+
+	return 0;
+}
+
+static const struct snd_soc_dai_ops adcx140_dai_ops = {
+	.hw_params	= adcx140_hw_params,
+	.set_fmt	= adcx140_set_dai_fmt,
+	.prepare	= adcx140_prepare,
+	.set_tdm_slot	= adcx140_set_dai_tdm_slot,
+};
+
+static int adcx140_codec_probe(struct snd_soc_component *component)
+{
+	struct adcx140_priv *adcx140 = snd_soc_component_get_drvdata(component);
+	int sleep_cfg_val = ADCX140_WAKE_DEV;
+	u8 bias_source;
+	u8 vref_source;
+	int ret;
+
+	ret = device_property_read_u8(adcx140->dev, "ti,mic-bias-source",
+				      &bias_source);
+	if (ret)
+		bias_source = ADCX140_MIC_BIAS_VAL_VREF;
+
+	if (bias_source < ADCX140_MIC_BIAS_VAL_VREF ||
+	    bias_source > ADCX140_MIC_BIAS_VAL_AVDD) {
+		dev_err(adcx140->dev, "Mic Bias source value is invalid\n");
+		return -EINVAL;
+	}
+
+	ret = device_property_read_u8(adcx140->dev, "ti,vref-source",
+				      &vref_source);
+	if (ret)
+		vref_source = ADCX140_MIC_BIAS_VREF_275V;
+
+	if (vref_source < ADCX140_MIC_BIAS_VREF_275V ||
+	    vref_source > ADCX140_MIC_BIAS_VREF_1375V) {
+		dev_err(adcx140->dev, "Mic Bias source value is invalid\n");
+		return -EINVAL;
+	}
+
+	bias_source |= vref_source;
+
+	ret = adcx140_reset(adcx140);
+	if (ret)
+		goto out;
+
+	if(adcx140->supply_areg == NULL)
+		sleep_cfg_val |= ADCX140_AREG_INTERNAL;
+
+	ret = regmap_write(adcx140->regmap, ADCX140_SLEEP_CFG, sleep_cfg_val);
+	if (ret) {
+		dev_err(adcx140->dev, "setting sleep config failed %d\n", ret);
+		goto out;
+	}
+
+	/* 8.4.3: Wait >= 1ms after entering active mode. */
+	usleep_range(1000, 100000);
+
+	ret = regmap_update_bits(adcx140->regmap, ADCX140_BIAS_CFG,
+				ADCX140_MIC_BIAS_VAL_MSK |
+				ADCX140_MIC_BIAS_VREF_MSK, bias_source);
+	if (ret)
+		dev_err(adcx140->dev, "setting MIC bias failed %d\n", ret);
+out:
+	return ret;
+}
+
+static int adcx140_set_bias_level(struct snd_soc_component *component,
+				  enum snd_soc_bias_level level)
+{
+	struct adcx140_priv *adcx140 = snd_soc_component_get_drvdata(component);
+	int pwr_cfg = 0;
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+	case SND_SOC_BIAS_PREPARE:
+	case SND_SOC_BIAS_STANDBY:
+		pwr_cfg = ADCX140_PWR_CFG_BIAS_PDZ | ADCX140_PWR_CFG_PLL_PDZ |
+			  ADCX140_PWR_CFG_ADC_PDZ;
+		break;
+	case SND_SOC_BIAS_OFF:
+		pwr_cfg = 0x0;
+		break;
+	}
+
+	return regmap_write(adcx140->regmap, ADCX140_PWR_CFG, pwr_cfg);
+}
+
+static const struct snd_soc_component_driver soc_codec_driver_adcx140 = {
+	.probe			= adcx140_codec_probe,
+	.set_bias_level		= adcx140_set_bias_level,
+	.controls		= adcx140_snd_controls,
+	.num_controls		= ARRAY_SIZE(adcx140_snd_controls),
+	.dapm_widgets		= adcx140_dapm_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(adcx140_dapm_widgets),
+	.dapm_routes		= adcx140_audio_map,
+	.num_dapm_routes	= ARRAY_SIZE(adcx140_audio_map),
+	.suspend_bias_off	= 1,
+	.idle_bias_on		= 0,
+	.use_pmdown_time	= 1,
+	.endianness		= 1,
+	.non_legacy_dai_naming	= 1,
+};
+
+static struct snd_soc_dai_driver adcx140_dai_driver[] = {
+	{
+		.name = "tlv320adcx140-codec",
+		.capture = {
+			.stream_name	 = "Capture",
+			.channels_min	 = 2,
+			.channels_max	 = ADCX140_MAX_CHANNELS,
+			.rates		 = ADCX140_RATES,
+			.formats	 = ADCX140_FORMATS,
+		},
+		.ops = &adcx140_dai_ops,
+		.symmetric_rates = 1,
+	}
+};
+
+static const struct of_device_id tlv320adcx140_of_match[] = {
+	{ .compatible = "ti,tlv320adc3140" },
+	{ .compatible = "ti,tlv320adc5140" },
+	{ .compatible = "ti,tlv320adc6140" },
+	{},
+};
+MODULE_DEVICE_TABLE(of, tlv320adcx140_of_match);
+
+static int adcx140_i2c_probe(struct i2c_client *i2c,
+			     const struct i2c_device_id *id)
+{
+	struct adcx140_priv *adcx140;
+	int ret;
+
+	adcx140 = devm_kzalloc(&i2c->dev, sizeof(*adcx140), GFP_KERNEL);
+	if (!adcx140)
+		return -ENOMEM;
+
+	adcx140->gpio_reset = devm_gpiod_get_optional(adcx140->dev,
+						      "reset", GPIOD_OUT_LOW);
+	if (IS_ERR(adcx140->gpio_reset))
+		dev_info(&i2c->dev, "Reset GPIO not defined\n");
+
+	adcx140->supply_areg = devm_regulator_get_optional(adcx140->dev,
+							   "areg");
+	if (IS_ERR(adcx140->supply_areg)) {
+		if (PTR_ERR(adcx140->supply_areg) == -EPROBE_DEFER)
+			return -EPROBE_DEFER;
+		else
+			adcx140->supply_areg = NULL;
+	} else {
+		ret = regulator_enable(adcx140->supply_areg);
+		if (ret) {
+			dev_err(adcx140->dev, "Failed to enable areg\n");
+			return ret;
+		}
+	}
+
+	adcx140->regmap = devm_regmap_init_i2c(i2c, &adcx140_i2c_regmap);
+	if (IS_ERR(adcx140->regmap)) {
+		ret = PTR_ERR(adcx140->regmap);
+		dev_err(&i2c->dev, "Failed to allocate register map: %d\n",
+			ret);
+		return ret;
+	}
+	adcx140->dev = &i2c->dev;
+	i2c_set_clientdata(i2c, adcx140);
+
+	return devm_snd_soc_register_component(&i2c->dev,
+					       &soc_codec_driver_adcx140,
+					       adcx140_dai_driver, 1);
+}
+
+static const struct i2c_device_id adcx140_i2c_id[] = {
+	{ "tlv320adc3140", 0 },
+	{ "tlv320adc5140", 1 },
+	{ "tlv320adc6140", 2 },
+	{}
+};
+MODULE_DEVICE_TABLE(i2c, adcx140_i2c_id);
+
+static struct i2c_driver adcx140_i2c_driver = {
+	.driver = {
+		.name	= "tlv320adcx140-codec",
+		.of_match_table = of_match_ptr(tlv320adcx140_of_match),
+	},
+	.probe		= adcx140_i2c_probe,
+	.id_table	= adcx140_i2c_id,
+};
+module_i2c_driver(adcx140_i2c_driver);
+
+MODULE_AUTHOR("Dan Murphy <[e-mail]>");
+MODULE_DESCRIPTION("ASoC TLV320ADCX140 CODEC Driver");
+MODULE_LICENSE("GPL v2");

sound/soc/codecs/tlv320adcx140.h

···
+// SPDX-License-Identifier: GPL-2.0
+// TLV320ADCX104 Sound driver
+// Copyright (C) 2020 Texas Instruments Incorporated - http://www.ti.com/
+
+#ifndef _TLV320ADCX140_H
+#define _TLV320ADCX140_H
+
+#define ADCX140_RATES	(SNDRV_PCM_RATE_44100 | \
+			 SNDRV_PCM_RATE_48000)
+
+#define ADCX140_FORMATS	(SNDRV_PCM_FMTBIT_S16_LE | \
+			 SNDRV_PCM_FMTBIT_S20_3LE | \
+			 SNDRV_PCM_FMTBIT_S24_3LE | \
+			 SNDRV_PCM_FMTBIT_S24_LE | \
+			 SNDRV_PCM_FMTBIT_S32_LE)
+
+#define ADCX140_PAGE_SELECT	0x00
+#define ADCX140_SW_RESET	0x01
+#define ADCX140_SLEEP_CFG 	0x02
+#define ADCX140_SHDN_CFG	0x05
+#define ADCX140_ASI_CFG0	0x07
+#define ADCX140_ASI_CFG1	0x08
+#define ADCX140_ASI_CFG2	0x09
+#define ADCX140_ASI_CH1		0x0b
+#define ADCX140_ASI_CH2		0x0c
+#define ADCX140_ASI_CH3		0x0d
+#define ADCX140_ASI_CH4		0x0e
+#define ADCX140_ASI_CH5		0x0f
+#define ADCX140_ASI_CH6		0x10
+#define ADCX140_ASI_CH7		0x11
+#define ADCX140_ASI_CH8		0x12
+#define ADCX140_MST_CFG0	0x13
+#define ADCX140_MST_CFG1	0x14
+#define ADCX140_ASI_STS		0x15
+#define ADCX140_CLK_SRC		0x16
+#define ADCX140_PDMCLK_CFG	0x1f
+#define ADCX140_PDM_CFG		0x20
+#define ADCX140_GPIO_CFG0	0x21
+#define ADCX140_GPO_CFG1	0x22
+#define ADCX140_GPO_CFG2	0x23
+#define ADCX140_GPO_CFG3	0x24
+#define ADCX140_GPO_CFG4	0x25
+#define ADCX140_GPO_VAL		0x29
+#define ADCX140_GPIO_MON	0x2a
+#define ADCX140_GPI_CFG0	0x2b
+#define ADCX140_GPI_CFG1	0x2c
+#define ADCX140_GPI_MON		0x2f
+#define ADCX140_INT_CFG		0x32
+#define ADCX140_INT_MASK0	0x33
+#define ADCX140_INT_LTCH0	0x36
+#define ADCX140_BIAS_CFG	0x3b
+#define ADCX140_CH1_CFG0	0x3c
+#define ADCX140_CH1_CFG1	0x3d
+#define ADCX140_CH1_CFG2	0x3e
+#define ADCX140_CH1_CFG3	0x3f
+#define ADCX140_CH1_CFG4	0x40
+#define ADCX140_CH2_CFG0	0x41
+#define ADCX140_CH2_CFG1	0x42
+#define ADCX140_CH2_CFG2	0x43
+#define ADCX140_CH2_CFG3	0x44
+#define ADCX140_CH2_CFG4	0x45
+#define ADCX140_CH3_CFG0	0x46
+#define ADCX140_CH3_CFG1	0x47
+#define ADCX140_CH3_CFG2	0x48
+#define ADCX140_CH3_CFG3	0x49
+#define ADCX140_CH3_CFG4 	0x4a
+#define ADCX140_CH4_CFG0	0x4b
+#define ADCX140_CH4_CFG1	0x4c
+#define ADCX140_CH4_CFG2	0x4d
+#define ADCX140_CH4_CFG3	0x4e
+#define ADCX140_CH4_CFG4	0x4f
+#define ADCX140_CH5_CFG2	0x52
+#define ADCX140_CH5_CFG3	0x53
+#define ADCX140_CH5_CFG4	0x54
+#define ADCX140_CH6_CFG2	0x57
+#define ADCX140_CH6_CFG3	0x58
+#define ADCX140_CH6_CFG4	0x59
+#define ADCX140_CH7_CFG2	0x5c
+#define ADCX140_CH7_CFG3	0x5d
+#define ADCX140_CH7_CFG4	0x5e
+#define ADCX140_CH8_CFG2	0x61
+#define ADCX140_CH8_CFG3	0x62
+#define ADCX140_CH8_CFG4	0x63
+#define ADCX140_DSP_CFG0	0x6b
+#define ADCX140_DSP_CFG1	0x6c
+#define ADCX140_DRE_CFG0	0x6d
+#define ADCX140_AGC_CFG0	0x70
+#define ADCX140_IN_CH_EN	0x73
+#define ADCX140_ASI_OUT_CH_EN	0x74
+#define ADCX140_PWR_CFG		0x75
+#define ADCX140_DEV_STS0	0x76
+#define ADCX140_DEV_STS1	0x77
+
+#define ADCX140_RESET	BIT(0)
+
+#define ADCX140_WAKE_DEV	BIT(0)
+#define ADCX140_AREG_INTERNAL	BIT(7)
+
+#define ADCX140_BCLKINV_BIT	BIT(2)
+#define ADCX140_FSYNCINV_BIT	BIT(3)
+#define ADCX140_INV_MSK		(ADCX140_BCLKINV_BIT | ADCX140_FSYNCINV_BIT)
+#define ADCX140_BCLK_FSYNC_MASTER	BIT(7)
+#define ADCX140_I2S_MODE_BIT	BIT(6)
+#define ADCX140_LEFT_JUST_BIT	BIT(7)
+#define ADCX140_ASI_FORMAT_MSK	(ADCX140_I2S_MODE_BIT | ADCX140_LEFT_JUST_BIT)
+
+#define ADCX140_16_BIT_WORD	0x0
+#define ADCX140_20_BIT_WORD	BIT(4)
+#define ADCX140_24_BIT_WORD	BIT(5)
+#define ADCX140_32_BIT_WORD	(BIT(4) | BIT(5))
+#define ADCX140_WORD_LEN_MSK	0x30
+
+#define ADCX140_MAX_CHANNELS	8
+
+#define ADCX140_MIC_BIAS_VAL_VREF	0
+#define ADCX140_MIC_BIAS_VAL_VREF_1096	1
+#define ADCX140_MIC_BIAS_VAL_AVDD	6
+#define ADCX140_MIC_BIAS_VAL_MSK GENMASK(6, 4)
+
+#define ADCX140_MIC_BIAS_VREF_275V	0
+#define ADCX140_MIC_BIAS_VREF_25V	1
+#define ADCX140_MIC_BIAS_VREF_1375V	2
+#define ADCX140_MIC_BIAS_VREF_MSK GENMASK(1, 0)
+
+#define ADCX140_PWR_CFG_BIAS_PDZ	BIT(7)
+#define ADCX140_PWR_CFG_ADC_PDZ		BIT(6)
+#define ADCX140_PWR_CFG_PLL_PDZ		BIT(5)
+
+#define ADCX140_TX_OFFSET_MASK		GENMASK(4, 0)
+
+#endif /* _TLV320ADCX140_ */

sound/soc/codecs/wcd9335.c

···
 		.name = "WCD9335",
 		.range_min =  0x0,
 		.range_max =  WCD9335_MAX_REGISTER,
-		.selector_reg = WCD9335_REG(0x0, 0),
+		.selector_reg = WCD9335_SEL_REGISTER,
 		.selector_mask = 0xff,
 		.selector_shift = 0,
-		.window_start = 0x0,
-		.window_len = 0x1000,
+		.window_start = 0x800,
+		.window_len = 0x100,
 	},
 };
 
···
 	{
 		.name = "WCD9335-IFC-DEV",
 		.range_min =  0x0,
-		.range_max = WCD9335_REG(0, 0x7ff),
-		.selector_reg = WCD9335_REG(0, 0x0),
-		.selector_mask = 0xff,
+		.range_max = WCD9335_MAX_REGISTER,
+		.selector_reg = WCD9335_SEL_REGISTER,
+		.selector_mask = 0xfff,
 		.selector_shift = 0,
-		.window_start = 0x0,
-		.window_len = 0x1000,
+		.window_start = 0x800,
+		.window_len = 0x400,
 	},
 };
 
···
 	.reg_bits = 16,
 	.val_bits = 8,
 	.can_multi_write = true,
-	.max_register = WCD9335_REG(0, 0x7FF),
+	.max_register = WCD9335_MAX_REGISTER,
 	.ranges = wcd9335_ifc_ranges,
 	.num_ranges = ARRAY_SIZE(wcd9335_ifc_ranges),
 };

sound/soc/codecs/wcd9335.h

···
  * in slimbus mode the reg base starts from 0x800
  * in i2s/i2c mode the reg base is 0x0
  */
-#define WCD9335_REG(pg, r)	((pg << 12) | (r) | 0x800)
+#define WCD9335_REG(pg, r)	((pg << 8) | (r))
 #define WCD9335_REG_OFFSET(r)	(r & 0xFF)
-#define WCD9335_PAGE_OFFSET(r)	((r >> 12) & 0xFF)
+#define WCD9335_PAGE_OFFSET(r)	((r >> 8) & 0xFF)
 
 /* Page-0 Registers */
 #define WCD9335_PAGE0_PAGE_REGISTER		WCD9335_REG(0x00, 0x000)
···
 #define WCD9335_CDC_CLK_RST_CTRL_FS_CNT_ENABLE	BIT(0)
 #define WCD9335_CDC_CLK_RST_CTRL_FS_CNT_DISABLE	0
 #define WCD9335_CDC_TOP_TOP_CFG1	WCD9335_REG(0x0d, 0x082)
-#define WCD9335_MAX_REGISTER	WCD9335_REG(0x80, 0x0FF)
+#define WCD9335_MAX_REGISTER	0xffff
+#define WCD9335_SEL_REGISTER	0x800
 
 /* SLIMBUS Slave Registers */
 #define WCD9335_SLIM_PGD_PORT_INT_EN0	WCD9335_REG(0, 0x30)

sound/soc/codecs/wcd934x.c

···
 
 #include <linux/clk.h>
 #include <linux/clk-provider.h>
-#include <linux/gpio.h>
 #include <linux/interrupt.h>
 #include <linux/kernel.h>
 #include <linux/mfd/wcd934x/registers.h>
···
 #include <linux/module.h>
 #include <linux/mutex.h>
 #include <linux/of_clk.h>
-#include <linux/of_device.h>
-#include <linux/of_gpio.h>
 #include <linux/of.h>
-#include <linux/of_irq.h>
 #include <linux/platform_device.h>
 #include <linux/regmap.h>
 #include <linux/regulator/consumer.h>
···
 		regmap_update_bits(wcd->regmap, WCD934X_ANA_RCO,
 				   WCD934X_ANA_RCO_BG_EN_MASK, 0);
 		usleep_range(100, 110);
-	} else if (sido_src == SIDO_SOURCE_RCO_BG) {
-		regmap_update_bits(wcd->regmap, WCD934X_ANA_RCO,
-				   WCD934X_ANA_RCO_BG_EN_MASK,
-				   WCD934X_ANA_RCO_BG_ENABLE);
-		usleep_range(100, 110);
 		regmap_update_bits(wcd->regmap, WCD934X_ANA_BUCK_CTL,
 				   WCD934X_ANA_BUCK_PRE_EN1_MASK,
 				   WCD934X_ANA_BUCK_PRE_EN1_ENABLE);
···
 		regmap_update_bits(wcd->regmap, WCD934X_ANA_BUCK_CTL,
 				   WCD934X_ANA_BUCK_HI_ACCU_EN_MASK,
 				   WCD934X_ANA_BUCK_HI_ACCU_ENABLE);
+		usleep_range(100, 110);
+	} else if (sido_src == SIDO_SOURCE_RCO_BG) {
+		regmap_update_bits(wcd->regmap, WCD934X_ANA_RCO,
+				   WCD934X_ANA_RCO_BG_EN_MASK,
+				   WCD934X_ANA_RCO_BG_ENABLE);
 		usleep_range(100, 110);
 	}
 	wcd->sido_input_src = sido_src;
···
 		return -EINVAL;
 	}
 
-	if (wcd->rx_chs) {
-		wcd->num_rx_port = rx_num;
-		for (i = 0; i < rx_num; i++) {
-			wcd->rx_chs[i].ch_num = rx_slot[i];
-			INIT_LIST_HEAD(&wcd->rx_chs[i].list);
-		}
+	wcd->num_rx_port = rx_num;
+	for (i = 0; i < rx_num; i++) {
+		wcd->rx_chs[i].ch_num = rx_slot[i];
+		INIT_LIST_HEAD(&wcd->rx_chs[i].list);
 	}
 
-	if (wcd->tx_chs) {
-		wcd->num_tx_port = tx_num;
-		for (i = 0; i < tx_num; i++) {
-			wcd->tx_chs[i].ch_num = tx_slot[i];
-			INIT_LIST_HEAD(&wcd->tx_chs[i].list);
-		}
+	wcd->num_tx_port = tx_num;
+	for (i = 0; i < tx_num; i++) {
+		wcd->tx_chs[i].ch_num = tx_slot[i];
+		INIT_LIST_HEAD(&wcd->tx_chs[i].list);
 	}
 
 	return 0;
···
 {
 	u8 hph_dly_mask;
 	u16 hph_lut_bypass_reg = 0;
-	u16 hph_comp_ctrl7 = 0;
 
 	switch (interp_idx) {
 	case INTERP_HPHL:
 		hph_dly_mask = 1;
 		hph_lut_bypass_reg = WCD934X_CDC_TOP_HPHL_COMP_LUT;
-		hph_comp_ctrl7 = WCD934X_CDC_COMPANDER1_CTL7;
 		break;
 	case INTERP_HPHR:
 		hph_dly_mask = 2;
 		hph_lut_bypass_reg = WCD934X_CDC_TOP_HPHR_COMP_LUT;
-		hph_comp_ctrl7 = WCD934X_CDC_COMPANDER2_CTL7;
 		break;
 	default:
 		return;

sound/soc/codecs/wm0010.c

···
 	u8 command;
 	u32 length:24;
 	u32 address;
-	uint8_t data[0];
+	uint8_t data[];
 } __packed;
 
 struct dfw_inforec {

sound/soc/codecs/wm5110.c

···
 	struct arizona *arizona = priv->core.arizona;
 	int n_adsp;
 
-	if (strcmp(rtd->codec_dai->name, "wm5110-dsp-voicectrl") == 0) {
+	if (strcmp(asoc_rtd_to_codec(rtd, 0)->name, "wm5110-dsp-voicectrl") == 0) {
 		n_adsp = 2;
-	} else if (strcmp(rtd->codec_dai->name, "wm5110-dsp-trace") == 0) {
+	} else if (strcmp(asoc_rtd_to_codec(rtd, 0)->name, "wm5110-dsp-trace") == 0) {
 		n_adsp = 0;
 	} else {
 		dev_err(arizona->dev,
 			"No suitable compressed stream for DAI '%s'\n",
-			rtd->codec_dai->name);
+			asoc_rtd_to_codec(rtd, 0)->name);
 		return -EINVAL;
 	}
 

sound/soc/codecs/wm8974.c

···
 SOC_DAPM_SINGLE("MicP Switch", WM8974_INPUT, 0, 1, 0),
 };
 
-/* AUX Input boost vol */
-static const struct snd_kcontrol_new wm8974_aux_boost_controls =
-SOC_DAPM_SINGLE("Aux Volume", WM8974_ADCBOOST, 0, 7, 0);
-
-/* Mic Input boost vol */
-static const struct snd_kcontrol_new wm8974_mic_boost_controls =
-SOC_DAPM_SINGLE("Mic Volume", WM8974_ADCBOOST, 4, 7, 0);
-
 static const struct snd_soc_dapm_widget wm8974_dapm_widgets[] = {
 SND_SOC_DAPM_MIXER("Speaker Mixer", WM8974_POWER3, 2, 0,
 	&wm8974_speaker_mixer_controls[0],

sound/soc/codecs/wm_adsp.c

···
 		subname = NULL; /* don't append subname */
 		break;
 	case 2:
-		ret = snprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN,
+		ret = scnprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN,
 				"%s%c %.12s %x", dsp->name, *region_name,
 				wm_adsp_fw_text[dsp->fw], alg_region->alg);
 		break;
 	default:
-		ret = snprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN,
+		ret = scnprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN,
 				"%s %.12s %x", dsp->name,
 				wm_adsp_fw_text[dsp->fw], alg_region->alg);
 		break;
···
 
 	if (wm_adsp_fw[dsp->fw].num_caps == 0) {
 		adsp_err(dsp, "%s: Firmware does not support compressed API\n",
-			 rtd->codec_dai->name);
+			 asoc_rtd_to_codec(rtd, 0)->name);
 		ret = -ENXIO;
 		goto out;
 	}
 
 	if (wm_adsp_fw[dsp->fw].compr_direction != stream->direction) {
 		adsp_err(dsp, "%s: Firmware does not support stream direction\n",
-			 rtd->codec_dai->name);
+			 asoc_rtd_to_codec(rtd, 0)->name);
 		ret = -EINVAL;
 		goto out;
 	}
 
 	list_for_each_entry(tmp, &dsp->compr_list, list) {
-		if (!strcmp(tmp->name, rtd->codec_dai->name)) {
+		if (!strcmp(tmp->name, asoc_rtd_to_codec(rtd, 0)->name)) {
 			adsp_err(dsp, "%s: Only a single stream supported per dai\n",
-				 rtd->codec_dai->name);
+				 asoc_rtd_to_codec(rtd, 0)->name);
 			ret = -EBUSY;
 			goto out;
 		}
···
 
 	compr->dsp = dsp;
 	compr->stream = stream;
-	compr->name = rtd->codec_dai->name;
+	compr->name = asoc_rtd_to_codec(rtd, 0)->name;
 
 	list_add_tail(&compr->list, &dsp->compr_list);
 

sound/soc/codecs/wsa881x.c

···
 	int active_ports;
 	bool port_prepared[WSA881X_MAX_SWR_PORTS];
 	bool port_enable[WSA881X_MAX_SWR_PORTS];
-	bool stream_prepared;
 };
 
 static void wsa881x_init(struct wsa881x_priv *wsa881x)
···
 	SND_SOC_DAPM_OUTPUT("SPKR"),
 };
 
-static int wsa881x_prepare(struct snd_pcm_substream *substream,
-			   struct snd_soc_dai *dai)
-{
-	struct wsa881x_priv *wsa881x = dev_get_drvdata(dai->dev);
-	int ret;
-
-	if (wsa881x->stream_prepared) {
-		sdw_disable_stream(wsa881x->sruntime);
-		sdw_deprepare_stream(wsa881x->sruntime);
-		wsa881x->stream_prepared = false;
-	}
-
-
-	ret = sdw_prepare_stream(wsa881x->sruntime);
-	if (ret)
-		return ret;
-
-	/**
-	 * NOTE: there is a strict hw requirement about the ordering of port
-	 * enables and actual PA enable. PA enable should only happen after
-	 * soundwire ports are enabled if not DC on the line is accumulated
-	 * resulting in Click/Pop Noise
-	 * PA enable/mute are handled as part of DAPM and digital mute.
-	 */
-
-	ret = sdw_enable_stream(wsa881x->sruntime);
-	if (ret) {
-		sdw_deprepare_stream(wsa881x->sruntime);
-		return ret;
-	}
-	wsa881x->stream_prepared = true;
-
-	return ret;
-}
-
 static int wsa881x_hw_params(struct snd_pcm_substream *substream,
 			     struct snd_pcm_hw_params *params,
 			     struct snd_soc_dai *dai)
···
 {
 	struct wsa881x_priv *wsa881x = dev_get_drvdata(dai->dev);
 
-	if (wsa881x->stream_prepared) {
-		sdw_disable_stream(wsa881x->sruntime);
-		sdw_deprepare_stream(wsa881x->sruntime);
-		sdw_stream_remove_slave(wsa881x->slave, wsa881x->sruntime);
-		wsa881x->stream_prepared = false;
-	}
+	sdw_stream_remove_slave(wsa881x->slave, wsa881x->sruntime);
 
 	return 0;
 }
···
 
 static struct snd_soc_dai_ops wsa881x_dai_ops = {
 	.hw_params = wsa881x_hw_params,
-	.prepare = wsa881x_prepare,
 	.hw_free = wsa881x_hw_free,
 	.mute_stream = wsa881x_digital_mute,
 	.set_sdw_stream = wsa881x_set_sdw_stream,
···
 	wsa881x->sconfig.type = SDW_STREAM_PDM;
 	pdev->prop.sink_ports = GENMASK(WSA881X_MAX_SWR_PORTS, 0);
 	pdev->prop.sink_dpn_prop = wsa_sink_dpn_prop;
-	gpiod_set_value(wsa881x->sd_n, 1);
+	gpiod_direction_output(wsa881x->sd_n, 1);
 
 	wsa881x->regmap = devm_regmap_init_sdw(pdev, &wsa881x_regmap_config);
 	if (IS_ERR(wsa881x->regmap)) {

sound/soc/dwc/dwc-i2s.c

···
 {
 	struct dw_i2s_dev *dev = snd_soc_component_get_drvdata(component);
 	struct snd_soc_dai *dai;
+	int stream;
 
 	if (dev->capability & DW_I2S_MASTER)
 		clk_enable(dev->clk);
 
 	for_each_component_dais(component, dai) {
-		if (dai->playback_active)
-			dw_i2s_config(dev, SNDRV_PCM_STREAM_PLAYBACK);
-		if (dai->capture_active)
-			dw_i2s_config(dev, SNDRV_PCM_STREAM_CAPTURE);
+		for_each_pcm_streams(stream)
+			if (dai->stream_active[stream])
+				dw_i2s_config(dev, stream);
 	}
 
 	return 0;

sound/soc/dwc/dwc-pcm.c

···
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+	struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
 
 	snd_soc_set_runtime_hwparams(substream, &dw_pcm_hardware);
 	snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS);

sound/soc/fsl/eukrea-tlv320.c

···
 			    struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	int ret;
 
 	ret = snd_soc_dai_set_sysclk(codec_dai, 0,

sound/soc/fsl/fsl-asoc-card.c

···
 		return 0;
 
 	/* Specific configurations of DAIs starts from here */
-	ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, cpu_priv->sysclk_id[tx],
+	ret = snd_soc_dai_set_sysclk(asoc_rtd_to_cpu(rtd, 0), cpu_priv->sysclk_id[tx],
 				     cpu_priv->sysclk_freq[tx],
 				     cpu_priv->sysclk_dir[tx]);
 	if (ret && ret != -ENOTSUPP) {
···
 	}
 
 	if (cpu_priv->slot_width) {
-		ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2,
+		ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_cpu(rtd, 0), 0x3, 0x3, 2,
 					       cpu_priv->slot_width);
 		if (ret && ret != -ENOTSUPP) {
 			dev_err(dev, "failed to set TDM slot for cpu dai\n");
···
 	int ret;
 
 	rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[0]);
-	codec_dai = rtd->codec_dai;
+	codec_dai = asoc_rtd_to_codec(rtd, 0);
 	if (dapm->dev != codec_dai->dev)
 		return 0;
 
···
 	struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card);
 	struct snd_soc_pcm_runtime *rtd = list_first_entry(
 			&card->rtd_list, struct snd_soc_pcm_runtime, list);
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	struct codec_priv *codec_priv = &priv->codec_priv;
 	struct device *dev = card->dev;
 	int ret;
 
 	if (fsl_asoc_card_is_ac97(priv)) {
 #if IS_ENABLED(CONFIG_SND_AC97_CODEC)
-		struct snd_soc_component *component = rtd->codec_dai->component;
+		struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
 		struct snd_ac97 *ac97 = snd_soc_component_get_drvdata(component);
 
 		/*

sound/soc/fsl/fsl_asrc_dma.c

···
 	for_each_dpcm_be(rtd, stream, dpcm) {
 		struct snd_soc_pcm_runtime *be = dpcm->be;
 		struct snd_pcm_substream *substream_be;
-		struct snd_soc_dai *dai = be->cpu_dai;
+		struct snd_soc_dai *dai = asoc_rtd_to_cpu(be, 0);
 
 		if (dpcm->fe != rtd)
 			continue;
···
 	}
 
 	/* Override dma_data of the Front-End and config its dmaengine */
-	dma_params_fe = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+	dma_params_fe = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream);
 	dma_params_fe->addr = asrc_priv->paddr + REG_ASRDx(!dir, index);
 	dma_params_fe->maxburst = dma_params_be->maxburst;
 
···
 		goto dma_chan_err;
 	}
 
-	dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+	dma_data = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream);
 
 	/* Refine the snd_imx_hardware according to caps of DMA. */
 	ret = snd_dmaengine_pcm_refine_runtime_hwparams(substream,
···
 		return ret;
 	}
 
-	for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_LAST; i++) {
+	for_each_pcm_streams(i) {
 		substream = pcm->streams[i].substream;
 		if (!substream)
 			continue;
···
 	struct snd_pcm_substream *substream;
 	int i;
 
-	for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_LAST; i++) {
+	for_each_pcm_streams(i) {
 		substream = pcm->streams[i].substream;
 		if (!substream)
 			continue;

sound/soc/fsl/fsl_spdif.c

···
 				int sample_rate)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+	struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
 	struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control;
 	struct regmap *regmap = spdif_priv->regmap;
 	struct platform_device *pdev = spdif_priv->pdev;
···
 			     struct snd_soc_dai *cpu_dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+	struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
 	struct platform_device *pdev = spdif_priv->pdev;
 	struct regmap *regmap = spdif_priv->regmap;
 	u32 scr, mask;
···
 				struct snd_soc_dai *cpu_dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+	struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
 	struct regmap *regmap = spdif_priv->regmap;
 	u32 scr, mask, i;
 
···
 				struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+	struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
 	struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control;
 	struct platform_device *pdev = spdif_priv->pdev;
 	u32 sample_rate = params_rate(params);
···
 				int cmd, struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+	struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
 	struct regmap *regmap = spdif_priv->regmap;
 	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
 	u32 intr = SIE_INTR_FOR(tx);

sound/soc/fsl/fsl_ssi.c

···
 			   struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct fsl_ssi *ssi = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+	struct fsl_ssi *ssi = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
 	int ret;
 
 	ret = clk_prepare_enable(ssi->clk);
···
 			     struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct fsl_ssi *ssi = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+	struct fsl_ssi *ssi = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
 
 	clk_disable_unprepare(ssi->clk);
 }
···
 			   struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct fsl_ssi *ssi = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+	struct fsl_ssi *ssi = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
 
 	if (fsl_ssi_is_i2s_master(ssi) &&
 	    ssi->baudclk_streams & BIT(substream->stream)) {
···
 			   struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct fsl_ssi *ssi = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+	struct fsl_ssi *ssi = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
 	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
 
 	switch (cmd) {

sound/soc/fsl/imx-audmix.c

···
 	dir  = tx ? SND_SOC_CLOCK_OUT : SND_SOC_CLOCK_IN;
 
 	/* set DAI configuration */
-	ret = snd_soc_dai_set_fmt(rtd->cpu_dai, fmt);
+	ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), fmt);
 	if (ret) {
 		dev_err(dev, "failed to set cpu dai fmt: %d\n", ret);
 		return ret;
 	}
 
-	ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, FSL_SAI_CLK_MAST1, 0, dir);
+	ret = snd_soc_dai_set_sysclk(asoc_rtd_to_cpu(rtd, 0), FSL_SAI_CLK_MAST1, 0, dir);
 	if (ret) {
 		dev_err(dev, "failed to set cpu sysclk: %d\n", ret);
 		return ret;
···
 	 * Per datasheet, AUDMIX expects 8 slots and 32 bits
 	 * for every slot in TDM mode.
 	 */
-	ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, BIT(channels) - 1,
+	ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_cpu(rtd, 0), BIT(channels) - 1,
 				       BIT(channels) - 1, 8, 32);
 	if (ret)
 		dev_err(dev, "failed to set cpu dai tdm slot: %d\n", ret);
···
 	fmt |= SND_SOC_DAIFMT_CBM_CFM;
 
 	/* set AUDMIX DAI configuration */
-	ret = snd_soc_dai_set_fmt(rtd->cpu_dai, fmt);
+	ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), fmt);
 	if (ret)
 		dev_err(dev, "failed to set AUDMIX DAI fmt: %d\n", ret);
 

sound/soc/fsl/imx-mc13783.c

···
 	struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	int ret;
 
 	ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x3, 0x3, 4, 16);

sound/soc/fsl/imx-sgtl5000.c

···
 	struct device *dev = rtd->card->dev;
 	int ret;
 
-	ret = snd_soc_dai_set_sysclk(rtd->codec_dai, SGTL5000_SYSCLK,
+	ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(rtd, 0), SGTL5000_SYSCLK,
 				     data->clk_frequency, SND_SOC_CLOCK_IN);
 	if (ret) {
 		dev_err(dev, "could not set codec driver clock params\n");

sound/soc/fsl/mpc5200_dma.c

···
 			   struct snd_pcm_substream *substream, int cmd)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+	struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct psc_dma_stream *s = to_psc_dma_stream(substream, psc_dma);
 	struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs;
···
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+	struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
 	struct psc_dma_stream *s;
 	int rc;
 
···
 			 struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+	struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
 	struct psc_dma_stream *s;
 
 	dev_dbg(psc_dma->dev, "psc_dma_close(substream=%p)\n", substream);
···
 		struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+	struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
 	struct psc_dma_stream *s;
 	dma_addr_t count;
 
···
 		       struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_card *card = rtd->card->snd_card;
-	struct snd_soc_dai *dai = rtd->cpu_dai;
+	struct snd_soc_dai *dai = asoc_rtd_to_cpu(rtd, 0);
 	struct snd_pcm *pcm = rtd->pcm;
 	size_t size = psc_dma_hardware.buffer_bytes_max;
 	int rc;

sound/soc/fsl/mpc5200_psc_i2s.c

···
 				 struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+	struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
 	u32 mode;
 
 	dev_dbg(psc_dma->dev, "%s(substream=%p) p_size=%i p_bytes=%i"

sound/soc/fsl/mpc8610_hpcd.c

···
 	int ret = 0;
 
 	/* Tell the codec driver what the serial protocol is. */
-	ret = snd_soc_dai_set_fmt(rtd->codec_dai, machine_data->dai_format);
+	ret = snd_soc_dai_set_fmt(asoc_rtd_to_codec(rtd, 0), machine_data->dai_format);
 	if (ret < 0) {
 		dev_err(dev, "could not set codec driver audio format\n");
 		return ret;
···
 	 * Tell the codec driver what the MCLK frequency is, and whether it's
 	 * a slave or master.
 	 */
-	ret = snd_soc_dai_set_sysclk(rtd->codec_dai, 0,
+	ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(rtd, 0), 0,
 				     machine_data->clk_frequency,
 				     machine_data->codec_clk_direction);
 	if (ret < 0) {

sound/soc/fsl/mx27vis-aic32x4.c

···
 			    struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	int ret;
 
 	ret = snd_soc_dai_set_sysclk(codec_dai, 0,

sound/soc/fsl/p1022_ds.c

···
 	int ret = 0;
 
 	/* Tell the codec driver what the serial protocol is. */
-	ret = snd_soc_dai_set_fmt(rtd->codec_dai, mdata->dai_format);
+	ret = snd_soc_dai_set_fmt(asoc_rtd_to_codec(rtd, 0), mdata->dai_format);
 	if (ret < 0) {
 		dev_err(dev, "could not set codec driver audio format\n");
 		return ret;
···
 	 * Tell the codec driver what the MCLK frequency is, and whether it's
 	 * a slave or master.
 	 */
-	ret = snd_soc_dai_set_sysclk(rtd->codec_dai, 0, mdata->clk_frequency,
+	ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(rtd, 0), 0, mdata->clk_frequency,
 				     mdata->codec_clk_direction);
 	if (ret < 0) {
 		dev_err(dev, "could not set codec driver clock params\n");

sound/soc/fsl/p1022_rdk.c

···
 	int ret = 0;
 
 	/* Tell the codec driver what the serial protocol is. */
-	ret = snd_soc_dai_set_fmt(rtd->codec_dai, mdata->dai_format);
+	ret = snd_soc_dai_set_fmt(asoc_rtd_to_codec(rtd, 0), mdata->dai_format);
 	if (ret < 0) {
 		dev_err(dev, "could not set codec driver audio format (ret=%i)\n",
 			ret);
 		return ret;
 	}
 
-	ret = snd_soc_dai_set_pll(rtd->codec_dai, 0, 0, mdata->clk_frequency,
+	ret = snd_soc_dai_set_pll(asoc_rtd_to_codec(rtd, 0), 0, 0, mdata->clk_frequency,
 		mdata->clk_frequency);
 	if (ret < 0) {
 		dev_err(dev, "could not set codec PLL frequency (ret=%i)\n",

sound/soc/fsl/wm1133-ev1.c

···
 				struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	int i, found = 0;
 	snd_pcm_format_t format = params_format(params);
 	unsigned int rate = params_rate(params);
···
 
 static int wm1133_ev1_init(struct snd_soc_pcm_runtime *rtd)
 {
-	struct snd_soc_component *component = rtd->codec_dai->component;
+	struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
 
 	/* Headphone jack detection */
 	snd_soc_card_jack_new(rtd->card, "Headphone", SND_JACK_HEADPHONE,

sound/soc/generic/simple-card-utils.c

···
 void asoc_simple_shutdown(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card);
 	struct simple_dai_props *dai_props =
 		simple_priv_to_props(priv, rtd->num);
···
 			  struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card);
 	struct simple_dai_props *dai_props =
 		simple_priv_to_props(priv, rtd->num);
···
 	return 0;
 }
 
+static int asoc_simple_init_dai_link_params(struct snd_soc_pcm_runtime *rtd,
+					    struct simple_dai_props *dai_props)
+{
+	struct snd_soc_dai_link *dai_link = rtd->dai_link;
+	struct snd_soc_component *component;
+	struct snd_soc_pcm_stream *params;
+	struct snd_pcm_hardware hw;
+	int i, ret, stream;
+
+	/* Only codecs should have non_legacy_dai_naming set. */
+	for_each_rtd_components(rtd, i, component) {
+		if (!component->driver->non_legacy_dai_naming)
+			return 0;
+	}
+
+	/* Assumes the capabilities are the same for all supported streams */
+	for_each_pcm_streams(stream) {
+		ret = snd_soc_runtime_calc_hw(rtd, &hw, stream);
+		if (ret == 0)
+			break;
+	}
+
+	if (ret < 0) {
+		dev_err(rtd->dev, "simple-card: no valid dai_link params\n");
+		return ret;
+	}
+
+	params = devm_kzalloc(rtd->dev, sizeof(*params), GFP_KERNEL);
+	if (!params)
+		return -ENOMEM;
+
+	params->formats = hw.formats;
+	params->rates = hw.rates;
+	params->rate_min = hw.rate_min;
+	params->rate_max = hw.rate_max;
+	params->channels_min = hw.channels_min;
+	params->channels_max = hw.channels_max;
+
+	dai_link->params = params;
+	dai_link->num_params = 1;
+
+	return 0;
+}
+
 int asoc_simple_dai_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card);
 	struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num);
 	int ret;
 
-	ret = asoc_simple_init_dai(rtd->codec_dai,
+	ret = asoc_simple_init_dai(asoc_rtd_to_codec(rtd, 0),
 				   dai_props->codec_dai);
 	if (ret < 0)
 		return ret;
 
-	ret = asoc_simple_init_dai(rtd->cpu_dai,
+	ret = asoc_simple_init_dai(asoc_rtd_to_cpu(rtd, 0),
 				   dai_props->cpu_dai);
+	if (ret < 0)
+		return ret;
+
+	ret = asoc_simple_init_dai_link_params(rtd, dai_props);
 	if (ret < 0)
 		return ret;
 

sound/soc/img/img-i2s-in.c

···
 	struct snd_dmaengine_dai_dma_data *dma_data;
 	int ret;
 
-	dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, st);
+	dma_data = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), st);
 
 	ret = snd_hwparams_to_dma_slave_config(st, params, sc);
 	if (ret)

sound/soc/img/img-i2s-out.c

···
 	struct snd_dmaengine_dai_dma_data *dma_data;
 	int ret;
 
-	dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, st);
+	dma_data = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), st);
 
 	ret = snd_hwparams_to_dma_slave_config(st, params, sc);
 	if (ret)

sound/soc/intel/atom/sst-atom-controls.c

···
 				dai->capture_widget->name);
 		w = dai->capture_widget;
 		snd_soc_dapm_widget_for_each_source_path(w, p) {
-			if (p->connected && !p->connected(w, p->sink))
+			if (p->connected && !p->connected(w, p->source))
 				continue;
 
 			if (p->connect &&  p->source->power &&

sound/soc/intel/atom/sst-mfld-platform-pcm.c

···
 static int sst_soc_pcm_new(struct snd_soc_component *component,
 			   struct snd_soc_pcm_runtime *rtd)
 {
-	struct snd_soc_dai *dai = rtd->cpu_dai;
+	struct snd_soc_dai *dai = asoc_rtd_to_cpu(rtd, 0);
 	struct snd_pcm *pcm = rtd->pcm;
 
 	if (dai->driver->playback.channels_min ||
···
 
 	/* set the SSPs to idle */
 	for_each_card_rtds(drv->soc_card, rtd) {
-		struct snd_soc_dai *dai = rtd->cpu_dai;
+		struct snd_soc_dai *dai = asoc_rtd_to_cpu(rtd, 0);
 
 		if (dai->active) {
 			send_ssp_cmd(dai, dai->name, 0);
···
 
 	/* restart SSPs */
 	for_each_card_rtds(drv->soc_card, rtd) {
-		struct snd_soc_dai *dai = rtd->cpu_dai;
+		struct snd_soc_dai *dai = asoc_rtd_to_cpu(rtd, 0);
 
 		if (dai->active) {
 			sst_handle_vb_timer(dai, true);

sound/soc/intel/atom/sst/sst_pci.c

···
 	dev_dbg(ctx->dev, "DRAM Ptr %p\n", ctx->dram);
 do_release_regions:
 	pci_release_regions(pci);
-	return 0;
+	return ret;
 }
 
 /*

sound/soc/intel/boards/Kconfig

···
 	select SND_SOC_DA7219
 	select SND_SOC_MAX98357A
 	select SND_SOC_DMIC
-	select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC
 	select SND_SOC_HDAC_HDMI
 
 config SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON
···
 	tristate "Broxton with DA7219 and MAX98357A in I2S Mode"
 	depends on I2C && ACPI
 	depends on MFD_INTEL_LPSS || COMPILE_TEST
+	depends on SND_HDA_CODEC_HDMI
 	select SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON
 	help
 	   This adds support for ASoC machine driver for Broxton-P platforms
···
 	tristate "GLK with DA7219 and MAX98357A in I2S Mode"
 	depends on I2C && ACPI
 	depends on MFD_INTEL_LPSS || COMPILE_TEST
+	depends on SND_HDA_CODEC_HDMI
 	select SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON
 	help
 	   This adds support for ASoC machine driver for Geminilake platforms
···
 	tristate "GLK with RT5682 and MAX98357A in I2S Mode"
 	depends on I2C && ACPI
 	depends on MFD_INTEL_LPSS || COMPILE_TEST
+	depends on SND_HDA_CODEC_HDMI
 	select SND_SOC_RT5682
 	select SND_SOC_MAX98357A
 	select SND_SOC_DMIC
-	select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC
 	select SND_SOC_HDAC_HDMI
 	help
 	   This adds support for ASoC machine driver for Geminilake platforms
···
 
 config SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH
 	tristate "SKL/KBL/BXT/APL with HDA Codecs"
-	select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC
+	depends on SND_HDA_CODEC_HDMI
 	select SND_SOC_HDAC_HDMI
 	select SND_SOC_DMIC
 	# SND_SOC_HDAC_HDA is already selected
···
 	depends on I2C && ACPI
 	depends on (SND_SOC_SOF_HDA_LINK && (MFD_INTEL_LPSS || COMPILE_TEST)) ||\
 		   (SND_SOC_SOF_BAYTRAIL && (X86_INTEL_LPSS || COMPILE_TEST))
+	depends on SND_HDA_CODEC_HDMI
+	select SND_SOC_MAX98373
+	select SND_SOC_RT1015
 	select SND_SOC_RT5682
 	select SND_SOC_DMIC
-	select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC
 	select SND_SOC_HDAC_HDMI
 	help
 	   This adds support for ASoC machine driver for SOF platforms
 	   with rt5682 codec.
 	   Say Y if you have such a device.
 	   If unsure select "N".
+
+config SND_SOC_INTEL_SOF_PCM512x_MACH
+	tristate "SOF with TI PCM512x codec"
+	depends on I2C && ACPI
+	depends on (SND_SOC_SOF_HDA_AUDIO_CODEC && (MFD_INTEL_LPSS || COMPILE_TEST)) ||\
+		   (SND_SOC_SOF_BAYTRAIL && (X86_INTEL_LPSS || COMPILE_TEST))
+	depends on SND_HDA_CODEC_HDMI
+	select SND_SOC_PCM512x_I2C
+	help
+	  This adds support for ASoC machine driver for SOF platforms
+	  with TI PCM512x I2S audio codec.
+	  Say Y or m if you have such a device.
+	  If unsure select "N".
+
 endif ## SND_SOC_SOF_HDA_LINK || SND_SOC_SOF_BAYTRAIL
 
 if (SND_SOC_SOF_COMETLAKE_LP && SND_SOC_SOF_HDA_LINK)
···
 	tristate "CML with RT1011 and RT5682 in I2S Mode"
 	depends on I2C && ACPI
 	depends on MFD_INTEL_LPSS || COMPILE_TEST
+	depends on SND_HDA_CODEC_HDMI
 	select SND_SOC_RT1011
 	select SND_SOC_RT5682
 	select SND_SOC_DMIC
 	select SND_SOC_HDAC_HDMI
-	select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC
 	help
 	  This adds support for ASoC machine driver for SOF platform with
 	  RT1011 + RT5682 I2S codec.
···
 if SND_SOC_SOF_JASPERLAKE
 
 config SND_SOC_INTEL_SOF_DA7219_MAX98373_MACH
-	tristate "SOF with DA7219 and MAX98373 in I2S Mode"
+	tristate "SOF with DA7219 and MAX98373/MAX98360A in I2S Mode"
 	depends on I2C && ACPI
 	depends on MFD_INTEL_LPSS || COMPILE_TEST
+	depends on SND_HDA_CODEC_HDMI
 	select SND_SOC_DA7219
 	select SND_SOC_MAX98373
 	select SND_SOC_DMIC
-	select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC
 	help
 	  This adds support for ASoC machine driver for SOF platforms
-	  with DA7219 + MAX98373 I2S audio codec.
+	  with DA7219 + MAX98373/MAX98360A I2S audio codec.
 	  Say Y if you have such a device.
 	  If unsure select "N".
 
 endif ## SND_SOC_SOF_JASPERLAKE
+
+if SND_SOC_SOF_INTEL_SOUNDWIRE
+
+config SND_SOC_INTEL_SOUNDWIRE_SOF_MACH
+	tristate "SoundWire generic machine driver"
+	depends on I2C && ACPI
+	depends on MFD_INTEL_LPSS || COMPILE_TEST
+	depends on SND_SOC_INTEL_USER_FRIENDLY_LONG_NAMES || COMPILE_TEST
+	depends on SOUNDWIRE
+	depends on SND_HDA_CODEC_HDMI
+	select SND_SOC_RT700_SDW
+	select SND_SOC_RT711_SDW
+	select SND_SOC_RT1308_SDW
+	select SND_SOC_RT1308
+	select SND_SOC_RT715_SDW
+	select SND_SOC_RT5682_SDW
+	select SND_SOC_DMIC
+        help
+	  Add support for Intel SoundWire-based platforms connected to
+	  RT700, RT711, RT1308 and RT715
+	  If unsure select "N".
+
+endif
+
 
 endif ## SND_SOC_INTEL_MACH

sound/soc/intel/boards/Makefile

···
 snd-soc-sst-broadwell-objs := broadwell.o
 snd-soc-sst-bxt-da7219_max98357a-objs := bxt_da7219_max98357a.o hda_dsp_common.o
 snd-soc-sst-bxt-rt298-objs := bxt_rt298.o hda_dsp_common.o
+snd-soc-sst-sof-pcm512x-objs := sof_pcm512x.o hda_dsp_common.o
 snd-soc-sst-glk-rt5682_max98357a-objs := glk_rt5682_max98357a.o hda_dsp_common.o
 snd-soc-sst-bytcr-rt5640-objs := bytcr_rt5640.o
 snd-soc-sst-bytcr-rt5651-objs := bytcr_rt5651.o
···
 snd-soc-sst-byt-cht-da7213-objs := bytcht_da7213.o
 snd-soc-sst-byt-cht-es8316-objs := bytcht_es8316.o
 snd-soc-sst-byt-cht-nocodec-objs := bytcht_nocodec.o
-snd-soc-sof_rt5682-objs := sof_rt5682.o hda_dsp_common.o
+snd-soc-sof_rt5682-objs := sof_rt5682.o hda_dsp_common.o sof_maxim_common.o
 snd-soc-cml_rt1011_rt5682-objs := cml_rt1011_rt5682.o hda_dsp_common.o
 snd-soc-kbl_da7219_max98357a-objs := kbl_da7219_max98357a.o
 snd-soc-kbl_da7219_max98927-objs := kbl_da7219_max98927.o
···
 snd-skl_nau88l25_max98357a-objs := skl_nau88l25_max98357a.o
 snd-soc-skl_nau88l25_ssm4567-objs := skl_nau88l25_ssm4567.o
 snd-soc-sof_da7219_max98373-objs := sof_da7219_max98373.o hda_dsp_common.o
-
+snd-soc-sof-sdw-objs += sof_sdw.o				\
+			sof_sdw_rt711.o sof_sdw_rt700.o		\
+			sof_sdw_rt1308.o sof_sdw_rt715.o	\
+			sof_sdw_rt5682.o			\
+			sof_sdw_dmic.o sof_sdw_hdmi.o hda_dsp_common.o
 obj-$(CONFIG_SND_SOC_INTEL_SOF_RT5682_MACH) += snd-soc-sof_rt5682.o
 obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o
 obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o
 obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o
 obj-$(CONFIG_SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON) += snd-soc-sst-bxt-da7219_max98357a.o
 obj-$(CONFIG_SND_SOC_INTEL_BXT_RT298_MACH) += snd-soc-sst-bxt-rt298.o
+obj-$(CONFIG_SND_SOC_INTEL_SOF_PCM512x_MACH) += snd-soc-sst-sof-pcm512x.o
 obj-$(CONFIG_SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH) += snd-soc-sst-glk-rt5682_max98357a.o
 obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o
 obj-$(CONFIG_SND_SOC_INTEL_BDW_RT5650_MACH) += snd-soc-sst-bdw-rt5650-mach.o
···
 obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_SSM4567_MACH) += snd-soc-skl_nau88l25_ssm4567.o
 obj-$(CONFIG_SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH) += snd-soc-skl_hda_dsp.o
 obj-$(CONFIG_SND_SOC_INTEL_SOF_DA7219_MAX98373_MACH) += snd-soc-sof_da7219_max98373.o
-
+obj-$(CONFIG_SND_SOC_INTEL_SOUNDWIRE_SOF_MACH) += snd-soc-sof-sdw.o

sound/soc/intel/boards/bdw-rt5650.c

···
 	struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	int ret;
 
 	/* Workaround: set codec PLL to 19.2MHz that PLL source is
···
 {
 	struct bdw_rt5650_priv *bdw_rt5650 =
 		snd_soc_card_get_drvdata(rtd->card);
-	struct snd_soc_component *component = rtd->codec_dai->component;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+	struct snd_soc_component *component = codec_dai->component;
 	int ret;
 
 	/* Enable codec ASRC function for Stereo DAC/Stereo1 ADC/DMIC/I2S1.
···
 #if IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL)
 SND_SOC_DAILINK_DEF(ssp0_port,
 	    DAILINK_COMP_ARRAY(COMP_CPU("ssp0-port")));
-#else
-SND_SOC_DAILINK_DEF(ssp0_port,
-	    DAILINK_COMP_ARRAY(COMP_DUMMY()));
 #endif
 
 static struct snd_soc_dai_link bdw_rt5650_dais[] = {
···
 		.dpcm_playback = 1,
 		.dpcm_capture = 1,
 		.init = bdw_rt5650_init,
+#if !IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL)
+		SND_SOC_DAILINK_REG(dummy, be, dummy),
+#else
 		SND_SOC_DAILINK_REG(ssp0_port, be, platform),
+#endif
 	},
 };
 
···
 		return -ENOMEM;
 
 	/* override plaform name, if required */
-	mach = (&pdev->dev)->platform_data;
+	mach = pdev->dev.platform_data;
 	ret = snd_soc_fixup_dai_links_platform_name(&bdw_rt5650_card,
 						    mach->mach_params.platform);
 

sound/soc/intel/boards/bdw-rt5677.c

···
 	struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	int ret;
 
 	ret = snd_soc_dai_set_sysclk(codec_dai, RT5677_SCLK_S_MCLK, 24576000,
···
 	struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	int ret;
 
 	ret = snd_soc_dai_set_sysclk(codec_dai, RT5677_SCLK_S_PLL1, 24576000,
···
 {
 	struct bdw_rt5677_priv *bdw_rt5677 =
 			snd_soc_card_get_drvdata(rtd->card);
-	struct snd_soc_component *component = rtd->codec_dai->component;
+	struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
 	struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component);
 	int ret;
 
···
 #if IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL)
 SND_SOC_DAILINK_DEF(ssp0_port,
 	    DAILINK_COMP_ARRAY(COMP_CPU("ssp0-port")));
-#else
-SND_SOC_DAILINK_DEF(ssp0_port,
-	    DAILINK_COMP_ARRAY(COMP_DUMMY()));
 #endif
 
 /* Wake on voice interface */
···
 		.dpcm_playback = 1,
 		.dpcm_capture = 1,
 		.init = bdw_rt5677_init,
+#if !IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL)
+		SND_SOC_DAILINK_REG(dummy, be, dummy),
+#else
 		SND_SOC_DAILINK_REG(ssp0_port, be, platform),
+#endif
 	},
 };
 
···
 	}
 
 	/* override plaform name, if required */
-	mach = (&pdev->dev)->platform_data;
+	mach = pdev->dev.platform_data;
 	ret = snd_soc_fixup_dai_links_platform_name(&bdw_rt5677_card,
 						    mach->mach_params.platform);
 	if (ret)

sound/soc/intel/boards/broadwell.c

···
 
 static int broadwell_rt286_codec_init(struct snd_soc_pcm_runtime *rtd)
 {
-	struct snd_soc_component *component = rtd->codec_dai->component;
+	struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
 	int ret = 0;
 	ret = snd_soc_card_jack_new(rtd->card, "Headset",
 		SND_JACK_HEADSET | SND_JACK_BTN_0, &broadwell_headset,
···
 	struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	int ret;
 
 	ret = snd_soc_dai_set_sysclk(codec_dai, RT286_SCLK_S_PLL, 24000000,
···
 #if IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL)
 SND_SOC_DAILINK_DEF(ssp0_port,
 	    DAILINK_COMP_ARRAY(COMP_CPU("ssp0-port")));
-#else
-SND_SOC_DAILINK_DEF(ssp0_port,
-	    DAILINK_COMP_ARRAY(COMP_DUMMY()));
 #endif
 
 /* broadwell digital audio interface glue - connects codec <--> CPU */
···
 		.ops = &broadwell_rt286_ops,
 		.dpcm_playback = 1,
 		.dpcm_capture = 1,
+#if !IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL)
+		SND_SOC_DAILINK_REG(dummy, codec, dummy),
+#else
 		SND_SOC_DAILINK_REG(ssp0_port, codec, platform),
+#endif
 	},
 };
 
···
 	broadwell_rt286.dev = &pdev->dev;
 
 	/* override plaform name, if required */
-	mach = (&pdev->dev)->platform_data;
+	mach = pdev->dev.platform_data;
 	ret = snd_soc_fixup_dai_links_platform_name(&broadwell_rt286,
 						    mach->mach_params.platform);
 	if (ret)

sound/soc/intel/boards/bxt_da7219_max98357a.c

···
 static int broxton_da7219_codec_init(struct snd_soc_pcm_runtime *rtd)
 {
 	int ret;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_component *component = rtd->codec_dai->component;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+	struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
 	int clk_freq;
 
 	/* Configure sysclk for codec */
···
 static int broxton_hdmi_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct bxt_card_private *ctx = snd_soc_card_get_drvdata(rtd->card);
-	struct snd_soc_dai *dai = rtd->codec_dai;
+	struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0);
 	struct bxt_hdmi_pcm *pcm;
 
 	pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
···
 static int broxton_da7219_fe_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_dapm_context *dapm;
-	struct snd_soc_component *component = rtd->cpu_dai->component;
+	struct snd_soc_component *component = asoc_rtd_to_cpu(rtd, 0)->component;
 
 	dapm = snd_soc_component_get_dapm(component);
 	snd_soc_dapm_ignore_suspend(dapm, "Reference Capture");
···
 	}
 
 	/* override plaform name, if required */
-	mach = (&pdev->dev)->platform_data;
+	mach = pdev->dev.platform_data;
 	platform_name = mach->mach_params.platform;
 
 	ret = snd_soc_fixup_dai_links_platform_name(&broxton_audio_card,

sound/soc/intel/boards/bxt_rt298.c

···
 static int broxton_rt298_fe_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_dapm_context *dapm;
-	struct snd_soc_component *component = rtd->cpu_dai->component;
+	struct snd_soc_component *component = asoc_rtd_to_cpu(rtd, 0)->component;
 
 	dapm = snd_soc_component_get_dapm(component);
 	snd_soc_dapm_ignore_suspend(dapm, "Reference Capture");
···
 
 static int broxton_rt298_codec_init(struct snd_soc_pcm_runtime *rtd)
 {
-	struct snd_soc_component *component = rtd->codec_dai->component;
+	struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
 	int ret = 0;
 
 	ret = snd_soc_card_jack_new(rtd->card, "Headset",
···
 static int broxton_hdmi_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct bxt_rt286_private *ctx = snd_soc_card_get_drvdata(rtd->card);
-	struct snd_soc_dai *dai = rtd->codec_dai;
+	struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0);
 	struct bxt_hdmi_pcm *pcm;
 
 	pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
···
 	struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	int ret;
 
 	ret = snd_soc_dai_set_sysclk(codec_dai, RT298_SCLK_S_PLL,
···
 	snd_soc_card_set_drvdata(card, ctx);
 
 	/* override plaform name, if required */
-	mach = (&pdev->dev)->platform_data;
+	mach = pdev->dev.platform_data;
 	platform_name = mach->mach_params.platform;
 
 	ret = snd_soc_fixup_dai_links_platform_name(card,

sound/soc/intel/boards/byt-max98090.c

···
 
 	card->dapm.idle_bias_off = true;
 
-	ret = snd_soc_dai_set_sysclk(runtime->codec_dai,
+	ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(runtime, 0),
 				     M98090_REG_SYSTEM_CLOCK,
 				     25000000, SND_SOC_CLOCK_IN);
 	if (ret < 0) {

sound/soc/intel/boards/byt-rt5640.c

···
 				struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	int ret;
 
 	ret = snd_soc_dai_set_sysclk(codec_dai, RT5640_SCLK_S_PLL1,
···
 static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime)
 {
 	int ret;
-	struct snd_soc_component *component = runtime->codec_dai->component;
+	struct snd_soc_component *component = asoc_rtd_to_codec(runtime, 0)->component;
 	struct snd_soc_card *card = runtime->card;
 	const struct snd_soc_dapm_route *custom_map;
 	int num_routes;

sound/soc/intel/boards/bytcht_cx2072x.c

···
 static int byt_cht_cx2072x_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_card *card = rtd->card;
-	struct snd_soc_component *codec = rtd->codec_dai->component;
+	struct snd_soc_component *codec = asoc_rtd_to_codec(rtd, 0)->component;
 	int ret;
 
 	if (devm_acpi_dev_add_driver_gpios(codec->dev,
···
 	card->dapm.idle_bias_off = true;
 
 	/* set the default PLL rate, the clock is handled by the codec driver */
-	ret = snd_soc_dai_set_sysclk(rtd->codec_dai, CX2072X_MCLK_EXTERNAL_PLL,
+	ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(rtd, 0), CX2072X_MCLK_EXTERNAL_PLL,
 				     19200000, SND_SOC_CLOCK_IN);
 	if (ret) {
 		dev_err(rtd->dev, "Could not set sysclk\n");
···
 
 	snd_soc_component_set_jack(codec, &byt_cht_cx2072x_headset, NULL);
 
-	snd_soc_dai_set_bclk_ratio(rtd->codec_dai, 50);
+	snd_soc_dai_set_bclk_ratio(asoc_rtd_to_codec(rtd, 0), 50);
 
 	return ret;
 }
···
 	 * with explicit setting to I2S 2ch 24-bit. The word length is set with
 	 * dai_set_tdm_slot() since there is no other API exposed
 	 */
-	ret = snd_soc_dai_set_fmt(rtd->cpu_dai,
+	ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0),
 				SND_SOC_DAIFMT_I2S     |
 				SND_SOC_DAIFMT_NB_NF   |
 				SND_SOC_DAIFMT_CBS_CFS);
···
 		return ret;
 	}
 
-	ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, 24);
+	ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_cpu(rtd, 0), 0x3, 0x3, 2, 24);
 	if (ret < 0) {
 		dev_err(rtd->dev, "can't set I2S config, err %d\n", ret);
 		return ret;

sound/soc/intel/boards/bytcht_da7213.c

···
 	 * with explicit setting to I2S 2ch 24-bit. The word length is set with
 	 * dai_set_tdm_slot() since there is no other API exposed
 	 */
-	ret = snd_soc_dai_set_fmt(rtd->cpu_dai,
+	ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0),
 				  SND_SOC_DAIFMT_I2S     |
 				  SND_SOC_DAIFMT_NB_NF   |
 				  SND_SOC_DAIFMT_CBS_CFS);
···
 		return ret;
 	}
 
-	ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, 24);
+	ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_cpu(rtd, 0), 0x3, 0x3, 2, 24);
 	if (ret < 0) {
 		dev_err(rtd->dev, "can't set I2S config, err %d\n", ret);
 		return ret;
···
 			  struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	int ret;
 
 	ret = snd_soc_dai_set_sysclk(codec_dai, DA7213_CLKSRC_MCLK,
···
 static int aif1_hw_free(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	int ret;
 
 	ret = snd_soc_dai_set_pll(codec_dai, 0,
···
 	int ret_val = 0;
 	int i;
 
-	mach = (&pdev->dev)->platform_data;
+	mach = pdev->dev.platform_data;
 	card = &bytcht_da7213_card;
 	card->dev = &pdev->dev;
 

sound/soc/intel/boards/bytcht_es8316.c

···
 
 static int byt_cht_es8316_init(struct snd_soc_pcm_runtime *runtime)
 {
-	struct snd_soc_component *codec = runtime->codec_dai->component;
+	struct snd_soc_component *codec = asoc_rtd_to_codec(runtime, 0)->component;
 	struct snd_soc_card *card = runtime->card;
 	struct byt_cht_es8316_private *priv = snd_soc_card_get_drvdata(card);
 	const struct snd_soc_dapm_route *custom_map;
···
 	if (ret)
 		dev_err(card->dev, "unable to enable MCLK\n");
 
-	ret = snd_soc_dai_set_sysclk(runtime->codec_dai, 0, 19200000,
+	ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(runtime, 0), 0, 19200000,
 				     SND_SOC_CLOCK_IN);
 	if (ret < 0) {
 		dev_err(card->dev, "can't set codec clock %d\n", ret);
···
 	 * with explicit setting to I2S 2ch 24-bit. The word length is set with
 	 * dai_set_tdm_slot() since there is no other API exposed
 	 */
-	ret = snd_soc_dai_set_fmt(rtd->cpu_dai,
+	ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0),
 				SND_SOC_DAIFMT_I2S     |
 				SND_SOC_DAIFMT_NB_NF   |
 				SND_SOC_DAIFMT_CBS_CFS
···
 		return ret;
 	}
 
-	ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, bits);
+	ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_cpu(rtd, 0), 0x3, 0x3, 2, bits);
 	if (ret < 0) {
 		dev_err(rtd->dev, "can't set I2S config, err %d\n", ret);
 		return ret;

sound/soc/intel/boards/bytcht_nocodec.c

···
 	 * with explicit setting to I2S 2ch 24-bit. The word length is set with
 	 * dai_set_tdm_slot() since there is no other API exposed
 	 */
-	ret = snd_soc_dai_set_fmt(rtd->cpu_dai,
+	ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0),
 				  SND_SOC_DAIFMT_I2S     |
 				  SND_SOC_DAIFMT_NB_NF   |
 				  SND_SOC_DAIFMT_CBS_CFS);
···
 		return ret;
 	}
 
-	ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, 24);
+	ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_cpu(rtd, 0), 0x3, 0x3, 2, 24);
 	if (ret < 0) {
 		dev_err(rtd->dev, "can't set I2S config, err %d\n", ret);
 		return ret;

sound/soc/intel/boards/bytcr_rt5640.c

···
 					struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *dai = rtd->codec_dai;
+	struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0);
 
 	return byt_rt5640_prepare_and_enable_pll1(dai, params_rate(params));
 }
···
 {
 	struct snd_soc_card *card = runtime->card;
 	struct byt_rt5640_private *priv = snd_soc_card_get_drvdata(card);
-	struct snd_soc_component *component = runtime->codec_dai->component;
+	struct snd_soc_component *component = asoc_rtd_to_codec(runtime, 0)->component;
 	const struct snd_soc_dapm_route *custom_map;
 	int num_routes;
 	int ret;
···
 	 * with explicit setting to I2S 2ch. The word length is set with
 	 * dai_set_tdm_slot() since there is no other API exposed
 	 */
-	ret = snd_soc_dai_set_fmt(rtd->cpu_dai,
+	ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0),
 				  SND_SOC_DAIFMT_I2S     |
 				  SND_SOC_DAIFMT_NB_NF   |
 				  SND_SOC_DAIFMT_CBS_CFS);
···
 		return ret;
 	}
 
-	ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, bits);
+	ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_cpu(rtd, 0), 0x3, 0x3, 2, bits);
 	if (ret < 0) {
 		dev_err(rtd->dev, "can't set I2S config, err %d\n", ret);
 		return ret;

sound/soc/intel/boards/bytcr_rt5651.c

···
 					struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	snd_pcm_format_t format = params_format(params);
 	int rate = params_rate(params);
 	int bclk_ratio;
···
 static int byt_rt5651_init(struct snd_soc_pcm_runtime *runtime)
 {
 	struct snd_soc_card *card = runtime->card;
-	struct snd_soc_component *codec = runtime->codec_dai->component;
+	struct snd_soc_component *codec = asoc_rtd_to_codec(runtime, 0)->component;
 	struct byt_rt5651_private *priv = snd_soc_card_get_drvdata(card);
 	const struct snd_soc_dapm_route *custom_map;
 	int num_routes;
···
 	 * with explicit setting to I2S 2ch. The word length is set with
 	 * dai_set_tdm_slot() since there is no other API exposed
 	 */
-	ret = snd_soc_dai_set_fmt(rtd->cpu_dai,
+	ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0),
 				  SND_SOC_DAIFMT_I2S     |
 				  SND_SOC_DAIFMT_NB_NF   |
 				  SND_SOC_DAIFMT_CBS_CFS
···
 		return ret;
 	}
 
-	ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, bits);
+	ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_cpu(rtd, 0), 0x3, 0x3, 2, bits);
 	if (ret < 0) {
 		dev_err(rtd->dev, "can't set I2S config, err %d\n", ret);
 		return ret;

sound/soc/intel/boards/cht_bsw_max98090_ti.c

···
 			     struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	int ret;
 
 	ret = snd_soc_dai_set_sysclk(codec_dai, M98090_REG_SYSTEM_CLOCK,
···
 	int ret = 0;
 	unsigned int fmt = 0;
 
-	ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, 16);
+	ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_cpu(rtd, 0), 0x3, 0x3, 2, 16);
 	if (ret < 0) {
 		dev_err(rtd->dev, "can't set cpu_dai slot fmt: %d\n", ret);
 		return ret;
···
 	fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
 				| SND_SOC_DAIFMT_CBS_CFS;
 
-	ret = snd_soc_dai_set_fmt(rtd->cpu_dai, fmt);
+	ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), fmt);
 	if (ret < 0) {
 		dev_err(rtd->dev, "can't set cpu_dai set fmt: %d\n", ret);
 		return ret;
···
 
 	/* override plaform name, if required */
 	snd_soc_card_cht.dev = &pdev->dev;
-	mach = (&pdev->dev)->platform_data;
+	mach = pdev->dev.platform_data;
 	platform_name = mach->mach_params.platform;
 
 	ret_val = snd_soc_fixup_dai_links_platform_name(&snd_soc_card_cht,

sound/soc/intel/boards/cht_bsw_nau8824.c

···
 	struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	int ret;
 
 	ret = snd_soc_dai_set_sysclk(codec_dai, NAU8824_CLK_FLL_FS, 0,
···
 {
 	struct cht_mc_private *ctx = snd_soc_card_get_drvdata(runtime->card);
 	struct snd_soc_jack *jack = &ctx->jack;
-	struct snd_soc_dai *codec_dai = runtime->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(runtime, 0);
 	struct snd_soc_component *component = codec_dai->component;
 	int ret, jack_type;
 
···
 
 	/* override plaform name, if required */
 	snd_soc_card_cht.dev = &pdev->dev;
-	mach = (&pdev->dev)->platform_data;
+	mach = pdev->dev.platform_data;
 	platform_name = mach->mach_params.platform;
 
 	ret_val = snd_soc_fixup_dai_links_platform_name(&snd_soc_card_cht,

sound/soc/intel/boards/cht_bsw_rt5645.c

···
 			     struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	int ret;
 
 	/* set codec PLL source to the 19.2MHz platform clock (MCLK) */
···
 {
 	struct snd_soc_card *card = runtime->card;
 	struct cht_mc_private *ctx = snd_soc_card_get_drvdata(runtime->card);
-	struct snd_soc_component *component = runtime->codec_dai->component;
+	struct snd_soc_component *component = asoc_rtd_to_codec(runtime, 0)->component;
 	int jack_type;
 	int ret;
 
···
 		 * with explicit setting to I2S 2ch 16-bit. The word length is set with
 		 * dai_set_tdm_slot() since there is no other API exposed
 		 */
-		ret = snd_soc_dai_set_fmt(rtd->cpu_dai,
+		ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0),
 					SND_SOC_DAIFMT_I2S     |
 					SND_SOC_DAIFMT_NB_NF   |
 					SND_SOC_DAIFMT_CBS_CFS
···
 			return ret;
 		}
 
-		ret = snd_soc_dai_set_fmt(rtd->codec_dai,
+		ret = snd_soc_dai_set_fmt(asoc_rtd_to_codec(rtd, 0),
 					SND_SOC_DAIFMT_I2S     |
 					SND_SOC_DAIFMT_NB_NF   |
 					SND_SOC_DAIFMT_CBS_CFS
···
 			return ret;
 		}
 
-		ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, 16);
+		ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_cpu(rtd, 0), 0x3, 0x3, 2, 16);
 		if (ret < 0) {
 			dev_err(rtd->dev, "can't set I2S config, err %d\n", ret);
 			return ret;
···
 		/*
 		 * Default mode for SSP configuration is TDM 4 slot
 		 */
-		ret = snd_soc_dai_set_fmt(rtd->codec_dai,
+		ret = snd_soc_dai_set_fmt(asoc_rtd_to_codec(rtd, 0),
 					SND_SOC_DAIFMT_DSP_B |
 					SND_SOC_DAIFMT_IB_NF |
 					SND_SOC_DAIFMT_CBS_CFS);
···
 		}
 
 		/* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */
-		ret = snd_soc_dai_set_tdm_slot(rtd->codec_dai, 0xF, 0xF, 4, 24);
+		ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_codec(rtd, 0), 0xF, 0xF, 4, 24);
 		if (ret < 0) {
 			dev_err(rtd->dev, "can't set codec TDM slot %d\n", ret);
 			return ret;
···
 	if (!drv)
 		return -ENOMEM;
 
-	mach = (&pdev->dev)->platform_data;
+	mach = pdev->dev.platform_data;
 
 	for (i = 0; i < ARRAY_SIZE(snd_soc_cards); i++) {
 		if (acpi_dev_found(snd_soc_cards[i].codec_id) &&

sound/soc/intel/boards/cht_bsw_rt5672.c

···
 					struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	int ret;
 
 	/* set codec PLL source to the 19.2MHz platform clock (MCLK) */
···
 static int cht_codec_init(struct snd_soc_pcm_runtime *runtime)
 {
 	int ret;
-	struct snd_soc_dai *codec_dai = runtime->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(runtime, 0);
 	struct snd_soc_component *component = codec_dai->component;
 	struct cht_mc_private *ctx = snd_soc_card_get_drvdata(runtime->card);
 
···
 	/*
 	 * Default mode for SSP configuration is TDM 4 slot
 	 */
-	ret = snd_soc_dai_set_fmt(rtd->codec_dai,
+	ret = snd_soc_dai_set_fmt(asoc_rtd_to_codec(rtd, 0),
 				  SND_SOC_DAIFMT_DSP_B |
 				  SND_SOC_DAIFMT_IB_NF |
 				  SND_SOC_DAIFMT_CBS_CFS);
···
 	}
 
 	/* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */
-	ret = snd_soc_dai_set_tdm_slot(rtd->codec_dai, 0xF, 0xF, 4, 24);
+	ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_codec(rtd, 0), 0xF, 0xF, 4, 24);
 	if (ret < 0) {
 		dev_err(rtd->dev, "can't set codec TDM slot %d\n", ret);
 		return ret;

sound/soc/intel/boards/cml_rt1011_rt5682.c

···
 static int cml_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct card_private *ctx = snd_soc_card_get_drvdata(rtd->card);
-	struct snd_soc_component *component = rtd->codec_dai->component;
+	struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
 	struct snd_soc_jack *jack;
 	int ret;
 
···
 				struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	int clk_id, clk_freq, pll_out, ret;
 
 	clk_id = RT5682_PLL1_S_MCLK;
···
 
 	srate = params_rate(params);
 
-	for (i = 0; i < rtd->num_codecs; i++) {
-		codec_dai = rtd->codec_dais[i];
+	for_each_rtd_codec_dais(rtd, i, codec_dai) {
 
 		/* 100 Fs to drive 24 bit data */
 		ret = snd_soc_dai_set_pll(codec_dai, 0, RT1011_PLL1_S_BCLK,
···
 static int hdmi_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct card_private *ctx = snd_soc_card_get_drvdata(rtd->card);
-	struct snd_soc_dai *dai = rtd->codec_dai;
+	struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0);
 	struct hdmi_pcm *pcm;
 
 	pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
···
 	const char *platform_name;
 	int ret;
 
-	ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC);
+	ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL);
 	if (!ctx)
 		return -ENOMEM;
 
 	INIT_LIST_HEAD(&ctx->hdmi_pcm_list);
-	mach = (&pdev->dev)->platform_data;
+	mach = pdev->dev.platform_data;
 	snd_soc_card_cml.dev = &pdev->dev;
 	platform_name = mach->mach_params.platform;
 

sound/soc/intel/boards/glk_rt5682_max98357a.c

···
 static int geminilake_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct glk_card_private *ctx = snd_soc_card_get_drvdata(rtd->card);
-	struct snd_soc_component *component = rtd->codec_dai->component;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	struct snd_soc_jack *jack;
 	int ret;
 
···
 	struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	int ret;
 
 	/* Set valid bitmask & configuration for I2S in 24 bit */
···
 static int geminilake_hdmi_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct glk_card_private *ctx = snd_soc_card_get_drvdata(rtd->card);
-	struct snd_soc_dai *dai = rtd->codec_dai;
+	struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0);
 	struct glk_hdmi_pcm *pcm;
 
 	pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
···
 
 static int geminilake_rt5682_fe_init(struct snd_soc_pcm_runtime *rtd)
 {
-	struct snd_soc_component *component = rtd->cpu_dai->component;
+	struct snd_soc_component *component = asoc_rtd_to_cpu(rtd, 0)->component;
 	struct snd_soc_dapm_context *dapm;
 	int ret;
 
···
 		.init = NULL,
 		.capture_only = 1,
 		.nonatomic = 1,
+		.dynamic = 1,
 		SND_SOC_DAILINK_REG(echoref, dummy, platform),
 	},
 	[GLK_DPCM_AUDIO_REF_CP] = {
···
 	snd_soc_card_set_drvdata(card, ctx);
 
 	/* override plaform name, if required */
-	mach = (&pdev->dev)->platform_data;
+	mach = pdev->dev.platform_data;
 	platform_name = mach->mach_params.platform;
 
 	ret = snd_soc_fixup_dai_links_platform_name(card, platform_name);

sound/soc/intel/boards/haswell.c

···
 	struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	int ret;
 
 	ret = snd_soc_dai_set_sysclk(codec_dai, RT5640_SCLK_S_MCLK, 12288000,
···
 	haswell_rt5640.dev = &pdev->dev;
 
 	/* override plaform name, if required */
-	mach = (&pdev->dev)->platform_data;
+	mach = pdev->dev.platform_data;
 	ret = snd_soc_fixup_dai_links_platform_name(&haswell_rt5640,
 						    mach->mach_params.platform);
 	if (ret)

sound/soc/intel/boards/kbl_da7219_max98357a.c

···
 static int kabylake_da7219_codec_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card);
-	struct snd_soc_component *component = rtd->codec_dai->component;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	struct snd_soc_jack *jack;
 	int ret;
 
···
 static int kabylake_hdmi_init(struct snd_soc_pcm_runtime *rtd, int device)
 {
 	struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card);
-	struct snd_soc_dai *dai = rtd->codec_dai;
+	struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0);
 	struct kbl_hdmi_pcm *pcm;
 
 	pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
···
 static int kabylake_da7219_fe_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_dapm_context *dapm;
-	struct snd_soc_component *component = rtd->cpu_dai->component;
+	struct snd_soc_component *component = asoc_rtd_to_cpu(rtd, 0)->component;
 
 	dapm = snd_soc_component_get_dapm(component);
 	snd_soc_dapm_ignore_suspend(dapm, "Reference Capture");

sound/soc/intel/boards/kbl_da7219_max98927.c

···
 	struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *runtime = substream->private_data;
+	struct snd_soc_dai *codec_dai;
 	int ret, j;
 
-	for (j = 0; j < runtime->num_codecs; j++) {
-		struct snd_soc_dai *codec_dai = runtime->codec_dais[j];
+	for_each_rtd_codec_dais(runtime, j, codec_dai) {
 
 		if (!strcmp(codec_dai->component->name, MAX98927_DEV0_NAME)) {
 			ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x30, 3, 8, 16);
···
 static int kabylake_ssp0_trigger(struct snd_pcm_substream *substream, int cmd)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai;
 	int j, ret;
 
-	for (j = 0; j < rtd->num_codecs; j++) {
-		struct snd_soc_dai *codec_dai = rtd->codec_dais[j];
+	for_each_rtd_codec_dais(rtd, j, codec_dai) {
 		const char *name = codec_dai->component->name;
 		struct snd_soc_component *component = codec_dai->component;
 		struct snd_soc_dapm_context *dapm =
···
 static int kabylake_da7219_codec_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card);
-	struct snd_soc_component *component = rtd->codec_dai->component;
+	struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
 	struct snd_soc_jack *jack;
 	struct snd_soc_card *card = rtd->card;
 	int ret;
···
 static int kabylake_hdmi_init(struct snd_soc_pcm_runtime *rtd, int device)
 {
 	struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card);
-	struct snd_soc_dai *dai = rtd->codec_dai;
+	struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0);
 	struct kbl_hdmi_pcm *pcm;
 
 	pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
···
 static int kabylake_da7219_fe_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_dapm_context *dapm;
-	struct snd_soc_component *component = rtd->cpu_dai->component;
+	struct snd_soc_component *component = asoc_rtd_to_cpu(rtd, 0)->component;
 
 	dapm = snd_soc_component_get_dapm(component);
 	snd_soc_dapm_ignore_suspend(dapm, "Reference Capture");

sound/soc/intel/boards/kbl_rt5660.c

···
 {
 	int ret;
 	struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card);
-	struct snd_soc_component *component = rtd->codec_dai->component;
+	struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
 	struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component);
 
 	ret = devm_acpi_dev_add_driver_gpios(component->dev, acpi_rt5660_gpios);
···
 static int kabylake_hdmi_init(struct snd_soc_pcm_runtime *rtd, int device)
 {
 	struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card);
-	struct snd_soc_dai *dai = rtd->codec_dai;
+	struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0);
 	struct kbl_hdmi_pcm *pcm;
 
 	pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
···
 	struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	int ret;
 
 	ret = snd_soc_dai_set_sysclk(codec_dai,

sound/soc/intel/boards/kbl_rt5663_max98927.c

···
 {
 	int ret;
 	struct snd_soc_dapm_context *dapm;
-	struct snd_soc_component *component = rtd->cpu_dai->component;
+	struct snd_soc_component *component = asoc_rtd_to_cpu(rtd, 0)->component;
 
 	dapm = snd_soc_component_get_dapm(component);
 	ret = snd_soc_dapm_ignore_suspend(dapm, "Reference Capture");
···
 {
 	int ret;
 	struct kbl_rt5663_private *ctx = snd_soc_card_get_drvdata(rtd->card);
-	struct snd_soc_component *component = rtd->codec_dai->component;
+	struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
 	struct snd_soc_jack *jack;
 
 	/*
···
 static int kabylake_hdmi_init(struct snd_soc_pcm_runtime *rtd, int device)
 {
 	struct kbl_rt5663_private *ctx = snd_soc_card_get_drvdata(rtd->card);
-	struct snd_soc_dai *dai = rtd->codec_dai;
+	struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0);
 	struct kbl_hdmi_pcm *pcm;
 
 	pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
···
 	struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	int ret;
 
 	/* use ASRC for internal clocks, as PLL rate isn't multiple of BCLK */
···
 	struct snd_soc_dai *codec_dai;
 	int ret = 0, j;
 
-	for_each_rtd_codec_dai(rtd, j, codec_dai) {
+	for_each_rtd_codec_dais(rtd, j, codec_dai) {
 		if (!strcmp(codec_dai->component->name, MAXIM_DEV0_NAME)) {
 			/*
 			 * Use channel 4 and 5 for the first amp
···
 	kabylake_audio_card->dev = &pdev->dev;
 	snd_soc_card_set_drvdata(kabylake_audio_card, ctx);
 
-	mach = (&pdev->dev)->platform_data;
+	mach = pdev->dev.platform_data;
 	if (mach)
 		dmic_constraints = mach->mach_params.dmic_num == 2 ?
 			&constraints_dmic_2ch : &constraints_dmic_channels;

sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c

···
 static int kabylake_rt5663_fe_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_dapm_context *dapm;
-	struct snd_soc_component *component = rtd->cpu_dai->component;
+	struct snd_soc_component *component = asoc_rtd_to_cpu(rtd, 0)->component;
 	int ret;
 
 	dapm = snd_soc_component_get_dapm(component);
···
 {
 	int ret;
 	struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card);
-	struct snd_soc_component *component = rtd->codec_dai->component;
+	struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
 	struct snd_soc_jack *jack;
 
 	/*
···
 static int kabylake_hdmi_init(struct snd_soc_pcm_runtime *rtd, int device)
 {
 	struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card);
-	struct snd_soc_dai *dai = rtd->codec_dai;
+	struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0);
 	struct kbl_hdmi_pcm *pcm;
 
 	pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
···
 	struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	int ret;
 
 	/* use ASRC for internal clocks, as PLL rate isn't multiple of BCLK */
···
 	struct snd_soc_dai *codec_dai;
 	int ret = 0, j;
 
-	for_each_rtd_codec_dai(rtd, j, codec_dai) {
+	for_each_rtd_codec_dais(rtd, j, codec_dai) {
 		if (!strcmp(codec_dai->component->name, RT5514_DEV_NAME)) {
 			ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0, 8, 16);
 			if (ret < 0) {
···
 	kabylake_audio_card.dev = &pdev->dev;
 	snd_soc_card_set_drvdata(&kabylake_audio_card, ctx);
 
-	mach = (&pdev->dev)->platform_data;
+	mach = pdev->dev.platform_data;
 	if (mach)
 		dmic_constraints = mach->mach_params.dmic_num == 2 ?
 			&constraints_dmic_2ch : &constraints_dmic_channels;

sound/soc/intel/boards/skl_hda_dsp_common.h

···
 	struct snd_soc_component *component;
 	struct skl_hda_hdmi_pcm *pcm;
 
+	/* HDMI disabled, do not create controls */
+	if (list_empty(&ctx->hdmi_pcm_list))
+		return 0;
+
 	pcm = list_first_entry(&ctx->hdmi_pcm_list, struct skl_hda_hdmi_pcm,
 			       head);
 	component = pcm->codec_dai->component;

sound/soc/intel/boards/skl_hda_dsp_generic.c

···
 	{ "Alt Analog CPU Capture", NULL, "Alt Analog Codec Capture" },
 };
 
+SND_SOC_DAILINK_DEF(dummy_codec,
+	DAILINK_COMP_ARRAY(COMP_CODEC("snd-soc-dummy", "snd-soc-dummy-dai")));
+
 static int skl_hda_card_late_probe(struct snd_soc_card *card)
 {
 	return skl_hda_hdmi_jack_init(card);
···
 {
 	struct snd_soc_card *card = &hda_soc_card;
 	struct snd_soc_dai_link *dai_link;
-	u32 codec_count, codec_mask;
+	u32 codec_count, codec_mask, idisp_mask;
 	int i, num_links, num_route;
 
 	codec_mask = mach_params->codec_mask;
 	codec_count = hweight_long(codec_mask);
+	idisp_mask = codec_mask & IDISP_CODEC_MASK;
 
-	if (codec_count == 1 && codec_mask & IDISP_CODEC_MASK) {
+	if (!codec_count || codec_count > 2 ||
+	    (codec_count == 2 && !idisp_mask))
+		return -EINVAL;
+
+	if (codec_mask == idisp_mask) {
+		/* topology with iDisp as the only HDA codec */
 		num_links = IDISP_DAI_COUNT + DMIC_DAI_COUNT;
 		num_route = IDISP_ROUTE_COUNT;
 
···
 				skl_hda_be_dai_links[IDISP_DAI_COUNT +
 					HDAC_DAI_COUNT + i];
 		}
-	} else if (codec_count == 2 && codec_mask & IDISP_CODEC_MASK) {
+	} else {
+		/* topology with external and iDisp HDA codecs */
 		num_links = ARRAY_SIZE(skl_hda_be_dai_links);
 		num_route = ARRAY_SIZE(skl_hda_map);
 		card->dapm_widgets = skl_hda_widgets;
 		card->num_dapm_widgets = ARRAY_SIZE(skl_hda_widgets);
-	} else {
-		return -EINVAL;
+		if (!idisp_mask) {
+			for (i = 0; i < IDISP_DAI_COUNT; i++) {
+				skl_hda_be_dai_links[i].codecs = dummy_codec;
+				skl_hda_be_dai_links[i].num_codecs =
+					ARRAY_SIZE(dummy_codec);
+			}
+		}
 	}
 
 	card->num_links = num_links;
···
 
 	INIT_LIST_HEAD(&ctx->hdmi_pcm_list);
 
-	mach = (&pdev->dev)->platform_data;
+	mach = pdev->dev.platform_data;
 	if (!mach)
 		return -EINVAL;
 

sound/soc/intel/boards/skl_nau88l25_max98357a.c

···
 static int skylake_nau8825_codec_init(struct snd_soc_pcm_runtime *rtd)
 {
 	int ret;
-	struct snd_soc_component *component = rtd->codec_dai->component;
+	struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
 
 	/*
 	 * Headset buttons map to the google Reference headset.
···
 static int skylake_hdmi1_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct skl_nau8825_private *ctx = snd_soc_card_get_drvdata(rtd->card);
-	struct snd_soc_dai *dai = rtd->codec_dai;
+	struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0);
 	struct skl_hdmi_pcm *pcm;
 
 	pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
···
 static int skylake_hdmi2_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct skl_nau8825_private *ctx = snd_soc_card_get_drvdata(rtd->card);
-	struct snd_soc_dai *dai = rtd->codec_dai;
+	struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0);
 	struct skl_hdmi_pcm *pcm;
 
 	pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
···
 static int skylake_hdmi3_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct skl_nau8825_private *ctx = snd_soc_card_get_drvdata(rtd->card);
-	struct snd_soc_dai *dai = rtd->codec_dai;
+	struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0);
 	struct skl_hdmi_pcm *pcm;
 
 	pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
···
 static int skylake_nau8825_fe_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_dapm_context *dapm;
-	struct snd_soc_component *component = rtd->cpu_dai->component;
+	struct snd_soc_component *component = asoc_rtd_to_cpu(rtd, 0)->component;
 
 	dapm = snd_soc_component_get_dapm(component);
 	snd_soc_dapm_ignore_suspend(dapm, "Reference Capture");
···
 	struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	int ret;
 
 	ret = snd_soc_dai_set_sysclk(codec_dai,
···
 	skylake_audio_card.dev = &pdev->dev;
 	snd_soc_card_set_drvdata(&skylake_audio_card, ctx);
 
-	mach = (&pdev->dev)->platform_data;
+	mach = pdev->dev.platform_data;
 	if (mach)
 		dmic_constraints = mach->mach_params.dmic_num == 2 ?
 			&constraints_dmic_2ch : &constraints_dmic_channels;

sound/soc/intel/boards/skl_nau88l25_ssm4567.c

···
 	int ret;
 
 	/* Slot 1 for left */
-	ret = snd_soc_dai_set_tdm_slot(rtd->codec_dais[0], 0x01, 0x01, 2, 48);
+	ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_codec(rtd, 0), 0x01, 0x01, 2, 48);
 	if (ret < 0)
 		return ret;
 
 	/* Slot 2 for right */
-	ret = snd_soc_dai_set_tdm_slot(rtd->codec_dais[1], 0x02, 0x02, 2, 48);
+	ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_codec(rtd, 1), 0x02, 0x02, 2, 48);
 	if (ret < 0)
 		return ret;
 
···
 static int skylake_nau8825_codec_init(struct snd_soc_pcm_runtime *rtd)
 {
 	int ret;
-	struct snd_soc_component *component = rtd->codec_dai->component;
+	struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
 
 	/*
 	 * 4 buttons here map to the google Reference headset
···
 static int skylake_hdmi1_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct skl_nau88125_private *ctx = snd_soc_card_get_drvdata(rtd->card);
-	struct snd_soc_dai *dai = rtd->codec_dai;
+	struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0);
 	struct skl_hdmi_pcm *pcm;
 
 	pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
···
 static int skylake_hdmi2_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct skl_nau88125_private *ctx = snd_soc_card_get_drvdata(rtd->card);
-	struct snd_soc_dai *dai = rtd->codec_dai;
+	struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0);
 	struct skl_hdmi_pcm *pcm;
 
 	pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
···
 static int skylake_hdmi3_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct skl_nau88125_private *ctx = snd_soc_card_get_drvdata(rtd->card);
-	struct snd_soc_dai *dai = rtd->codec_dai;
+	struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0);
 	struct skl_hdmi_pcm *pcm;
 
 	pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
···
 static int skylake_nau8825_fe_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_dapm_context *dapm;
-	struct snd_soc_component *component = rtd->cpu_dai->component;
+	struct snd_soc_component *component = asoc_rtd_to_cpu(rtd, 0)->component;
 
 	dapm = snd_soc_component_get_dapm(component);
 	snd_soc_dapm_ignore_suspend(dapm, "Reference Capture");
···
 	struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	int ret;
 
 	ret = snd_soc_dai_set_sysclk(codec_dai,
···
 	.codec_conf = ssm4567_codec_conf,
 	.num_configs = ARRAY_SIZE(ssm4567_codec_conf),
 	.fully_routed = true,
+	.disable_route_checks = true,
 	.late_probe = skylake_card_late_probe,
 };
 
···
 	skylake_audio_card.dev = &pdev->dev;
 	snd_soc_card_set_drvdata(&skylake_audio_card, ctx);
 
-	mach = (&pdev->dev)->platform_data;
+	mach = pdev->dev.platform_data;
 	if (mach)
 		dmic_constraints = mach->mach_params.dmic_num == 2 ?
 			&constraints_dmic_2ch : &constraints_dmic_channels;

sound/soc/intel/boards/skl_rt286.c

···
 static int skylake_rt286_fe_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_dapm_context *dapm;
-	struct snd_soc_component *component = rtd->cpu_dai->component;
+	struct snd_soc_component *component = asoc_rtd_to_cpu(rtd, 0)->component;
 
 	dapm = snd_soc_component_get_dapm(component);
 	snd_soc_dapm_ignore_suspend(dapm, "Reference Capture");
···
 
 static int skylake_rt286_codec_init(struct snd_soc_pcm_runtime *rtd)
 {
-	struct snd_soc_component *component = rtd->codec_dai->component;
+	struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
 	int ret;
 
 	ret = snd_soc_card_jack_new(rtd->card, "Headset",
···
 static int skylake_hdmi_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct skl_rt286_private *ctx = snd_soc_card_get_drvdata(rtd->card);
-	struct snd_soc_dai *dai = rtd->codec_dai;
+	struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0);
 	struct skl_hdmi_pcm *pcm;
 
 	pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
···
 	struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	int ret;
 
 	ret = snd_soc_dai_set_sysclk(codec_dai, RT286_SCLK_S_PLL, 24000000,

sound/soc/intel/boards/sof_da7219_max98373.c

···
 // Copyright(c) 2019 Intel Corporation.
 
 /*
- * Intel SOF Machine driver for DA7219 + MAX98373 codec
+ * Intel SOF Machine driver for DA7219 + MAX98373/MAX98360A codec
  */
 
 #include <linux/input.h>
···
 	SOC_DAPM_PIN_SWITCH("Right Spk"),
 };
 
+static const struct snd_kcontrol_new m98360a_controls[] = {
+	SOC_DAPM_PIN_SWITCH("Headphone Jack"),
+	SOC_DAPM_PIN_SWITCH("Headset Mic"),
+	SOC_DAPM_PIN_SWITCH("Spk"),
+};
+
+/* For MAX98373 amp */
 static const struct snd_soc_dapm_widget widgets[] = {
 	SND_SOC_DAPM_HP("Headphone Jack", NULL),
 	SND_SOC_DAPM_MIC("Headset Mic", NULL),
+
 	SND_SOC_DAPM_SPK("Left Spk", NULL),
 	SND_SOC_DAPM_SPK("Right Spk", NULL),
+
 	SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0,
 			    platform_clock_control, SND_SOC_DAPM_POST_PMD |
 			    SND_SOC_DAPM_PRE_PMU),
···
 	{ "Headphone Jack", NULL, "HPL" },
 	{ "Headphone Jack", NULL, "HPR" },
 
+	{ "MIC", NULL, "Headset Mic" },
+
+	{ "Headphone Jack", NULL, "Platform Clock" },
+	{ "Headset Mic", NULL, "Platform Clock" },
+
 	{ "Left Spk", NULL, "Left BE_OUT" },
 	{ "Right Spk", NULL, "Right BE_OUT" },
+};
+
+/* For MAX98360A amp */
+static const struct snd_soc_dapm_widget max98360a_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_MIC("Headset Mic", NULL),
+
+	SND_SOC_DAPM_SPK("Spk", NULL),
+
+	SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0,
+			    platform_clock_control, SND_SOC_DAPM_POST_PMD |
+			    SND_SOC_DAPM_PRE_PMU),
+};
+
+static const struct snd_soc_dapm_route max98360a_map[] = {
+	{ "Headphone Jack", NULL, "HPL" },
+	{ "Headphone Jack", NULL, "HPR" },
 
 	{ "MIC", NULL, "Headset Mic" },
 
 	{ "Headphone Jack", NULL, "Platform Clock" },
 	{ "Headset Mic", NULL, "Platform Clock" },
+
+	{"Spk", NULL, "Speaker"},
 };
 
 static struct snd_soc_jack headset;
 
 static int da7219_codec_init(struct snd_soc_pcm_runtime *rtd)
 {
-	struct snd_soc_component *component = rtd->codec_dai->component;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+	struct snd_soc_component *component = codec_dai->component;
 	struct snd_soc_jack *jack;
 	int ret;
 
···
 	int ret, j;
 
 	for (j = 0; j < runtime->num_codecs; j++) {
-		struct snd_soc_dai *codec_dai = runtime->codec_dais[j];
+		struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(runtime, j);
 
 		if (!strcmp(codec_dai->component->name, MAXIM_DEV0_NAME)) {
 			/* vmon_slot_no = 0 imon_slot_no = 1 for TX slots */
···
 static int hdmi_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct card_private *ctx = snd_soc_card_get_drvdata(rtd->card);
-	struct snd_soc_dai *dai = rtd->codec_dai;
+	struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0);
 	struct hdmi_pcm *pcm;
 
 	pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
···
 	DAILINK_COMP_ARRAY(
 	/* Left */	COMP_CODEC(MAXIM_DEV0_NAME, MAX98373_CODEC_DAI),
 	/* Right */	COMP_CODEC(MAXIM_DEV1_NAME, MAX98373_CODEC_DAI)));
+
+SND_SOC_DAILINK_DEF(ssp1_m98360a,
+	DAILINK_COMP_ARRAY(COMP_CODEC("MX98360A:00", "HiFi")));
 
 SND_SOC_DAILINK_DEF(dmic_pin,
 	DAILINK_COMP_ARRAY(COMP_CPU("DMIC01 Pin")));
···
 	.late_probe = card_late_probe,
 };
 
+static struct snd_soc_card card_da7219_m98360a = {
+	.name = "da7219max98360a",
+	.owner = THIS_MODULE,
+	.dai_link = dais,
+	.num_links = ARRAY_SIZE(dais),
+	.controls = m98360a_controls,
+	.num_controls = ARRAY_SIZE(m98360a_controls),
+	.dapm_widgets = max98360a_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(max98360a_widgets),
+	.dapm_routes = max98360a_map,
+	.num_dapm_routes = ARRAY_SIZE(max98360a_map),
+	.fully_routed = true,
+	.late_probe = card_late_probe,
+};
+
 static int audio_probe(struct platform_device *pdev)
 {
 	static struct snd_soc_card *card;
···
 	struct card_private *ctx;
 	int ret;
 
-	ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC);
+	ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL);
 	if (!ctx)
 		return -ENOMEM;
+
+	/* By default dais[0] is configured for max98373 */
+	if (!strcmp(pdev->name, "sof_da7219_max98360a")) {
+		dais[0] = (struct snd_soc_dai_link) {
+			.name = "SSP1-Codec",
+			.id = 0,
+			.no_pcm = 1,
+			.dpcm_playback = 1,
+			.ignore_pmdown_time = 1,
+			SND_SOC_DAILINK_REG(ssp1_pin, ssp1_m98360a, platform) };
+	}
 
 	INIT_LIST_HEAD(&ctx->hdmi_pcm_list);
 	card = (struct snd_soc_card *)pdev->id_entry->driver_data;
 	card->dev = &pdev->dev;
 
-	mach = (&pdev->dev)->platform_data;
+	mach = pdev->dev.platform_data;
 	ret = snd_soc_fixup_dai_links_platform_name(card,
 						    mach->mach_params.platform);
 	if (ret)
···
 		.name = "sof_da7219_max98373",
 		.driver_data = (kernel_ulong_t)&card_da7219_m98373,
 	},
+	{
+		.name = "sof_da7219_max98360a",
+		.driver_data = (kernel_ulong_t)&card_da7219_m98360a,
+	},
 	{ }
 };
 
 static struct platform_driver audio = {
 	.probe = audio_probe,
 	.driver = {
-		.name = "sof_da7219_max98373",
+		.name = "sof_da7219_max98_360a_373",
 		.pm = &snd_soc_pm_ops,
 	},
 	.id_table = board_ids,
···
 MODULE_DESCRIPTION("ASoC Intel(R) SOF Machine driver");
 MODULE_AUTHOR("Yong Zhi <[e-mail]>");
 MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:sof_da7219_max98360a");
 MODULE_ALIAS("platform:sof_da7219_max98373");

sound/soc/intel/boards/sof_maxim_common.c

···
+// SPDX-License-Identifier: GPL-2.0
+//
+// Copyright(c) 2020 Intel Corporation. All rights reserved.
+#include <linux/string.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+#include <sound/soc-dapm.h>
+#include <uapi/sound/asound.h>
+#include "sof_maxim_common.h"
+
+static const struct snd_soc_dapm_route max_98373_dapm_routes[] = {
+	/* speaker */
+	{ "Left Spk", NULL, "Left BE_OUT" },
+	{ "Right Spk", NULL, "Right BE_OUT" },
+};
+
+static struct snd_soc_codec_conf max_98373_codec_conf[] = {
+	{
+		.dlc = COMP_CODEC_CONF(MAX_98373_DEV0_NAME),
+		.name_prefix = "Right",
+	},
+	{
+		.dlc = COMP_CODEC_CONF(MAX_98373_DEV1_NAME),
+		.name_prefix = "Left",
+	},
+};
+
+struct snd_soc_dai_link_component max_98373_components[] = {
+	{  /* For Left */
+		.name = MAX_98373_DEV0_NAME,
+		.dai_name = MAX_98373_CODEC_DAI,
+	},
+	{  /* For Right */
+		.name = MAX_98373_DEV1_NAME,
+		.dai_name = MAX_98373_CODEC_DAI,
+	},
+};
+
+static int max98373_hw_params(struct snd_pcm_substream *substream,
+			      struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai;
+	int j;
+
+	for_each_rtd_codec_dais(rtd, j, codec_dai) {
+		if (!strcmp(codec_dai->component->name, MAX_98373_DEV0_NAME)) {
+			/* DEV0 tdm slot configuration */
+			snd_soc_dai_set_tdm_slot(codec_dai, 0x30, 3, 8, 16);
+		}
+		if (!strcmp(codec_dai->component->name, MAX_98373_DEV1_NAME)) {
+			/* DEV1 tdm slot configuration */
+			snd_soc_dai_set_tdm_slot(codec_dai, 0xC0, 3, 8, 16);
+		}
+	}
+	return 0;
+}
+
+struct snd_soc_ops max_98373_ops = {
+	.hw_params = max98373_hw_params,
+};
+
+int max98373_spk_codec_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_card *card = rtd->card;
+	int ret;
+
+	ret = snd_soc_dapm_add_routes(&card->dapm, max_98373_dapm_routes,
+				      ARRAY_SIZE(max_98373_dapm_routes));
+	if (ret)
+		dev_err(rtd->dev, "Speaker map addition failed: %d\n", ret);
+	return ret;
+}
+
+void sof_max98373_codec_conf(struct snd_soc_card *card)
+{
+	card->codec_conf = max_98373_codec_conf;
+	card->num_configs = ARRAY_SIZE(max_98373_codec_conf);
+}

sound/soc/intel/boards/sof_maxim_common.h

···
+/* SPDX-License-Identifier: GPL-2.0 */
+/*
+ * Copyright(c) 2020 Intel Corporation.
+ */
+
+/*
+ * This file defines data structures used in Machine Driver for Intel
+ * platforms with Maxim Codecs.
+ */
+#ifndef __SOF_MAXIM_COMMON_H
+#define __SOF_MAXIM_COMMON_H
+
+#include <sound/soc.h>
+
+#define MAX_98373_CODEC_DAI	"max98373-aif1"
+#define MAX_98373_DEV0_NAME	"i2c-MX98373:00"
+#define MAX_98373_DEV1_NAME	"i2c-MX98373:01"
+
+extern struct snd_soc_dai_link_component max_98373_components[2];
+extern struct snd_soc_ops max_98373_ops;
+
+int max98373_spk_codec_init(struct snd_soc_pcm_runtime *rtd);
+void sof_max98373_codec_conf(struct snd_soc_card *card);
+#endif /* __SOF_MAXIM_COMMON_H */

sound/soc/intel/boards/sof_pcm512x.c

···
+// SPDX-License-Identifier: GPL-2.0
+// Copyright(c) 2018-2020 Intel Corporation.
+
+/*
+ * Intel SOF Machine Driver for Intel platforms with TI PCM512x codec,
+ * e.g. Up or Up2 with Hifiberry DAC+ HAT
+ */
+#include <linux/clk.h>
+#include <linux/dmi.h>
+#include <linux/i2c.h>
+#include <linux/input.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/types.h>
+#include <sound/core.h>
+#include <sound/jack.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-acpi.h>
+#include "../../codecs/pcm512x.h"
+#include "../common/soc-intel-quirks.h"
+#include "hda_dsp_common.h"
+
+#define NAME_SIZE 32
+
+#define SOF_PCM512X_SSP_CODEC(quirk)		((quirk) & GENMASK(3, 0))
+#define SOF_PCM512X_SSP_CODEC_MASK			(GENMASK(3, 0))
+
+#define IDISP_CODEC_MASK	0x4
+
+/* Default: SSP5 */
+static unsigned long sof_pcm512x_quirk = SOF_PCM512X_SSP_CODEC(5);
+
+static bool is_legacy_cpu;
+
+struct sof_hdmi_pcm {
+	struct list_head head;
+	struct snd_soc_dai *codec_dai;
+	int device;
+};
+
+struct sof_card_private {
+	struct list_head hdmi_pcm_list;
+	bool idisp_codec;
+};
+
+static int sof_pcm512x_quirk_cb(const struct dmi_system_id *id)
+{
+	sof_pcm512x_quirk = (unsigned long)id->driver_data;
+	return 1;
+}
+
+static const struct dmi_system_id sof_pcm512x_quirk_table[] = {
+	{
+		.callback = sof_pcm512x_quirk_cb,
+		.matches = {
+			DMI_MATCH(DMI_SYS_VENDOR, "AAEON"),
+			DMI_MATCH(DMI_PRODUCT_NAME, "UP-CHT01"),
+		},
+		.driver_data = (void *)(SOF_PCM512X_SSP_CODEC(2)),
+	},
+	{}
+};
+
+static int sof_hdmi_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct sof_card_private *ctx = snd_soc_card_get_drvdata(rtd->card);
+	struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0);
+	struct sof_hdmi_pcm *pcm;
+
+	pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
+	if (!pcm)
+		return -ENOMEM;
+
+	/* dai_link id is 1:1 mapped to the PCM device */
+	pcm->device = rtd->dai_link->id;
+	pcm->codec_dai = dai;
+
+	list_add_tail(&pcm->head, &ctx->hdmi_pcm_list);
+
+	return 0;
+}
+
+static int sof_pcm512x_codec_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_component *codec = asoc_rtd_to_codec(rtd, 0)->component;
+
+	snd_soc_component_update_bits(codec, PCM512x_GPIO_EN, 0x08, 0x08);
+	snd_soc_component_update_bits(codec, PCM512x_GPIO_OUTPUT_4, 0x0f, 0x02);
+	snd_soc_component_update_bits(codec, PCM512x_GPIO_CONTROL_1,
+				      0x08, 0x08);
+
+	return 0;
+}
+
+static int aif1_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_component *codec = asoc_rtd_to_codec(rtd, 0)->component;
+
+	snd_soc_component_update_bits(codec, PCM512x_GPIO_CONTROL_1,
+				      0x08, 0x08);
+
+	return 0;
+}
+
+static void aif1_shutdown(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_component *codec = asoc_rtd_to_codec(rtd, 0)->component;
+
+	snd_soc_component_update_bits(codec, PCM512x_GPIO_CONTROL_1,
+				      0x08, 0x00);
+}
+
+static const struct snd_soc_ops sof_pcm512x_ops = {
+	.startup = aif1_startup,
+	.shutdown = aif1_shutdown,
+};
+
+static struct snd_soc_dai_link_component platform_component[] = {
+	{
+		/* name might be overridden during probe */
+		.name = "0000:00:1f.3"
+	}
+};
+
+#if IS_ENABLED(CONFIG_SND_HDA_CODEC_HDMI)
+static int sof_card_late_probe(struct snd_soc_card *card)
+{
+	struct sof_card_private *ctx = snd_soc_card_get_drvdata(card);
+	struct sof_hdmi_pcm *pcm;
+
+	/* HDMI is not supported by SOF on Baytrail/CherryTrail */
+	if (is_legacy_cpu)
+		return 0;
+
+	if (list_empty(&ctx->hdmi_pcm_list))
+		return -EINVAL;
+
+	if (!ctx->idisp_codec)
+		return 0;
+
+	pcm = list_first_entry(&ctx->hdmi_pcm_list, struct sof_hdmi_pcm, head);
+
+	return hda_dsp_hdmi_build_controls(card, pcm->codec_dai->component);
+}
+#else
+static int sof_card_late_probe(struct snd_soc_card *card)
+{
+	return 0;
+}
+#endif
+
+static const struct snd_kcontrol_new sof_controls[] = {
+	SOC_DAPM_PIN_SWITCH("Ext Spk"),
+};
+
+static const struct snd_soc_dapm_widget sof_widgets[] = {
+	SND_SOC_DAPM_SPK("Ext Spk", NULL),
+};
+
+static const struct snd_soc_dapm_widget dmic_widgets[] = {
+	SND_SOC_DAPM_MIC("SoC DMIC", NULL),
+};
+
+static const struct snd_soc_dapm_route sof_map[] = {
+	/* Speaker */
+	{"Ext Spk", NULL, "OUTR"},
+	{"Ext Spk", NULL, "OUTL"},
+};
+
+static const struct snd_soc_dapm_route dmic_map[] = {
+	/* digital mics */
+	{"DMic", NULL, "SoC DMIC"},
+};
+
+static int dmic_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_card *card = rtd->card;
+	int ret;
+
+	ret = snd_soc_dapm_new_controls(&card->dapm, dmic_widgets,
+					ARRAY_SIZE(dmic_widgets));
+	if (ret) {
+		dev_err(card->dev, "DMic widget addition failed: %d\n", ret);
+		/* Don't need to add routes if widget addition failed */
+		return ret;
+	}
+
+	ret = snd_soc_dapm_add_routes(&card->dapm, dmic_map,
+				      ARRAY_SIZE(dmic_map));
+
+	if (ret)
+		dev_err(card->dev, "DMic map addition failed: %d\n", ret);
+
+	return ret;
+}
+
+/* sof audio machine driver for pcm512x codec */
+static struct snd_soc_card sof_audio_card_pcm512x = {
+	.name = "pcm512x",
+	.owner = THIS_MODULE,
+	.controls = sof_controls,
+	.num_controls = ARRAY_SIZE(sof_controls),
+	.dapm_widgets = sof_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(sof_widgets),
+	.dapm_routes = sof_map,
+	.num_dapm_routes = ARRAY_SIZE(sof_map),
+	.fully_routed = true,
+	.late_probe = sof_card_late_probe,
+};
+
+SND_SOC_DAILINK_DEF(pcm512x_component,
+	DAILINK_COMP_ARRAY(COMP_CODEC("i2c-104C5122:00", "pcm512x-hifi")));
+SND_SOC_DAILINK_DEF(dmic_component,
+	DAILINK_COMP_ARRAY(COMP_CODEC("dmic-codec", "dmic-hifi")));
+
+static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev,
+							  int ssp_codec,
+							  int dmic_be_num,
+							  int hdmi_num,
+							  bool idisp_codec)
+{
+	struct snd_soc_dai_link_component *idisp_components;
+	struct snd_soc_dai_link_component *cpus;
+	struct snd_soc_dai_link *links;
+	int i, id = 0;
+
+	links = devm_kcalloc(dev, sof_audio_card_pcm512x.num_links,
+			sizeof(struct snd_soc_dai_link), GFP_KERNEL);
+	cpus = devm_kcalloc(dev, sof_audio_card_pcm512x.num_links,
+			sizeof(struct snd_soc_dai_link_component), GFP_KERNEL);
+	if (!links || !cpus)
+		goto devm_err;
+
+	/* codec SSP */
+	links[id].name = devm_kasprintf(dev, GFP_KERNEL,
+					"SSP%d-Codec", ssp_codec);
+	if (!links[id].name)
+		goto devm_err;
+
+	links[id].id = id;
+	links[id].codecs = pcm512x_component;
+	links[id].num_codecs = ARRAY_SIZE(pcm512x_component);
+	links[id].platforms = platform_component;
+	links[id].num_platforms = ARRAY_SIZE(platform_component);
+	links[id].init = sof_pcm512x_codec_init;
+	links[id].ops = &sof_pcm512x_ops;
+	links[id].nonatomic = true;
+	links[id].dpcm_playback = 1;
+	/*
+	 * capture only supported with specific versions of the Hifiberry DAC+
+	 * links[id].dpcm_capture = 1;
+	 */
+	links[id].no_pcm = 1;
+	links[id].cpus = &cpus[id];
+	links[id].num_cpus = 1;
+	if (is_legacy_cpu) {
+		links[id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL,
+							  "ssp%d-port",
+							  ssp_codec);
+		if (!links[id].cpus->dai_name)
+			goto devm_err;
+	} else {
+		links[id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL,
+							  "SSP%d Pin",
+							  ssp_codec);
+		if (!links[id].cpus->dai_name)
+			goto devm_err;
+	}
+	id++;
+
+	/* dmic */
+	if (dmic_be_num > 0) {
+		/* at least we have dmic01 */
+		links[id].name = "dmic01";
+		links[id].cpus = &cpus[id];
+		links[id].cpus->dai_name = "DMIC01 Pin";
+		links[id].init = dmic_init;
+		if (dmic_be_num > 1) {
+			/* set up 2 BE links at most */
+			links[id + 1].name = "dmic16k";
+			links[id + 1].cpus = &cpus[id + 1];
+			links[id + 1].cpus->dai_name = "DMIC16k Pin";
+			dmic_be_num = 2;
+		}
+	}
+
+	for (i = 0; i < dmic_be_num; i++) {
+		links[id].id = id;
+		links[id].num_cpus = 1;
+		links[id].codecs = dmic_component;
+		links[id].num_codecs = ARRAY_SIZE(dmic_component);
+		links[id].platforms = platform_component;
+		links[id].num_platforms = ARRAY_SIZE(platform_component);
+		links[id].ignore_suspend = 1;
+		links[id].dpcm_capture = 1;
+		links[id].no_pcm = 1;
+		id++;
+	}
+
+	/* HDMI */
+	if (hdmi_num > 0) {
+		idisp_components = devm_kcalloc(dev, hdmi_num,
+				sizeof(struct snd_soc_dai_link_component),
+				GFP_KERNEL);
+		if (!idisp_components)
+			goto devm_err;
+	}
+	for (i = 1; i <= hdmi_num; i++) {
+		links[id].name = devm_kasprintf(dev, GFP_KERNEL,
+						"iDisp%d", i);
+		if (!links[id].name)
+			goto devm_err;
+
+		links[id].id = id;
+		links[id].cpus = &cpus[id];
+		links[id].num_cpus = 1;
+		links[id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL,
+							  "iDisp%d Pin", i);
+		if (!links[id].cpus->dai_name)
+			goto devm_err;
+
+		/*
+		 * topology cannot be loaded if codec is missing, so
+		 * use the dummy codec if needed
+		 */
+		if (idisp_codec) {
+			idisp_components[i - 1].name = "ehdaudio0D2";
+			idisp_components[i - 1].dai_name =
+				devm_kasprintf(dev, GFP_KERNEL,
+					       "intel-hdmi-hifi%d", i);
+		} else {
+			idisp_components[i - 1].name = "snd-soc-dummy";
+			idisp_components[i - 1].dai_name = "snd-soc-dummy-dai";
+		}
+		if (!idisp_components[i - 1].dai_name)
+			goto devm_err;
+
+		links[id].codecs = &idisp_components[i - 1];
+		links[id].num_codecs = 1;
+		links[id].platforms = platform_component;
+		links[id].num_platforms = ARRAY_SIZE(platform_component);
+		links[id].init = sof_hdmi_init;
+		links[id].dpcm_playback = 1;
+		links[id].no_pcm = 1;
+		id++;
+	}
+
+	return links;
+devm_err:
+	return NULL;
+}
+
+static int sof_audio_probe(struct platform_device *pdev)
+{
+	struct snd_soc_acpi_mach *mach = pdev->dev.platform_data;
+	struct snd_soc_dai_link *dai_links;
+	struct sof_card_private *ctx;
+	int dmic_be_num, hdmi_num;
+	int ret, ssp_codec;
+
+	ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL);
+	if (!ctx)
+		return -ENOMEM;
+
+	hdmi_num = 0;
+	if (soc_intel_is_byt() || soc_intel_is_cht()) {
+		is_legacy_cpu = true;
+		dmic_be_num = 0;
+		/* default quirk for legacy cpu */
+		sof_pcm512x_quirk = SOF_PCM512X_SSP_CODEC(2);
+	} else {
+		dmic_be_num = 2;
+#if IS_ENABLED(CONFIG_SND_HDA_CODEC_HDMI)
+		if (mach->mach_params.common_hdmi_codec_drv &&
+		    (mach->mach_params.codec_mask & IDISP_CODEC_MASK))
+			ctx->idisp_codec = true;
+
+		/* links are always present in topology */
+		hdmi_num = 3;
+#endif
+	}
+
+	dmi_check_system(sof_pcm512x_quirk_table);
+
+	dev_dbg(&pdev->dev, "sof_pcm512x_quirk = %lx\n", sof_pcm512x_quirk);
+
+	ssp_codec = sof_pcm512x_quirk & SOF_PCM512X_SSP_CODEC_MASK;
+
+	/* compute number of dai links */
+	sof_audio_card_pcm512x.num_links = 1 + dmic_be_num + hdmi_num;
+
+	dai_links = sof_card_dai_links_create(&pdev->dev, ssp_codec,
+					      dmic_be_num, hdmi_num,
+					      ctx->idisp_codec);
+	if (!dai_links)
+		return -ENOMEM;
+
+	sof_audio_card_pcm512x.dai_link = dai_links;
+
+	INIT_LIST_HEAD(&ctx->hdmi_pcm_list);
+
+	sof_audio_card_pcm512x.dev = &pdev->dev;
+
+	/* set platform name for each dailink */
+	ret = snd_soc_fixup_dai_links_platform_name(&sof_audio_card_pcm512x,
+						    mach->mach_params.platform);
+	if (ret)
+		return ret;
+
+	snd_soc_card_set_drvdata(&sof_audio_card_pcm512x, ctx);
+
+	return devm_snd_soc_register_card(&pdev->dev,
+					  &sof_audio_card_pcm512x);
+}
+
+static int sof_pcm512x_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+	struct snd_soc_component *component = NULL;
+
+	for_each_card_components(card, component) {
+		if (!strcmp(component->name, pcm512x_component[0].name)) {
+			snd_soc_component_set_jack(component, NULL, NULL);
+			break;
+		}
+	}
+
+	return 0;
+}
+
+static struct platform_driver sof_audio = {
+	.probe = sof_audio_probe,
+	.remove = sof_pcm512x_remove,
+	.driver = {
+		.name = "sof_pcm512x",
+		.pm = &snd_soc_pm_ops,
+	},
+};
+module_platform_driver(sof_audio)
+
+MODULE_DESCRIPTION("ASoC Intel(R) SOF + PCM512x Machine driver");
+MODULE_AUTHOR("Pierre-Louis Bossart");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:sof_pcm512x");

sound/soc/intel/boards/sof_rt5682.c

···
 // SPDX-License-Identifier: GPL-2.0
-// Copyright(c) 2019 Intel Corporation.
+// Copyright(c) 2019-2020 Intel Corporation.
 
 /*
  * Intel SOF Machine Driver with Realtek rt5682 Codec
- * and speaker codec MAX98357A
+ * and speaker codec MAX98357A or RT1015.
  */
 #include <linux/i2c.h>
 #include <linux/input.h>
···
 #include <sound/soc.h>
 #include <sound/rt5682.h>
 #include <sound/soc-acpi.h>
+#include "../../codecs/rt1015.h"
 #include "../../codecs/rt5682.h"
 #include "../../codecs/hdac_hdmi.h"
 #include "../common/soc-intel-quirks.h"
 #include "hda_dsp_common.h"
+#include "sof_maxim_common.h"
 
 #define NAME_SIZE 32
 
···
 #define SOF_RT5682_NUM_HDMIDEV_MASK		(GENMASK(12, 10))
 #define SOF_RT5682_NUM_HDMIDEV(quirk)	\
 	((quirk << SOF_RT5682_NUM_HDMIDEV_SHIFT) & SOF_RT5682_NUM_HDMIDEV_MASK)
+#define SOF_RT1015_SPEAKER_AMP_PRESENT		BIT(13)
+#define SOF_MAX98373_SPEAKER_AMP_PRESENT	BIT(14)
 
 /* Default: MCLK on, MCLK 19.2M, SSP0  */
 static unsigned long sof_rt5682_quirk = SOF_RT5682_MCLK_EN |
···
 static int sof_hdmi_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct sof_card_private *ctx = snd_soc_card_get_drvdata(rtd->card);
-	struct snd_soc_dai *dai = rtd->codec_dai;
+	struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0);
 	struct sof_hdmi_pcm *pcm;
 
 	pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
···
 static int sof_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct sof_card_private *ctx = snd_soc_card_get_drvdata(rtd->card);
-	struct snd_soc_component *component = rtd->codec_dai->component;
+	struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
 	struct snd_soc_jack *jack;
 	int ret;
 
···
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct sof_card_private *ctx = snd_soc_card_get_drvdata(rtd->card);
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	int clk_id, clk_freq, pll_out, ret;
 
 	if (sof_rt5682_quirk & SOF_RT5682_MCLK_EN) {
···
 
 static struct snd_soc_ops sof_rt5682_ops = {
 	.hw_params = sof_rt5682_hw_params,
+};
+
+static int sof_rt1015_hw_params(struct snd_pcm_substream *substream,
+				struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_card *card = rtd->card;
+	struct snd_soc_dai *codec_dai;
+	int i, ret;
+
+	if (!snd_soc_card_get_codec_dai(card, "rt1015-aif"))
+		return 0;
+
+	for_each_rtd_codec_dais(rtd, i, codec_dai) {
+		ret = snd_soc_dai_set_pll(codec_dai, 0, RT1015_PLL_S_BCLK,
+					  params_rate(params) * 50,
+					  params_rate(params) * 256);
+		if (ret < 0) {
+			dev_err(card->dev, "failed to set pll\n");
+			return ret;
+		}
+		/* Configure sysclk for codec */
+		ret = snd_soc_dai_set_sysclk(codec_dai, RT1015_SCLK_S_PLL,
+					     params_rate(params) * 256,
+					     SND_SOC_CLOCK_IN);
+		if (ret < 0) {
+			dev_err(card->dev, "failed to set sysclk\n");
+			return ret;
+		}
+	}
+
+	return 0;
+}
+
+static struct snd_soc_ops sof_rt1015_ops = {
+	.hw_params = sof_rt1015_hw_params,
 };
 
 static struct snd_soc_dai_link_component platform_component[] = {
···
 	SOC_DAPM_PIN_SWITCH("Headphone Jack"),
 	SOC_DAPM_PIN_SWITCH("Headset Mic"),
 	SOC_DAPM_PIN_SWITCH("Spk"),
+	SOC_DAPM_PIN_SWITCH("Left Spk"),
+	SOC_DAPM_PIN_SWITCH("Right Spk"),
+
 };
 
 static const struct snd_soc_dapm_widget sof_widgets[] = {
 	SND_SOC_DAPM_HP("Headphone Jack", NULL),
 	SND_SOC_DAPM_MIC("Headset Mic", NULL),
 	SND_SOC_DAPM_SPK("Spk", NULL),
+	SND_SOC_DAPM_SPK("Left Spk", NULL),
+	SND_SOC_DAPM_SPK("Right Spk", NULL),
 };
 
 static const struct snd_soc_dapm_widget dmic_widgets[] = {
···
 	{ "Spk", NULL, "Speaker" },
 };
 
+static const struct snd_soc_dapm_route speaker_map_lr[] = {
+	{ "Left Spk", NULL, "Left SPO" },
+	{ "Right Spk", NULL, "Right SPO" },
+};
+
 static const struct snd_soc_dapm_route dmic_map[] = {
 	/* digital mics */
 	{"DMic", NULL, "SoC DMIC"},
 };
+
+static int speaker_codec_init_lr(struct snd_soc_pcm_runtime *rtd)
+{
+	return snd_soc_dapm_add_routes(&rtd->card->dapm, speaker_map_lr,
+				       ARRAY_SIZE(speaker_map_lr));
+}
 
 static int speaker_codec_init(struct snd_soc_pcm_runtime *rtd)
 {
···
 	return ret;
 }
 
+static struct snd_soc_codec_conf rt1015_amp_conf[] = {
+	{
+		.dlc = COMP_CODEC_CONF("i2c-10EC1015:00"),
+		.name_prefix = "Left",
+	},
+	{
+		.dlc = COMP_CODEC_CONF("i2c-10EC1015:01"),
+		.name_prefix = "Right",
+	},
+};
+
 /* sof audio machine driver for rt5682 codec */
 static struct snd_soc_card sof_audio_card_rt5682 = {
 	.name = "rt5682", /* the sof- prefix is added by the core */
···
 		.name = "MX98357A:00",
 		.dai_name = "HiFi",
 	}
+};
+
+static struct snd_soc_dai_link_component rt1015_components[] = {
+	{
+		.name = "i2c-10EC1015:00",
+		.dai_name = "rt1015-aif",
+	},
+	{
+		.name = "i2c-10EC1015:01",
+		.dai_name = "rt1015-aif",
+	},
 };
 
 static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev,
···
 			goto devm_err;
 
 		links[id].id = id;
-		links[id].codecs = max98357a_component;
-		links[id].num_codecs = ARRAY_SIZE(max98357a_component);
+		if (sof_rt5682_quirk & SOF_RT1015_SPEAKER_AMP_PRESENT) {
+			links[id].codecs = rt1015_components;
+			links[id].num_codecs = ARRAY_SIZE(rt1015_components);
+			links[id].init = speaker_codec_init_lr;
+			links[id].ops = &sof_rt1015_ops;
+		} else if (sof_rt5682_quirk &
+				SOF_MAX98373_SPEAKER_AMP_PRESENT) {
+			links[id].codecs = max_98373_components;
+			links[id].num_codecs = ARRAY_SIZE(max_98373_components);
+			links[id].init = max98373_spk_codec_init;
+			links[id].ops = &max_98373_ops;
+		} else {
+			links[id].codecs = max98357a_component;
+			links[id].num_codecs = ARRAY_SIZE(max98357a_component);
+			links[id].init = speaker_codec_init;
+		}
 		links[id].platforms = platform_component;
 		links[id].num_platforms = ARRAY_SIZE(platform_component);
-		links[id].init = speaker_codec_init,
 		links[id].nonatomic = true;
 		links[id].dpcm_playback = 1;
 		links[id].no_pcm = 1;
···
 
 	dmi_check_system(sof_rt5682_quirk_table);
 
-	mach = (&pdev->dev)->platform_data;
+	mach = pdev->dev.platform_data;
 
 	/* A speaker amp might not be present when the quirk claims one is.
 	 * Detect this via whether the machine driver match includes quirk_data.
···
 	if (sof_rt5682_quirk & SOF_SPEAKER_AMP_PRESENT)
 		sof_audio_card_rt5682.num_links++;
 
+	if (sof_rt5682_quirk & SOF_MAX98373_SPEAKER_AMP_PRESENT)
+		sof_max98373_codec_conf(&sof_audio_card_rt5682);
+
 	dai_links = sof_card_dai_links_create(&pdev->dev, ssp_codec, ssp_amp,
 					      dmic_be_num, hdmi_num);
 	if (!dai_links)
 		return -ENOMEM;
 
 	sof_audio_card_rt5682.dai_link = dai_links;
+
+	if (sof_rt5682_quirk & SOF_RT1015_SPEAKER_AMP_PRESENT) {
+		sof_audio_card_rt5682.codec_conf = rt1015_amp_conf;
+		sof_audio_card_rt5682.num_configs = ARRAY_SIZE(rt1015_amp_conf);
+	}
 
 	INIT_LIST_HEAD(&ctx->hdmi_pcm_list);
 
···
 					SOF_RT5682_SSP_AMP(1) |
 					SOF_RT5682_NUM_HDMIDEV(4)),
 	},
+	{
+		.name = "jsl_rt5682_rt1015",
+		.driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN |
+					SOF_RT5682_MCLK_24MHZ |
+					SOF_RT5682_SSP_CODEC(0) |
+					SOF_SPEAKER_AMP_PRESENT |
+					SOF_RT1015_SPEAKER_AMP_PRESENT |
+					SOF_RT5682_SSP_AMP(1)),
+	},
+	{
+		.name = "tgl_max98373_rt5682",
+		.driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN |
+					SOF_RT5682_SSP_CODEC(0) |
+					SOF_SPEAKER_AMP_PRESENT |
+					SOF_MAX98373_SPEAKER_AMP_PRESENT |
+					SOF_RT5682_SSP_AMP(1) |
+					SOF_RT5682_NUM_HDMIDEV(4)),
+	},
 	{ }
 };
 
···
 MODULE_LICENSE("GPL v2");
 MODULE_ALIAS("platform:sof_rt5682");
 MODULE_ALIAS("platform:tgl_max98357a_rt5682");
+MODULE_ALIAS("platform:jsl_rt5682_rt1015");
+MODULE_ALIAS("platform:tgl_max98373_rt5682");

sound/soc/intel/boards/sof_sdw.c

···
+// SPDX-License-Identifier: GPL-2.0
+// Copyright (c) 2020 Intel Corporation
+
+/*
+ *  sof_sdw - ASOC Machine driver for Intel SoundWire platforms
+ */
+
+#include <linux/device.h>
+#include <linux/dmi.h>
+#include <linux/module.h>
+#include <linux/soundwire/sdw.h>
+#include <linux/soundwire/sdw_type.h>
+#include <sound/soc.h>
+#include <sound/soc-acpi.h>
+#include "sof_sdw_common.h"
+
+unsigned long sof_sdw_quirk = SOF_RT711_JD_SRC_JD1;
+
+#define INC_ID(BE, CPU, LINK)	do { (BE)++; (CPU)++; (LINK)++; } while (0)
+
+static int sof_sdw_quirk_cb(const struct dmi_system_id *id)
+{
+	sof_sdw_quirk = (unsigned long)id->driver_data;
+	return 1;
+}
+
+static const struct dmi_system_id sof_sdw_quirk_table[] = {
+	{
+		.callback = sof_sdw_quirk_cb,
+		.matches = {
+			DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"),
+			DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "09C6")
+		},
+		.driver_data = (void *)(SOF_RT711_JD_SRC_JD2 |
+					SOF_RT715_DAI_ID_FIX),
+	},
+	{
+		/* early version of SKU 09C6 */
+		.callback = sof_sdw_quirk_cb,
+		.matches = {
+			DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"),
+			DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0983")
+		},
+		.driver_data = (void *)(SOF_RT711_JD_SRC_JD2 |
+					SOF_RT715_DAI_ID_FIX),
+	},
+	{
+		.callback = sof_sdw_quirk_cb,
+		.matches = {
+			DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"),
+			DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "098F"),
+		},
+		.driver_data = (void *)(SOF_RT711_JD_SRC_JD2 |
+					SOF_RT715_DAI_ID_FIX |
+					SOF_SDW_FOUR_SPK),
+	},
+		{
+		.callback = sof_sdw_quirk_cb,
+		.matches = {
+			DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"),
+			DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0990"),
+		},
+		.driver_data = (void *)(SOF_RT711_JD_SRC_JD2 |
+					SOF_RT715_DAI_ID_FIX |
+					SOF_SDW_FOUR_SPK),
+	},
+	{
+		.callback = sof_sdw_quirk_cb,
+		.matches = {
+			DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"),
+			DMI_MATCH(DMI_PRODUCT_NAME,
+				  "Tiger Lake Client Platform"),
+		},
+		.driver_data = (void *)(SOF_RT711_JD_SRC_JD1 |
+				SOF_SDW_TGL_HDMI | SOF_SDW_PCH_DMIC |
+				SOF_SSP_PORT(SOF_I2S_SSP2)),
+	},
+	{
+		.callback = sof_sdw_quirk_cb,
+		.matches = {
+			DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"),
+			DMI_MATCH(DMI_PRODUCT_NAME, "Ice Lake Client"),
+		},
+		.driver_data = (void *)SOF_SDW_PCH_DMIC,
+	},
+	{
+		.callback = sof_sdw_quirk_cb,
+		.matches = {
+			DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"),
+			DMI_MATCH(DMI_PRODUCT_NAME, "CometLake Client"),
+		},
+		.driver_data = (void *)SOF_SDW_PCH_DMIC,
+	},
+	{
+		.callback = sof_sdw_quirk_cb,
+		.matches = {
+			DMI_MATCH(DMI_SYS_VENDOR, "Google"),
+			DMI_MATCH(DMI_PRODUCT_NAME, "Volteer"),
+		},
+		.driver_data = (void *)(SOF_SDW_TGL_HDMI | SOF_SDW_PCH_DMIC),
+	},
+
+	{}
+};
+
+static struct snd_soc_codec_conf codec_conf[] = {
+	{
+		.dlc = COMP_CODEC_CONF("sdw:0:25d:711:0"),
+		.name_prefix = "rt711",
+	},
+	/* rt1308 w/ I2S connection */
+	{
+		.dlc = COMP_CODEC_CONF("i2c-10EC1308:00"),
+		.name_prefix = "rt1308-1",
+	},
+	/* rt1308 left on link 1 */
+	{
+		.dlc = COMP_CODEC_CONF("sdw:1:25d:1308:0"),
+		.name_prefix = "rt1308-1",
+	},
+	/* two 1308s on link1 with different unique id */
+	{
+		.dlc = COMP_CODEC_CONF("sdw:1:25d:1308:0:0"),
+		.name_prefix = "rt1308-1",
+	},
+	{
+		.dlc = COMP_CODEC_CONF("sdw:1:25d:1308:0:2"),
+		.name_prefix = "rt1308-2",
+	},
+	/* rt1308 right on link 2 */
+	{
+		.dlc = COMP_CODEC_CONF("sdw:2:25d:1308:0"),
+		.name_prefix = "rt1308-2",
+	},
+	{
+		.dlc = COMP_CODEC_CONF("sdw:3:25d:715:0"),
+		.name_prefix = "rt715",
+	},
+	{
+		.dlc = COMP_CODEC_CONF("sdw:0:25d:5682:0"),
+		.name_prefix = "rt5682",
+	},
+};
+
+static struct snd_soc_dai_link_component dmic_component[] = {
+	{
+		.name = "dmic-codec",
+		.dai_name = "dmic-hifi",
+	}
+};
+
+static struct snd_soc_dai_link_component platform_component[] = {
+	{
+		/* name might be overridden during probe */
+		.name = "0000:00:1f.3"
+	}
+};
+
+/* these wrappers are only needed to avoid typecast compilation errors */
+static int sdw_startup(struct snd_pcm_substream *substream)
+{
+	return sdw_startup_stream(substream);
+}
+
+static void sdw_shutdown(struct snd_pcm_substream *substream)
+{
+	sdw_shutdown_stream(substream);
+}
+
+static const struct snd_soc_ops sdw_ops = {
+	.startup = sdw_startup,
+	.shutdown = sdw_shutdown,
+};
+
+static struct sof_sdw_codec_info codec_info_list[] = {
+	{
+		.id = 0x700,
+		.direction = {true, true},
+		.dai_name = "rt700-aif1",
+		.init = sof_sdw_rt700_init,
+	},
+	{
+		.id = 0x711,
+		.direction = {true, true},
+		.dai_name = "rt711-aif1",
+		.init = sof_sdw_rt711_init,
+	},
+	{
+		.id = 0x1308,
+		.acpi_id = "10EC1308",
+		.direction = {true, false},
+		.dai_name = "rt1308-aif",
+		.ops = &sof_sdw_rt1308_i2s_ops,
+		.init = sof_sdw_rt1308_init,
+	},
+	{
+		.id = 0x715,
+		.direction = {false, true},
+		.dai_name = "rt715-aif2",
+		.init = sof_sdw_rt715_init,
+	},
+	{
+		.id = 0x5682,
+		.direction = {true, true},
+		.dai_name = "rt5682-sdw",
+		.init = sof_sdw_rt5682_init,
+	},
+};
+
+static inline int find_codec_info_part(unsigned int part_id)
+{
+	int i;
+
+	for (i = 0; i < ARRAY_SIZE(codec_info_list); i++)
+		if (part_id == codec_info_list[i].id)
+			break;
+
+	if (i == ARRAY_SIZE(codec_info_list))
+		return -EINVAL;
+
+	return i;
+}
+
+static inline int find_codec_info_acpi(const u8 *acpi_id)
+{
+	int i;
+
+	if (!acpi_id[0])
+		return -EINVAL;
+
+	for (i = 0; i < ARRAY_SIZE(codec_info_list); i++)
+		if (!memcmp(codec_info_list[i].acpi_id, acpi_id,
+			    ACPI_ID_LEN))
+			break;
+
+	if (i == ARRAY_SIZE(codec_info_list))
+		return -EINVAL;
+
+	return i;
+}
+
+/*
+ * get BE dailink number and CPU DAI number based on sdw link adr.
+ * Since some sdw slaves may be aggregated, the CPU DAI number
+ * may be larger than the number of BE dailinks.
+ */
+static int get_sdw_dailink_info(const struct snd_soc_acpi_link_adr *links,
+				int *sdw_be_num, int *sdw_cpu_dai_num)
+{
+	const struct snd_soc_acpi_link_adr *link;
+	bool group_visited[SDW_MAX_GROUPS];
+	bool no_aggregation;
+	int i;
+
+	no_aggregation = sof_sdw_quirk & SOF_SDW_NO_AGGREGATION;
+	*sdw_cpu_dai_num = 0;
+	*sdw_be_num  = 0;
+
+	if (!links)
+		return -EINVAL;
+
+	for (i = 0; i < SDW_MAX_GROUPS; i++)
+		group_visited[i] = false;
+
+	for (link = links; link->num_adr; link++) {
+		const struct snd_soc_acpi_endpoint *endpoint;
+		int part_id, codec_index;
+		int stream;
+		u64 adr;
+
+		adr = link->adr_d->adr;
+		part_id = SDW_PART_ID(adr);
+		codec_index = find_codec_info_part(part_id);
+		if (codec_index < 0)
+			return codec_index;
+
+		endpoint = link->adr_d->endpoints;
+
+		/* count DAI number for playback and capture */
+		for_each_pcm_streams(stream) {
+			if (!codec_info_list[codec_index].direction[stream])
+				continue;
+
+			(*sdw_cpu_dai_num)++;
+
+			/* count BE for each non-aggregated slave or group */
+			if (!endpoint->aggregated || no_aggregation ||
+			    !group_visited[endpoint->group_id])
+				(*sdw_be_num)++;
+		}
+
+		if (endpoint->aggregated)
+			group_visited[endpoint->group_id] = true;
+	}
+
+	return 0;
+}
+
+static void init_dai_link(struct snd_soc_dai_link *dai_links, int be_id,
+			  char *name, int playback, int capture,
+			  struct snd_soc_dai_link_component *cpus,
+			  int cpus_num,
+			  struct snd_soc_dai_link_component *codecs,
+			  int codecs_num,
+			  int (*init)(struct snd_soc_pcm_runtime *rtd),
+			  const struct snd_soc_ops *ops)
+{
+	dai_links->id = be_id;
+	dai_links->name = name;
+	dai_links->platforms = platform_component;
+	dai_links->num_platforms = ARRAY_SIZE(platform_component);
+	dai_links->nonatomic = true;
+	dai_links->no_pcm = 1;
+	dai_links->cpus = cpus;
+	dai_links->num_cpus = cpus_num;
+	dai_links->codecs = codecs;
+	dai_links->num_codecs = codecs_num;
+	dai_links->dpcm_playback = playback;
+	dai_links->dpcm_capture = capture;
+	dai_links->init = init;
+	dai_links->ops = ops;
+}
+
+static bool is_unique_device(const struct snd_soc_acpi_link_adr *link,
+			     unsigned int sdw_version,
+			     unsigned int mfg_id,
+			     unsigned int part_id,
+			     unsigned int class_id,
+			     int index_in_link
+			    )
+{
+	int i;
+
+	for (i = 0; i < link->num_adr; i++) {
+		unsigned int sdw1_version, mfg1_id, part1_id, class1_id;
+		u64 adr;
+
+		/* skip itself */
+		if (i == index_in_link)
+			continue;
+
+		adr = link->adr_d[i].adr;
+
+		sdw1_version = SDW_VERSION(adr);
+		mfg1_id = SDW_MFG_ID(adr);
+		part1_id = SDW_PART_ID(adr);
+		class1_id = SDW_CLASS_ID(adr);
+
+		if (sdw_version == sdw1_version &&
+		    mfg_id == mfg1_id &&
+		    part_id == part1_id &&
+		    class_id == class1_id)
+			return false;
+	}
+
+	return true;
+}
+
+static int create_codec_dai_name(struct device *dev,
+				 const struct snd_soc_acpi_link_adr *link,
+				 struct snd_soc_dai_link_component *codec,
+				 int offset)
+{
+	int i;
+
+	for (i = 0; i < link->num_adr; i++) {
+		unsigned int sdw_version, unique_id, mfg_id;
+		unsigned int link_id, part_id, class_id;
+		int codec_index, comp_index;
+		char *codec_str;
+		u64 adr;
+
+		adr = link->adr_d[i].adr;
+
+		sdw_version = SDW_VERSION(adr);
+		link_id = SDW_DISCO_LINK_ID(adr);
+		unique_id = SDW_UNIQUE_ID(adr);
+		mfg_id = SDW_MFG_ID(adr);
+		part_id = SDW_PART_ID(adr);
+		class_id = SDW_CLASS_ID(adr);
+
+		comp_index = i + offset;
+		if (is_unique_device(link, sdw_version, mfg_id, part_id,
+				     class_id, i)) {
+			codec_str = "sdw:%x:%x:%x:%x";
+			codec[comp_index].name =
+				devm_kasprintf(dev, GFP_KERNEL, codec_str,
+					       link_id, mfg_id, part_id,
+					       class_id);
+		} else {
+			codec_str = "sdw:%x:%x:%x:%x:%x";
+			codec[comp_index].name =
+				devm_kasprintf(dev, GFP_KERNEL, codec_str,
+					       link_id, mfg_id, part_id,
+					       class_id, unique_id);
+		}
+
+		if (!codec[comp_index].name)
+			return -ENOMEM;
+
+		codec_index = find_codec_info_part(part_id);
+		if (codec_index < 0)
+			return codec_index;
+
+		codec[comp_index].dai_name =
+			codec_info_list[codec_index].dai_name;
+	}
+
+	return 0;
+}
+
+static int set_codec_init_func(const struct snd_soc_acpi_link_adr *link,
+			       struct snd_soc_dai_link *dai_links,
+			       bool playback)
+{
+	int i;
+
+	for (i = 0; i < link->num_adr; i++) {
+		unsigned int part_id;
+		int codec_index;
+
+		part_id = SDW_PART_ID(link->adr_d[i].adr);
+		codec_index = find_codec_info_part(part_id);
+
+		if (codec_index < 0)
+			return codec_index;
+
+		if (codec_info_list[codec_index].init)
+			codec_info_list[codec_index].init(link, dai_links,
+						 &codec_info_list[codec_index],
+						 playback);
+	}
+
+	return 0;
+}
+
+/*
+ * check endpoint status in slaves and gather link ID for all slaves in
+ * the same group to generate different CPU DAI. Now only support
+ * one sdw link with all slaves set with only single group id.
+ *
+ * one slave on one sdw link with aggregated = 0
+ * one sdw BE DAI <---> one-cpu DAI <---> one-codec DAI
+ *
+ * two or more slaves on one sdw link with aggregated = 0
+ * one sdw BE DAI  <---> one-cpu DAI <---> multi-codec DAIs
+ *
+ * multiple links with multiple slaves with aggregated = 1
+ * one sdw BE DAI  <---> 1 .. N CPU DAIs <----> 1 .. N codec DAIs
+ */
+static int get_slave_info(const struct snd_soc_acpi_link_adr *adr_link,
+			  struct device *dev, int *cpu_dai_id, int *cpu_dai_num,
+			  int *codec_num, int *group_id,
+			  bool *group_generated)
+{
+	const struct snd_soc_acpi_adr_device *adr_d;
+	const struct snd_soc_acpi_link_adr *adr_next;
+	bool no_aggregation;
+	int index = 0;
+
+	no_aggregation = sof_sdw_quirk & SOF_SDW_NO_AGGREGATION;
+	*codec_num = adr_link->num_adr;
+	adr_d = adr_link->adr_d;
+
+	/* make sure the link mask has a single bit set */
+	if (!is_power_of_2(adr_link->mask))
+		return -EINVAL;
+
+	cpu_dai_id[index++] = ffs(adr_link->mask) - 1;
+	if (!adr_d->endpoints->aggregated || no_aggregation) {
+		*cpu_dai_num = 1;
+		*group_id = 0;
+		return 0;
+	}
+
+	*group_id = adr_d->endpoints->group_id;
+
+	/* gather other link ID of slaves in the same group */
+	for (adr_next = adr_link + 1; adr_next && adr_next->num_adr;
+		adr_next++) {
+		const struct snd_soc_acpi_endpoint *endpoint;
+
+		endpoint = adr_next->adr_d->endpoints;
+		if (!endpoint->aggregated ||
+		    endpoint->group_id != *group_id)
+			continue;
+
+		/* make sure the link mask has a single bit set */
+		if (!is_power_of_2(adr_next->mask))
+			return -EINVAL;
+
+		if (index >= SDW_MAX_CPU_DAIS) {
+			dev_err(dev, " cpu_dai_id array overflows");
+			return -EINVAL;
+		}
+
+		cpu_dai_id[index++] = ffs(adr_next->mask) - 1;
+		*codec_num += adr_next->num_adr;
+	}
+
+	/*
+	 * indicate CPU DAIs for this group have been generated
+	 * to avoid generating CPU DAIs for this group again.
+	 */
+	group_generated[*group_id] = true;
+	*cpu_dai_num = index;
+
+	return 0;
+}
+
+static int create_sdw_dailink(struct device *dev, int *be_index,
+			      struct snd_soc_dai_link *dai_links,
+			      int sdw_be_num, int sdw_cpu_dai_num,
+			      struct snd_soc_dai_link_component *cpus,
+			      const struct snd_soc_acpi_link_adr *link,
+			      int *cpu_id, bool *group_generated)
+{
+	const struct snd_soc_acpi_link_adr *link_next;
+	struct snd_soc_dai_link_component *codecs;
+	int cpu_dai_id[SDW_MAX_CPU_DAIS];
+	int cpu_dai_num, cpu_dai_index;
+	unsigned int part_id, group_id;
+	int codec_idx = 0;
+	int i = 0, j = 0;
+	int codec_index;
+	int codec_num;
+	int stream;
+	int ret;
+	int k;
+
+	ret = get_slave_info(link, dev, cpu_dai_id, &cpu_dai_num, &codec_num,
+			     &group_id, group_generated);
+	if (ret)
+		return ret;
+
+	codecs = devm_kcalloc(dev, codec_num, sizeof(*codecs), GFP_KERNEL);
+	if (!codecs)
+		return -ENOMEM;
+
+	/* generate codec name on different links in the same group */
+	for (link_next = link; link_next && link_next->num_adr &&
+	     i < cpu_dai_num; link_next++) {
+		const struct snd_soc_acpi_endpoint *endpoints;
+
+		endpoints = link_next->adr_d->endpoints;
+		if (group_id && (!endpoints->aggregated ||
+				 endpoints->group_id != group_id))
+			continue;
+
+		/* skip the link excluded by this processed group */
+		if (cpu_dai_id[i] != ffs(link_next->mask) - 1)
+			continue;
+
+		ret = create_codec_dai_name(dev, link_next, codecs, codec_idx);
+		if (ret < 0)
+			return ret;
+
+		/* check next link to create codec dai in the processed group */
+		i++;
+		codec_idx += link_next->num_adr;
+	}
+
+	/* find codec info to create BE DAI */
+	part_id = SDW_PART_ID(link->adr_d[0].adr);
+	codec_index = find_codec_info_part(part_id);
+	if (codec_index < 0)
+		return codec_index;
+
+	cpu_dai_index = *cpu_id;
+	for_each_pcm_streams(stream) {
+		char *name, *cpu_name;
+		int playback, capture;
+		static const char * const sdw_stream_name[] = {
+			"SDW%d-Playback",
+			"SDW%d-Capture",
+		};
+
+		if (!codec_info_list[codec_index].direction[stream])
+			continue;
+
+		/* create stream name according to first link id */
+		name = devm_kasprintf(dev, GFP_KERNEL,
+				      sdw_stream_name[stream], cpu_dai_id[0]);
+		if (!name)
+			return -ENOMEM;
+
+		/*
+		 * generate CPU DAI name base on the sdw link ID and
+		 * PIN ID with offset of 2 according to sdw dai driver.
+		 */
+		for (k = 0; k < cpu_dai_num; k++) {
+			cpu_name = devm_kasprintf(dev, GFP_KERNEL,
+						  "SDW%d Pin%d", cpu_dai_id[k],
+						  j + SDW_INTEL_BIDIR_PDI_BASE);
+			if (!cpu_name)
+				return -ENOMEM;
+
+			if (cpu_dai_index >= sdw_cpu_dai_num) {
+				dev_err(dev, "invalid cpu dai index %d",
+					cpu_dai_index);
+				return -EINVAL;
+			}
+
+			cpus[cpu_dai_index++].dai_name = cpu_name;
+		}
+
+		if (*be_index >= sdw_be_num) {
+			dev_err(dev, " invalid be dai index %d", *be_index);
+			return -EINVAL;
+		}
+
+		if (*cpu_id >= sdw_cpu_dai_num) {
+			dev_err(dev, " invalid cpu dai index %d", *cpu_id);
+			return -EINVAL;
+		}
+
+		playback = (stream == SNDRV_PCM_STREAM_PLAYBACK);
+		capture = (stream == SNDRV_PCM_STREAM_CAPTURE);
+		init_dai_link(dai_links + *be_index, *be_index, name,
+			      playback, capture,
+			      cpus + *cpu_id, cpu_dai_num,
+			      codecs, codec_num,
+			      NULL, &sdw_ops);
+
+		ret = set_codec_init_func(link, dai_links + (*be_index)++,
+					  playback);
+		if (ret < 0) {
+			dev_err(dev, "failed to init codec %d", codec_index);
+			return ret;
+		}
+
+		*cpu_id += cpu_dai_num;
+		j++;
+	}
+
+	return 0;
+}
+
+/*
+ * DAI link ID of SSP & DMIC & HDMI are based on last
+ * link ID used by sdw link. Since be_id may be changed
+ * in init func of sdw codec, it is not equal to be_id
+ */
+static inline int get_next_be_id(struct snd_soc_dai_link *links,
+				 int be_id)
+{
+	return links[be_id - 1].id + 1;
+}
+
+static int sof_card_dai_links_create(struct device *dev,
+				     struct snd_soc_acpi_mach *mach,
+				     struct snd_soc_card *card)
+{
+	int ssp_num, sdw_be_num = 0, hdmi_num = 0, dmic_num;
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC)
+	struct snd_soc_dai_link_component *idisp_components;
+#endif
+	struct snd_soc_dai_link_component *ssp_components;
+	struct snd_soc_acpi_mach_params *mach_params;
+	const struct snd_soc_acpi_link_adr *adr_link;
+	struct snd_soc_dai_link_component *cpus;
+	bool group_generated[SDW_MAX_GROUPS];
+	int ssp_codec_index, ssp_mask;
+	struct snd_soc_dai_link *links;
+	int num_links, link_id = 0;
+	char *name, *cpu_name;
+	int total_cpu_dai_num;
+	int sdw_cpu_dai_num;
+	int i, j, be_id = 0;
+	int cpu_id = 0;
+	int comp_num;
+	int ret;
+
+	/* reset amp_num to ensure amp_num++ starts from 0 in each probe */
+	for (i = 0; i < ARRAY_SIZE(codec_info_list); i++)
+		codec_info_list[i].amp_num = 0;
+
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC)
+	hdmi_num = sof_sdw_quirk & SOF_SDW_TGL_HDMI ?
+				SOF_TGL_HDMI_COUNT : SOF_PRE_TGL_HDMI_COUNT;
+#endif
+
+	ssp_mask = SOF_SSP_GET_PORT(sof_sdw_quirk);
+	/*
+	 * on generic tgl platform, I2S or sdw mode is supported
+	 * based on board rework. A ACPI device is registered in
+	 * system only when I2S mode is supported, not sdw mode.
+	 * Here check ACPI ID to confirm I2S is supported.
+	 */
+	ssp_codec_index = find_codec_info_acpi(mach->id);
+	ssp_num = ssp_codec_index >= 0 ? hweight_long(ssp_mask) : 0;
+	comp_num = hdmi_num + ssp_num;
+
+	mach_params = &mach->mach_params;
+	ret = get_sdw_dailink_info(mach_params->links,
+				   &sdw_be_num, &sdw_cpu_dai_num);
+	if (ret < 0) {
+		dev_err(dev, "failed to get sdw link info %d", ret);
+		return ret;
+	}
+
+	/* enable dmic01 & dmic16k */
+	dmic_num = (sof_sdw_quirk & SOF_SDW_PCH_DMIC) ? 2 : 0;
+	comp_num += dmic_num;
+
+	dev_dbg(dev, "sdw %d, ssp %d, dmic %d, hdmi %d", sdw_be_num, ssp_num,
+		dmic_num, hdmi_num);
+
+	/* allocate BE dailinks */
+	num_links = comp_num + sdw_be_num;
+	links = devm_kcalloc(dev, num_links, sizeof(*links), GFP_KERNEL);
+
+	/* allocated CPU DAIs */
+	total_cpu_dai_num = comp_num + sdw_cpu_dai_num;
+	cpus = devm_kcalloc(dev, total_cpu_dai_num, sizeof(*cpus),
+			    GFP_KERNEL);
+
+	if (!links || !cpus)
+		return -ENOMEM;
+
+	/* SDW */
+	if (!sdw_be_num)
+		goto SSP;
+
+	adr_link = mach_params->links;
+	if (!adr_link)
+		return -EINVAL;
+
+	/*
+	 * SoundWire Slaves aggregated in the same group may be
+	 * located on different hardware links. Clear array to indicate
+	 * CPU DAIs for this group have not been generated.
+	 */
+	for (i = 0; i < SDW_MAX_GROUPS; i++)
+		group_generated[i] = false;
+
+	/* generate DAI links by each sdw link */
+	for (; adr_link->num_adr; adr_link++) {
+		const struct snd_soc_acpi_endpoint *endpoint;
+
+		endpoint = adr_link->adr_d->endpoints;
+		if (endpoint->aggregated && !endpoint->group_id) {
+			dev_err(dev, "invalid group id on link %x",
+				adr_link->mask);
+			continue;
+		}
+
+		/* this group has been generated */
+		if (endpoint->aggregated &&
+		    group_generated[endpoint->group_id])
+			continue;
+
+		ret = create_sdw_dailink(dev, &be_id, links, sdw_be_num,
+					 sdw_cpu_dai_num, cpus, adr_link,
+					 &cpu_id, group_generated);
+		if (ret < 0) {
+			dev_err(dev, "failed to create dai link %d", be_id);
+			return -ENOMEM;
+		}
+	}
+
+	/* non-sdw DAI follows sdw DAI */
+	link_id = be_id;
+
+	/* get BE ID for non-sdw DAI */
+	be_id = get_next_be_id(links, be_id);
+
+SSP:
+	/* SSP */
+	if (!ssp_num)
+		goto DMIC;
+
+	for (i = 0, j = 0; ssp_mask; i++, ssp_mask >>= 1) {
+		struct sof_sdw_codec_info *info;
+		int playback, capture;
+		char *codec_name;
+
+		if (!(ssp_mask & 0x1))
+			continue;
+
+		name = devm_kasprintf(dev, GFP_KERNEL,
+				      "SSP%d-Codec", i);
+		if (!name)
+			return -ENOMEM;
+
+		cpu_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", i);
+		if (!cpu_name)
+			return -ENOMEM;
+
+		ssp_components = devm_kzalloc(dev, sizeof(*ssp_components),
+					      GFP_KERNEL);
+		if (!ssp_components)
+			return -ENOMEM;
+
+		info = &codec_info_list[ssp_codec_index];
+		codec_name = devm_kasprintf(dev, GFP_KERNEL, "i2c-%s:0%d",
+					    info->acpi_id, j++);
+		if (!codec_name)
+			return -ENOMEM;
+
+		ssp_components->name = codec_name;
+		ssp_components->dai_name = info->dai_name;
+		cpus[cpu_id].dai_name = cpu_name;
+
+		playback = info->direction[SNDRV_PCM_STREAM_PLAYBACK];
+		capture = info->direction[SNDRV_PCM_STREAM_CAPTURE];
+		init_dai_link(links + link_id, be_id, name,
+			      playback, capture,
+			      cpus + cpu_id, 1,
+			      ssp_components, 1,
+			      NULL, info->ops);
+
+		ret = info->init(NULL, links + link_id, info, 0);
+		if (ret < 0)
+			return ret;
+
+		INC_ID(be_id, cpu_id, link_id);
+	}
+
+DMIC:
+	/* dmic */
+	if (dmic_num > 0) {
+		cpus[cpu_id].dai_name = "DMIC01 Pin";
+		init_dai_link(links + link_id, be_id, "dmic01",
+			      0, 1, // DMIC only supports capture
+			      cpus + cpu_id, 1,
+			      dmic_component, 1,
+			      sof_sdw_dmic_init, NULL);
+		INC_ID(be_id, cpu_id, link_id);
+
+		cpus[cpu_id].dai_name = "DMIC16k Pin";
+		init_dai_link(links + link_id, be_id, "dmic16k",
+			      0, 1, // DMIC only supports capture
+			      cpus + cpu_id, 1,
+			      dmic_component, 1,
+			      /* don't call sof_sdw_dmic_init() twice */
+			      NULL, NULL);
+		INC_ID(be_id, cpu_id, link_id);
+	}
+
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC)
+	/* HDMI */
+	if (hdmi_num > 0) {
+		idisp_components = devm_kcalloc(dev, hdmi_num,
+						sizeof(*idisp_components),
+						GFP_KERNEL);
+		if (!idisp_components)
+			return -ENOMEM;
+	}
+
+	for (i = 0; i < hdmi_num; i++) {
+		name = devm_kasprintf(dev, GFP_KERNEL,
+				      "iDisp%d", i + 1);
+		if (!name)
+			return -ENOMEM;
+
+		idisp_components[i].name = "ehdaudio0D2";
+		idisp_components[i].dai_name = devm_kasprintf(dev,
+							      GFP_KERNEL,
+							      "intel-hdmi-hifi%d",
+							      i + 1);
+		if (!idisp_components[i].dai_name)
+			return -ENOMEM;
+
+		cpu_name = devm_kasprintf(dev, GFP_KERNEL,
+					  "iDisp%d Pin", i + 1);
+		if (!cpu_name)
+			return -ENOMEM;
+
+		cpus[cpu_id].dai_name = cpu_name;
+		init_dai_link(links + link_id, be_id, name,
+			      1, 0, // HDMI only supports playback
+			      cpus + cpu_id, 1,
+			      idisp_components + i, 1,
+			      sof_sdw_hdmi_init, NULL);
+		INC_ID(be_id, cpu_id, link_id);
+	}
+#endif
+
+	card->dai_link = links;
+	card->num_links = num_links;
+
+	return 0;
+}
+
+/* SoC card */
+static const char sdw_card_long_name[] = "Intel Soundwire SOF";
+
+static struct snd_soc_card card_sof_sdw = {
+	.name = "soundwire",
+	.late_probe = sof_sdw_hdmi_card_late_probe,
+	.codec_conf = codec_conf,
+	.num_configs = ARRAY_SIZE(codec_conf),
+};
+
+static int mc_probe(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = &card_sof_sdw;
+	struct snd_soc_acpi_mach *mach;
+	struct mc_private *ctx;
+	int ret;
+
+	dev_dbg(&pdev->dev, "Entry %s\n", __func__);
+
+	ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL);
+	if (!ctx)
+		return -ENOMEM;
+
+	dmi_check_system(sof_sdw_quirk_table);
+
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC)
+	INIT_LIST_HEAD(&ctx->hdmi_pcm_list);
+#endif
+
+	card->dev = &pdev->dev;
+
+	mach = pdev->dev.platform_data;
+	ret = sof_card_dai_links_create(&pdev->dev, mach,
+					card);
+	if (ret < 0)
+		return ret;
+
+	ctx->common_hdmi_codec_drv = mach->mach_params.common_hdmi_codec_drv;
+
+	snd_soc_card_set_drvdata(card, ctx);
+
+	card->components = devm_kasprintf(card->dev, GFP_KERNEL,
+					  "cfg-spk:%d",
+					  (sof_sdw_quirk & SOF_SDW_FOUR_SPK) ? 4 : 2);
+	if (!card->components)
+		return -ENOMEM;
+
+	card->long_name = sdw_card_long_name;
+
+	/* Register the card */
+	ret = devm_snd_soc_register_card(&pdev->dev, card);
+	if (ret) {
+		dev_err(card->dev, "snd_soc_register_card failed %d\n", ret);
+		return ret;
+	}
+
+	platform_set_drvdata(pdev, card);
+
+	return ret;
+}
+
+static struct platform_driver sof_sdw_driver = {
+	.driver = {
+		.name = "sof_sdw",
+		.pm = &snd_soc_pm_ops,
+	},
+	.probe = mc_probe,
+};
+
+module_platform_driver(sof_sdw_driver);
+
+MODULE_DESCRIPTION("ASoC SoundWire Generic Machine driver");
+MODULE_AUTHOR("Bard Liao <[e-mail]>");
+MODULE_AUTHOR("Rander Wang <[e-mail]>");
+MODULE_AUTHOR("Pierre-Louis Bossart <[e-mail]>");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:sof_sdw");

sound/soc/intel/boards/sof_sdw_common.h

···
+/* SPDX-License-Identifier: GPL-2.0
+ *  Copyright (c) 2020 Intel Corporation
+ */
+
+/*
+ *  sof_sdw_common.h - prototypes for common helpers
+ */
+
+#ifndef SND_SOC_SOF_SDW_COMMON_H
+#define SND_SOC_SOF_SDW_COMMON_H
+
+#include <linux/bits.h>
+#include <linux/types.h>
+
+#define MAX_NO_PROPS 2
+#define MAX_HDMI_NUM 4
+#define SDW_DMIC_DAI_ID 4
+#define SDW_MAX_CPU_DAIS 16
+#define SDW_INTEL_BIDIR_PDI_BASE 2
+
+/* 8 combinations with 4 links + unused group 0 */
+#define SDW_MAX_GROUPS 9
+
+enum {
+	SOF_RT711_JD_SRC_JD1 = 1,
+	SOF_RT711_JD_SRC_JD2 = 2,
+};
+
+enum {
+	SOF_PRE_TGL_HDMI_COUNT = 3,
+	SOF_TGL_HDMI_COUNT = 4,
+};
+
+enum {
+	SOF_I2S_SSP0 = BIT(0),
+	SOF_I2S_SSP1 = BIT(1),
+	SOF_I2S_SSP2 = BIT(2),
+	SOF_I2S_SSP3 = BIT(3),
+	SOF_I2S_SSP4 = BIT(4),
+	SOF_I2S_SSP5 = BIT(5),
+};
+
+#define SOF_RT711_JDSRC(quirk)		((quirk) & GENMASK(1, 0))
+#define SOF_SDW_FOUR_SPK		BIT(2)
+#define SOF_SDW_TGL_HDMI		BIT(3)
+#define SOF_SDW_PCH_DMIC		BIT(4)
+#define SOF_SSP_PORT(x)		(((x) & GENMASK(5, 0)) << 5)
+#define SOF_SSP_GET_PORT(quirk)	(((quirk) >> 5) & GENMASK(5, 0))
+#define SOF_RT715_DAI_ID_FIX		BIT(11)
+#define SOF_SDW_NO_AGGREGATION		BIT(12)
+
+struct sof_sdw_codec_info {
+	const int id;
+	int amp_num;
+	const u8 acpi_id[ACPI_ID_LEN];
+	const bool direction[2]; // playback & capture support
+	const char *dai_name;
+	const struct snd_soc_ops *ops;
+
+	int  (*init)(const struct snd_soc_acpi_link_adr *link,
+		     struct snd_soc_dai_link *dai_links,
+		     struct sof_sdw_codec_info *info,
+		     bool playback);
+};
+
+struct mc_private {
+	struct list_head hdmi_pcm_list;
+	bool common_hdmi_codec_drv;
+	struct snd_soc_jack sdw_headset;
+};
+
+extern unsigned long sof_sdw_quirk;
+
+/* generic HDMI support */
+int sof_sdw_hdmi_init(struct snd_soc_pcm_runtime *rtd);
+
+int sof_sdw_hdmi_card_late_probe(struct snd_soc_card *card);
+
+/* DMIC support */
+int sof_sdw_dmic_init(struct snd_soc_pcm_runtime *rtd);
+
+/* RT711 support */
+int sof_sdw_rt711_init(const struct snd_soc_acpi_link_adr *link,
+		       struct snd_soc_dai_link *dai_links,
+		       struct sof_sdw_codec_info *info,
+		       bool playback);
+
+/* RT700 support */
+int sof_sdw_rt700_init(const struct snd_soc_acpi_link_adr *link,
+		       struct snd_soc_dai_link *dai_links,
+		       struct sof_sdw_codec_info *info,
+		       bool playback);
+
+/* RT1308 support */
+extern struct snd_soc_ops sof_sdw_rt1308_i2s_ops;
+
+int sof_sdw_rt1308_init(const struct snd_soc_acpi_link_adr *link,
+			struct snd_soc_dai_link *dai_links,
+			struct sof_sdw_codec_info *info,
+			bool playback);
+
+/* RT715 support */
+int sof_sdw_rt715_init(const struct snd_soc_acpi_link_adr *link,
+		       struct snd_soc_dai_link *dai_links,
+		       struct sof_sdw_codec_info *info,
+		       bool playback);
+
+/* RT5682 support */
+int sof_sdw_rt5682_init(const struct snd_soc_acpi_link_adr *link,
+			struct snd_soc_dai_link *dai_links,
+			struct sof_sdw_codec_info *info,
+			bool playback);
+
+#endif

sound/soc/intel/boards/sof_sdw_dmic.c

···
+// SPDX-License-Identifier: GPL-2.0
+// Copyright (c) 2020 Intel Corporation
+
+/*
+ *  sof_sdw_dmic - Helpers to handle dmic from generic machine driver
+ */
+
+#include <sound/soc.h>
+#include <sound/soc-acpi.h>
+#include "sof_sdw_common.h"
+
+static const struct snd_soc_dapm_widget dmic_widgets[] = {
+	SND_SOC_DAPM_MIC("SoC DMIC", NULL),
+};
+
+static const struct snd_soc_dapm_route dmic_map[] = {
+	/* digital mics */
+	{"DMic", NULL, "SoC DMIC"},
+};
+
+int sof_sdw_dmic_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_card *card = rtd->card;
+	int ret;
+
+	ret = snd_soc_dapm_new_controls(&card->dapm, dmic_widgets,
+					ARRAY_SIZE(dmic_widgets));
+	if (ret) {
+		dev_err(card->dev, "DMic widget addition failed: %d\n", ret);
+		/* Don't need to add routes if widget addition failed */
+		return ret;
+	}
+
+	ret = snd_soc_dapm_add_routes(&card->dapm, dmic_map,
+				      ARRAY_SIZE(dmic_map));
+
+	if (ret)
+		dev_err(card->dev, "DMic map addition failed: %d\n", ret);
+
+	return ret;
+}
+

sound/soc/intel/boards/sof_sdw_hdmi.c

···
+// SPDX-License-Identifier: GPL-2.0
+// Copyright (c) 2020 Intel Corporation
+
+/*
+ *  sof_sdw_hdmi - Helpers to handle HDMI from generic machine driver
+ */
+
+#include <linux/device.h>
+#include <linux/errno.h>
+#include <linux/kernel.h>
+#include <linux/list.h>
+#include <sound/soc.h>
+#include <sound/soc-acpi.h>
+#include <sound/jack.h>
+#include "sof_sdw_common.h"
+#include "../../codecs/hdac_hdmi.h"
+#include "hda_dsp_common.h"
+
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC)
+static struct snd_soc_jack hdmi[MAX_HDMI_NUM];
+
+struct hdmi_pcm {
+	struct list_head head;
+	struct snd_soc_dai *codec_dai;
+	int device;
+};
+
+int sof_sdw_hdmi_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct mc_private *ctx = snd_soc_card_get_drvdata(rtd->card);
+	struct snd_soc_dai *dai = rtd->codec_dai;
+	struct hdmi_pcm *pcm;
+
+	pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
+	if (!pcm)
+		return -ENOMEM;
+
+	/* dai_link id is 1:1 mapped to the PCM device */
+	pcm->device = rtd->dai_link->id;
+	pcm->codec_dai = dai;
+
+	list_add_tail(&pcm->head, &ctx->hdmi_pcm_list);
+
+	return 0;
+}
+
+#define NAME_SIZE	32
+int sof_sdw_hdmi_card_late_probe(struct snd_soc_card *card)
+{
+	struct mc_private *ctx = snd_soc_card_get_drvdata(card);
+	struct hdmi_pcm *pcm;
+	struct snd_soc_component *component = NULL;
+	int err, i = 0;
+	char jack_name[NAME_SIZE];
+
+	pcm = list_first_entry(&ctx->hdmi_pcm_list, struct hdmi_pcm,
+			       head);
+	component = pcm->codec_dai->component;
+
+	if (ctx->common_hdmi_codec_drv)
+		return hda_dsp_hdmi_build_controls(card, component);
+
+	list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) {
+		component = pcm->codec_dai->component;
+		snprintf(jack_name, sizeof(jack_name),
+			 "HDMI/DP, pcm=%d Jack", pcm->device);
+		err = snd_soc_card_jack_new(card, jack_name,
+					    SND_JACK_AVOUT, &hdmi[i],
+					    NULL, 0);
+
+		if (err)
+			return err;
+
+		err = snd_jack_add_new_kctl(hdmi[i].jack,
+					    jack_name, SND_JACK_AVOUT);
+		if (err)
+			dev_warn(component->dev, "failed creating Jack kctl\n");
+
+		err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device,
+					  &hdmi[i]);
+		if (err < 0)
+			return err;
+
+		i++;
+	}
+
+	if (!component)
+		return -EINVAL;
+
+	return hdac_hdmi_jack_port_init(component, &card->dapm);
+}
+#else
+int hdmi_card_late_probe(struct snd_soc_card *card)
+{
+	return 0;
+}
+#endif

sound/soc/intel/boards/sof_sdw_rt1308.c

···
+// SPDX-License-Identifier: GPL-2.0
+// Copyright (c) 2020 Intel Corporation
+
+/*
+ *  sof_sdw_rt1308 - Helpers to handle RT1308 from generic machine driver
+ */
+
+#include <linux/device.h>
+#include <linux/errno.h>
+#include <sound/soc.h>
+#include <sound/soc-acpi.h>
+#include "sof_sdw_common.h"
+#include "../../codecs/rt1308.h"
+
+static const struct snd_soc_dapm_widget rt1308_widgets[] = {
+	SND_SOC_DAPM_SPK("Speaker", NULL),
+};
+
+/*
+ * dapm routes for rt1308 will be registered dynamically according
+ * to the number of rt1308 used. The first two entries will be registered
+ * for one codec case, and the last two entries are also registered
+ * if two 1308s are used.
+ */
+static const struct snd_soc_dapm_route rt1308_map[] = {
+	{ "Speaker", NULL, "rt1308-1 SPOL" },
+	{ "Speaker", NULL, "rt1308-1 SPOR" },
+	{ "Speaker", NULL, "rt1308-2 SPOL" },
+	{ "Speaker", NULL, "rt1308-2 SPOR" },
+};
+
+static const struct snd_kcontrol_new rt1308_controls[] = {
+	SOC_DAPM_PIN_SWITCH("Speaker"),
+};
+
+static int first_spk_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_card *card = rtd->card;
+	int ret;
+
+	card->components = devm_kasprintf(card->dev, GFP_KERNEL,
+					  "%s spk:rt1308",
+					  card->components);
+	if (!card->components)
+		return -ENOMEM;
+
+	ret = snd_soc_add_card_controls(card, rt1308_controls,
+					ARRAY_SIZE(rt1308_controls));
+	if (ret) {
+		dev_err(card->dev, "rt1308 controls addition failed: %d\n", ret);
+		return ret;
+	}
+
+	ret = snd_soc_dapm_new_controls(&card->dapm, rt1308_widgets,
+					ARRAY_SIZE(rt1308_widgets));
+	if (ret) {
+		dev_err(card->dev, "rt1308 widgets addition failed: %d\n", ret);
+		return ret;
+	}
+
+	ret = snd_soc_dapm_add_routes(&card->dapm, rt1308_map, 2);
+	if (ret)
+		dev_err(rtd->dev, "failed to add first SPK map: %d\n", ret);
+
+	return ret;
+}
+
+static int second_spk_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_card *card = rtd->card;
+	int ret;
+
+	ret = snd_soc_dapm_add_routes(&card->dapm, rt1308_map + 2, 2);
+	if (ret)
+		dev_err(rtd->dev, "failed to add second SPK map: %d\n", ret);
+
+	return ret;
+}
+
+static int all_spk_init(struct snd_soc_pcm_runtime *rtd)
+{
+	int ret;
+
+	ret = first_spk_init(rtd);
+	if (ret)
+		return ret;
+
+	return second_spk_init(rtd);
+}
+
+static int rt1308_i2s_hw_params(struct snd_pcm_substream *substream,
+				struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_card *card = rtd->card;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	int clk_id, clk_freq, pll_out;
+	int err;
+
+	clk_id = RT1308_PLL_S_MCLK;
+	clk_freq = 38400000;
+
+	pll_out = params_rate(params) * 512;
+
+	/* Set rt1308 pll */
+	err = snd_soc_dai_set_pll(codec_dai, 0, clk_id, clk_freq, pll_out);
+	if (err < 0) {
+		dev_err(card->dev, "Failed to set RT1308 PLL: %d\n", err);
+		return err;
+	}
+
+	/* Set rt1308 sysclk */
+	err = snd_soc_dai_set_sysclk(codec_dai, RT1308_FS_SYS_S_PLL, pll_out,
+				     SND_SOC_CLOCK_IN);
+	if (err < 0) {
+		dev_err(card->dev, "Failed to set RT1308 SYSCLK: %d\n", err);
+		return err;
+	}
+
+	return 0;
+}
+
+/* machine stream operations */
+struct snd_soc_ops sof_sdw_rt1308_i2s_ops = {
+	.hw_params = rt1308_i2s_hw_params,
+};
+
+int sof_sdw_rt1308_init(const struct snd_soc_acpi_link_adr *link,
+			struct snd_soc_dai_link *dai_links,
+			struct sof_sdw_codec_info *info,
+			bool playback)
+{
+	info->amp_num++;
+	if (info->amp_num == 1)
+		dai_links->init = first_spk_init;
+
+	if (info->amp_num == 2) {
+		/*
+		 * if two 1308s are in one dai link, the init function
+		 * in this dai link will be first set for the first speaker,
+		 * and it should be reset to initialize all speakers when
+		 * the second speaker is found.
+		 */
+		if (dai_links->init)
+			dai_links->init = all_spk_init;
+		else
+			dai_links->init = second_spk_init;
+	}
+
+	return 0;
+}

sound/soc/intel/boards/sof_sdw_rt5682.c

···
+// SPDX-License-Identifier: GPL-2.0
+// Copyright (c) 2020 Intel Corporation
+
+/*
+ *  sof_sdw_rt5682 - Helpers to handle RT5682 from generic machine driver
+ */
+
+#include <linux/device.h>
+#include <linux/errno.h>
+#include <linux/input.h>
+#include <linux/soundwire/sdw.h>
+#include <linux/soundwire/sdw_type.h>
+#include <sound/soc.h>
+#include <sound/soc-acpi.h>
+#include <sound/jack.h>
+#include "sof_sdw_common.h"
+
+static const struct snd_soc_dapm_widget rt5682_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone", NULL),
+	SND_SOC_DAPM_MIC("Headset Mic", NULL),
+};
+
+static const struct snd_soc_dapm_route rt5682_map[] = {
+	/*Headphones*/
+	{ "Headphone", NULL, "rt5682 HPOL" },
+	{ "Headphone", NULL, "rt5682 HPOR" },
+	{ "rt5682 IN1P", NULL, "Headset Mic" },
+};
+
+static const struct snd_kcontrol_new rt5682_controls[] = {
+	SOC_DAPM_PIN_SWITCH("Headphone"),
+	SOC_DAPM_PIN_SWITCH("Headset Mic"),
+};
+
+static struct snd_soc_jack_pin rt5682_jack_pins[] = {
+	{
+		.pin    = "Headphone",
+		.mask   = SND_JACK_HEADPHONE,
+	},
+	{
+		.pin    = "Headset Mic",
+		.mask   = SND_JACK_MICROPHONE,
+	},
+};
+
+static int rt5682_rtd_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_card *card = rtd->card;
+	struct mc_private *ctx = snd_soc_card_get_drvdata(card);
+	struct snd_soc_component *component = rtd->codec_dai->component;
+	struct snd_soc_jack *jack;
+	int ret;
+
+	card->components = devm_kasprintf(card->dev, GFP_KERNEL,
+					  "%s hs:rt5682",
+					  card->components);
+	if (!card->components)
+		return -ENOMEM;
+
+	ret = snd_soc_add_card_controls(card, rt5682_controls,
+					ARRAY_SIZE(rt5682_controls));
+	if (ret) {
+		dev_err(card->dev, "rt5682 control addition failed: %d\n", ret);
+		return ret;
+	}
+
+	ret = snd_soc_dapm_new_controls(&card->dapm, rt5682_widgets,
+					ARRAY_SIZE(rt5682_widgets));
+	if (ret) {
+		dev_err(card->dev, "rt5682 widgets addition failed: %d\n", ret);
+		return ret;
+	}
+
+	ret = snd_soc_dapm_add_routes(&card->dapm, rt5682_map,
+				      ARRAY_SIZE(rt5682_map));
+
+	if (ret) {
+		dev_err(card->dev, "rt5682 map addition failed: %d\n", ret);
+		return ret;
+	}
+
+	ret = snd_soc_card_jack_new(rtd->card, "Headset Jack",
+				    SND_JACK_HEADSET | SND_JACK_BTN_0 |
+				    SND_JACK_BTN_1 | SND_JACK_BTN_2 |
+				    SND_JACK_BTN_3,
+				    &ctx->sdw_headset,
+				    rt5682_jack_pins,
+				    ARRAY_SIZE(rt5682_jack_pins));
+	if (ret) {
+		dev_err(rtd->card->dev, "Headset Jack creation failed: %d\n",
+			ret);
+		return ret;
+	}
+
+	jack = &ctx->sdw_headset;
+
+	snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE);
+	snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND);
+	snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP);
+	snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN);
+
+	ret = snd_soc_component_set_jack(component, jack, NULL);
+
+	if (ret)
+		dev_err(rtd->card->dev, "Headset Jack call-back failed: %d\n",
+			ret);
+
+	return ret;
+}
+
+int sof_sdw_rt5682_init(const struct snd_soc_acpi_link_adr *link,
+			struct snd_soc_dai_link *dai_links,
+			struct sof_sdw_codec_info *info,
+			bool playback)
+{
+	/*
+	 * headset should be initialized once.
+	 * Do it with dai link for playback.
+	 */
+	if (!playback)
+		return 0;
+
+	dai_links->init = rt5682_rtd_init;
+
+	return 0;
+}

sound/soc/intel/boards/sof_sdw_rt700.c

···
+// SPDX-License-Identifier: GPL-2.0
+// Copyright (c) 2020 Intel Corporation
+
+/*
+ *  sof_sdw_rt700 - Helpers to handle RT700 from generic machine driver
+ */
+
+#include <linux/device.h>
+#include <linux/errno.h>
+#include <linux/input.h>
+#include <sound/soc.h>
+#include <sound/soc-acpi.h>
+#include <sound/jack.h>
+#include "sof_sdw_common.h"
+
+static const struct snd_soc_dapm_widget rt700_widgets[] = {
+	SND_SOC_DAPM_HP("Headphones", NULL),
+	SND_SOC_DAPM_MIC("AMIC", NULL),
+	SND_SOC_DAPM_SPK("Speaker", NULL),
+};
+
+static const struct snd_soc_dapm_route rt700_map[] = {
+	/* Headphones */
+	{ "Headphones", NULL, "HP" },
+	{ "Speaker", NULL, "SPK" },
+	{ "MIC2", NULL, "AMIC" },
+};
+
+static const struct snd_kcontrol_new rt700_controls[] = {
+	SOC_DAPM_PIN_SWITCH("Headphones"),
+	SOC_DAPM_PIN_SWITCH("AMIC"),
+	SOC_DAPM_PIN_SWITCH("Speaker"),
+};
+
+static struct snd_soc_jack_pin rt700_jack_pins[] = {
+	{
+		.pin    = "Headphones",
+		.mask   = SND_JACK_HEADPHONE,
+	},
+	{
+		.pin    = "AMIC",
+		.mask   = SND_JACK_MICROPHONE,
+	},
+};
+
+static int rt700_rtd_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_card *card = rtd->card;
+	struct mc_private *ctx = snd_soc_card_get_drvdata(card);
+	struct snd_soc_component *component = rtd->codec_dai->component;
+	struct snd_soc_jack *jack;
+	int ret;
+
+	card->components = devm_kasprintf(card->dev, GFP_KERNEL,
+					  "%s hs:rt700",
+					  card->components);
+	if (!card->components)
+		return -ENOMEM;
+
+	ret = snd_soc_add_card_controls(card, rt700_controls,
+					ARRAY_SIZE(rt700_controls));
+	if (ret) {
+		dev_err(card->dev, "rt700 controls addition failed: %d\n", ret);
+		return ret;
+	}
+
+	ret = snd_soc_dapm_new_controls(&card->dapm, rt700_widgets,
+					ARRAY_SIZE(rt700_widgets));
+	if (ret) {
+		dev_err(card->dev, "rt700 widgets addition failed: %d\n", ret);
+		return ret;
+	}
+
+	ret = snd_soc_dapm_add_routes(&card->dapm, rt700_map,
+				      ARRAY_SIZE(rt700_map));
+
+	if (ret) {
+		dev_err(card->dev, "rt700 map addition failed: %d\n", ret);
+		return ret;
+	}
+
+	ret = snd_soc_card_jack_new(rtd->card, "Headset Jack",
+				    SND_JACK_HEADSET | SND_JACK_BTN_0 |
+				    SND_JACK_BTN_1 | SND_JACK_BTN_2 |
+				    SND_JACK_BTN_3,
+				    &ctx->sdw_headset,
+				    rt700_jack_pins,
+				    ARRAY_SIZE(rt700_jack_pins));
+	if (ret) {
+		dev_err(rtd->card->dev, "Headset Jack creation failed: %d\n",
+			ret);
+		return ret;
+	}
+
+	jack = &ctx->sdw_headset;
+
+	snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_VOLUMEUP);
+	snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_PLAYPAUSE);
+	snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN);
+	snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND);
+
+	ret = snd_soc_component_set_jack(component, jack, NULL);
+	if (ret)
+		dev_err(rtd->card->dev, "Headset Jack call-back failed: %d\n",
+			ret);
+
+	return ret;
+}
+
+int sof_sdw_rt700_init(const struct snd_soc_acpi_link_adr *link,
+		       struct snd_soc_dai_link *dai_links,
+		       struct sof_sdw_codec_info *info,
+		       bool playback)
+{
+	/*
+	 * headset should be initialized once.
+	 * Do it with dai link for playback.
+	 */
+	if (!playback)
+		return 0;
+
+	dai_links->init = rt700_rtd_init;
+
+	return 0;
+}

sound/soc/intel/boards/sof_sdw_rt711.c

···
+// SPDX-License-Identifier: GPL-2.0
+// Copyright (c) 2020 Intel Corporation
+
+/*
+ *  sof_sdw_rt711 - Helpers to handle RT711 from generic machine driver
+ */
+
+#include <linux/device.h>
+#include <linux/errno.h>
+#include <linux/input.h>
+#include <linux/soundwire/sdw.h>
+#include <linux/soundwire/sdw_type.h>
+#include <sound/soc.h>
+#include <sound/soc-acpi.h>
+#include <sound/jack.h>
+#include "sof_sdw_common.h"
+
+/*
+ * Note this MUST be called before snd_soc_register_card(), so that the props
+ * are in place before the codec component driver's probe function parses them.
+ */
+static int rt711_add_codec_device_props(const char *sdw_dev_name)
+{
+	struct property_entry props[MAX_NO_PROPS] = {};
+	struct device *sdw_dev;
+	int ret;
+
+	sdw_dev = bus_find_device_by_name(&sdw_bus_type, NULL, sdw_dev_name);
+	if (!sdw_dev)
+		return -EPROBE_DEFER;
+
+	if (SOF_RT711_JDSRC(sof_sdw_quirk)) {
+		props[0] = PROPERTY_ENTRY_U32("realtek,jd-src",
+					      SOF_RT711_JDSRC(sof_sdw_quirk));
+	}
+
+	ret = device_add_properties(sdw_dev, props);
+	put_device(sdw_dev);
+
+	return ret;
+}
+
+static const struct snd_soc_dapm_widget rt711_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone", NULL),
+	SND_SOC_DAPM_MIC("Headset Mic", NULL),
+};
+
+static const struct snd_soc_dapm_route rt711_map[] = {
+	/* Headphones */
+	{ "Headphone", NULL, "rt711 HP" },
+	{ "rt711 MIC2", NULL, "Headset Mic" },
+};
+
+static const struct snd_kcontrol_new rt711_controls[] = {
+	SOC_DAPM_PIN_SWITCH("Headphone"),
+	SOC_DAPM_PIN_SWITCH("Headset Mic"),
+};
+
+static struct snd_soc_jack_pin rt711_jack_pins[] = {
+	{
+		.pin    = "Headphone",
+		.mask   = SND_JACK_HEADPHONE,
+	},
+	{
+		.pin    = "Headset Mic",
+		.mask   = SND_JACK_MICROPHONE,
+	},
+};
+
+static int rt711_rtd_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_card *card = rtd->card;
+	struct mc_private *ctx = snd_soc_card_get_drvdata(card);
+	struct snd_soc_component *component = rtd->codec_dai->component;
+	struct snd_soc_jack *jack;
+	int ret;
+
+	card->components = devm_kasprintf(card->dev, GFP_KERNEL,
+					  "%s hs:rt711",
+					  card->components);
+	if (!card->components)
+		return -ENOMEM;
+
+	ret = snd_soc_add_card_controls(card, rt711_controls,
+					ARRAY_SIZE(rt711_controls));
+	if (ret) {
+		dev_err(card->dev, "rt711 controls addition failed: %d\n", ret);
+		return ret;
+	}
+
+	ret = snd_soc_dapm_new_controls(&card->dapm, rt711_widgets,
+					ARRAY_SIZE(rt711_widgets));
+	if (ret) {
+		dev_err(card->dev, "rt711 widgets addition failed: %d\n", ret);
+		return ret;
+	}
+
+	ret = snd_soc_dapm_add_routes(&card->dapm, rt711_map,
+				      ARRAY_SIZE(rt711_map));
+
+	if (ret) {
+		dev_err(card->dev, "rt711 map addition failed: %d\n", ret);
+		return ret;
+	}
+
+	ret = snd_soc_card_jack_new(rtd->card, "Headset Jack",
+				    SND_JACK_HEADSET | SND_JACK_BTN_0 |
+				    SND_JACK_BTN_1 | SND_JACK_BTN_2 |
+				    SND_JACK_BTN_3,
+				    &ctx->sdw_headset,
+				    rt711_jack_pins,
+				    ARRAY_SIZE(rt711_jack_pins));
+	if (ret) {
+		dev_err(rtd->card->dev, "Headset Jack creation failed: %d\n",
+			ret);
+		return ret;
+	}
+
+	jack = &ctx->sdw_headset;
+
+	snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_VOLUMEUP);
+	snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_PLAYPAUSE);
+	snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN);
+	snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND);
+
+	ret = snd_soc_component_set_jack(component, jack, NULL);
+
+	if (ret)
+		dev_err(rtd->card->dev, "Headset Jack call-back failed: %d\n",
+			ret);
+
+	return ret;
+}
+
+int sof_sdw_rt711_init(const struct snd_soc_acpi_link_adr *link,
+		       struct snd_soc_dai_link *dai_links,
+		       struct sof_sdw_codec_info *info,
+		       bool playback)
+{
+	int ret;
+
+	/*
+	 * headset should be initialized once.
+	 * Do it with dai link for playback.
+	 */
+	if (!playback)
+		return 0;
+
+	ret = rt711_add_codec_device_props("sdw:0:25d:711:0");
+	if (ret < 0)
+		return ret;
+
+	dai_links->init = rt711_rtd_init;
+
+	return 0;
+}

sound/soc/intel/boards/sof_sdw_rt715.c

···
+// SPDX-License-Identifier: GPL-2.0
+// Copyright (c) 2020 Intel Corporation
+
+/*
+ *  sof_sdw_rt715 - Helpers to handle RT715 from generic machine driver
+ */
+
+#include <linux/device.h>
+#include <linux/errno.h>
+#include <sound/soc.h>
+#include <sound/soc-acpi.h>
+#include "sof_sdw_common.h"
+
+static int rt715_rtd_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_card *card = rtd->card;
+
+	card->components = devm_kasprintf(card->dev, GFP_KERNEL,
+					  "%s mic:rt715",
+					  card->components);
+	if (!card->components)
+		return -ENOMEM;
+
+	return 0;
+}
+
+int sof_sdw_rt715_init(const struct snd_soc_acpi_link_adr *link,
+		       struct snd_soc_dai_link *dai_links,
+		       struct sof_sdw_codec_info *info,
+		       bool playback)
+{
+	/*
+	 * DAI ID is fixed at SDW_DMIC_DAI_ID for 715 to
+	 * keep sdw DMIC and HDMI setting static in UCM
+	 */
+	if (sof_sdw_quirk & SOF_RT715_DAI_ID_FIX)
+		dai_links->id = SDW_DMIC_DAI_ID;
+
+	dai_links->init = rt715_rtd_init;
+
+	return 0;
+}

sound/soc/intel/common/soc-acpi-intel-bxt-match.c

···
 	},
 	{
 		.id = "104C5122",
-		.drv_name = "bxt-pcm512x",
+		.drv_name = "sof_pcm512x",
 		.sof_fw_filename = "sof-apl.ri",
 		.sof_tplg_filename = "sof-apl-pcm512x.tplg",
 	},

sound/soc/intel/common/soc-acpi-intel-cht-match.c

···
 		.sof_fw_filename = "sof-cht.ri",
 		.sof_tplg_filename = "sof-cht-cx2072x.tplg",
 	},
+	{
+		.id = "104C5122",
+		.drv_name = "sof_pcm512x",
+		.sof_fw_filename = "sof-cht.ri",
+		.sof_tplg_filename = "sof-cht-src-50khz-pcm512x.tplg",
+	},
+
 #if IS_ENABLED(CONFIG_SND_SOC_INTEL_BYT_CHT_NOCODEC_MACH)
 	/*
 	 * This is always last in the table so that it is selected only when

sound/soc/intel/common/soc-acpi-intel-cml-match.c

···
 };
 EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_cml_machines);
 
-static const u64 rt711_0_adr[] = {
-	0x000010025D071100
+static const struct snd_soc_acpi_endpoint single_endpoint = {
+	.num = 0,
+	.aggregated = 0,
+	.group_position = 0,
+	.group_id = 0,
 };
 
-static const u64 rt1308_1_adr[] = {
-	0x000110025D130800
+static const struct snd_soc_acpi_endpoint spk_l_endpoint = {
+	.num = 0,
+	.aggregated = 1,
+	.group_position = 0,
+	.group_id = 1,
 };
 
-static const u64 rt1308_2_adr[] = {
-	0x000210025D130800
+static const struct snd_soc_acpi_endpoint spk_r_endpoint = {
+	.num = 0,
+	.aggregated = 1,
+	.group_position = 1,
+	.group_id = 1,
 };
 
-static const u64 rt715_3_adr[] = {
-	0x000310025D071500
+static const struct snd_soc_acpi_adr_device rt700_1_adr[] = {
+	{
+		.adr = 0x000110025D070000,
+		.num_endpoints = 1,
+		.endpoints = &single_endpoint,
+	}
+};
+
+static const struct snd_soc_acpi_link_adr cml_rvp[] = {
+	{
+		.mask = BIT(1),
+		.num_adr = ARRAY_SIZE(rt700_1_adr),
+		.adr_d = rt700_1_adr,
+	},
+	{}
+};
+
+static const struct snd_soc_acpi_adr_device rt711_0_adr[] = {
+	{
+		.adr = 0x000010025D071100,
+		.num_endpoints = 1,
+		.endpoints = &single_endpoint,
+	}
+};
+
+static const struct snd_soc_acpi_adr_device rt1308_1_adr[] = {
+	{
+		.adr = 0x000110025D130800,
+		.num_endpoints = 1,
+		.endpoints = &single_endpoint,
+	}
+};
+
+static const struct snd_soc_acpi_adr_device rt1308_2_adr[] = {
+	{
+		.adr = 0x000210025D130800,
+		.num_endpoints = 1,
+		.endpoints = &single_endpoint,
+	}
+};
+
+static const struct snd_soc_acpi_adr_device rt1308_1_group1_adr[] = {
+	{
+		.adr = 0x000110025D130800,
+		.num_endpoints = 1,
+		.endpoints = &spk_l_endpoint,
+	}
+};
+
+static const struct snd_soc_acpi_adr_device rt1308_2_group1_adr[] = {
+	{
+		.adr = 0x000210025D130800,
+		.num_endpoints = 1,
+		.endpoints = &spk_r_endpoint,
+	}
+};
+
+static const struct snd_soc_acpi_adr_device rt715_3_adr[] = {
+	{
+		.adr = 0x000310025D071500,
+		.num_endpoints = 1,
+		.endpoints = &single_endpoint,
+	}
 };
 
 static const struct snd_soc_acpi_link_adr cml_3_in_1_default[] = {
 	{
 		.mask = BIT(0),
 		.num_adr = ARRAY_SIZE(rt711_0_adr),
-		.adr = rt711_0_adr,
+		.adr_d = rt711_0_adr,
 	},
 	{
 		.mask = BIT(1),
-		.num_adr = ARRAY_SIZE(rt1308_1_adr),
-		.adr = rt1308_1_adr,
+		.num_adr = ARRAY_SIZE(rt1308_1_group1_adr),
+		.adr_d = rt1308_1_group1_adr,
 	},
 	{
 		.mask = BIT(2),
-		.num_adr = ARRAY_SIZE(rt1308_2_adr),
-		.adr = rt1308_2_adr,
+		.num_adr = ARRAY_SIZE(rt1308_2_group1_adr),
+		.adr_d = rt1308_2_group1_adr,
 	},
 	{
 		.mask = BIT(3),
 		.num_adr = ARRAY_SIZE(rt715_3_adr),
-		.adr = rt715_3_adr,
+		.adr_d = rt715_3_adr,
 	},
 	{}
 };
···
 	{
 		.mask = BIT(0),
 		.num_adr = ARRAY_SIZE(rt711_0_adr),
-		.adr = rt711_0_adr,
+		.adr_d = rt711_0_adr,
 	},
 	{
 		.mask = BIT(1),
 		.num_adr = ARRAY_SIZE(rt1308_1_adr),
-		.adr = rt1308_1_adr,
+		.adr_d = rt1308_1_adr,
 	},
 	{
 		.mask = BIT(3),
 		.num_adr = ARRAY_SIZE(rt715_3_adr),
-		.adr = rt715_3_adr,
+		.adr_d = rt715_3_adr,
 	},
 	{}
 };
···
 	{
 		.link_mask = 0xF, /* 4 active links required */
 		.links = cml_3_in_1_default,
-		.drv_name = "sdw_rt711_rt1308_rt715",
+		.drv_name = "sof_sdw",
 		.sof_fw_filename = "sof-cml.ri",
 		.sof_tplg_filename = "sof-cml-rt711-rt1308-rt715.tplg",
 	},
···
 		 */
 		.link_mask = 0xF,
 		.links = cml_3_in_1_mono_amp,
-		.drv_name = "sdw_rt711_rt1308_rt715",
+		.drv_name = "sof_sdw",
 		.sof_fw_filename = "sof-cml.ri",
 		.sof_tplg_filename = "sof-cml-rt711-rt1308-mono-rt715.tplg",
 	},
 	{
 		.link_mask = 0x2, /* RT700 connected on Link1 */
-		.drv_name = "sdw_rt700",
+		.links = cml_rvp,
+		.drv_name = "sof_sdw",
 		.sof_fw_filename = "sof-cml.ri",
 		.sof_tplg_filename = "sof-cml-rt700.tplg",
 	},

sound/soc/intel/common/soc-acpi-intel-icl-match.c

···
 };
 EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_icl_machines);
 
-static const u64 rt700_0_adr[] = {
-	0x000010025D070000
+static const struct snd_soc_acpi_endpoint single_endpoint = {
+	.num = 0,
+	.aggregated = 0,
+	.group_position = 0,
+	.group_id = 0,
+};
+
+static const struct snd_soc_acpi_endpoint spk_l_endpoint = {
+	.num = 0,
+	.aggregated = 1,
+	.group_position = 0,
+	.group_id = 1,
+};
+
+static const struct snd_soc_acpi_endpoint spk_r_endpoint = {
+	.num = 0,
+	.aggregated = 1,
+	.group_position = 1,
+	.group_id = 1,
+};
+
+static const struct snd_soc_acpi_adr_device rt700_0_adr[] = {
+	{
+		.adr = 0x000010025D070000,
+		.num_endpoints = 1,
+		.endpoints = &single_endpoint,
+	}
 };
 
 static const struct snd_soc_acpi_link_adr icl_rvp[] = {
 	{
 		.mask = BIT(0),
 		.num_adr = ARRAY_SIZE(rt700_0_adr),
-		.adr = rt700_0_adr,
+		.adr_d = rt700_0_adr,
 	},
 	{}
 };
 
-static const u64 rt711_0_adr[] = {
-	0x000010025D071100
+static const struct snd_soc_acpi_adr_device rt711_0_adr[] = {
+	{
+		.adr = 0x000010025D071100,
+		.num_endpoints = 1,
+		.endpoints = &single_endpoint,
+	}
 };
 
-static const u64 rt1308_1_adr[] = {
-	0x000110025D130800
+static const struct snd_soc_acpi_adr_device rt1308_1_adr[] = {
+	{
+		.adr = 0x000110025D130800,
+		.num_endpoints = 1,
+		.endpoints = &single_endpoint,
+	}
 };
 
-static const u64 rt1308_2_adr[] = {
-	0x000210025D130800
+static const struct snd_soc_acpi_adr_device rt1308_2_adr[] = {
+	{
+		.adr = 0x000210025D130800,
+		.num_endpoints = 1,
+		.endpoints = &single_endpoint,
+	}
 };
 
-static const u64 rt715_3_adr[] = {
-	0x000310025D071500
+static const struct snd_soc_acpi_adr_device rt1308_1_group1_adr[] = {
+	{
+		.adr = 0x000110025D130800,
+		.num_endpoints = 1,
+		.endpoints = &spk_l_endpoint,
+	}
+};
+
+static const struct snd_soc_acpi_adr_device rt1308_2_group1_adr[] = {
+	{
+		.adr = 0x000210025D130800,
+		.num_endpoints = 1,
+		.endpoints = &spk_r_endpoint,
+	}
+};
+
+static const struct snd_soc_acpi_adr_device rt715_3_adr[] = {
+	{
+		.adr = 0x000310025D071500,
+		.num_endpoints = 1,
+		.endpoints = &single_endpoint,
+	}
 };
 
 static const struct snd_soc_acpi_link_adr icl_3_in_1_default[] = {
 	{
 		.mask = BIT(0),
 		.num_adr = ARRAY_SIZE(rt711_0_adr),
-		.adr = rt711_0_adr,
+		.adr_d = rt711_0_adr,
 	},
 	{
 		.mask = BIT(1),
-		.num_adr = ARRAY_SIZE(rt1308_1_adr),
-		.adr = rt1308_1_adr,
+		.num_adr = ARRAY_SIZE(rt1308_1_group1_adr),
+		.adr_d = rt1308_1_group1_adr,
 	},
 	{
 		.mask = BIT(2),
-		.num_adr = ARRAY_SIZE(rt1308_2_adr),
-		.adr = rt1308_2_adr,
+		.num_adr = ARRAY_SIZE(rt1308_2_group1_adr),
+		.adr_d = rt1308_2_group1_adr,
 	},
 	{
 		.mask = BIT(3),
 		.num_adr = ARRAY_SIZE(rt715_3_adr),
-		.adr = rt715_3_adr,
+		.adr_d = rt715_3_adr,
 	},
 	{}
 };
···
 	{
 		.mask = BIT(0),
 		.num_adr = ARRAY_SIZE(rt711_0_adr),
-		.adr = rt711_0_adr,
+		.adr_d = rt711_0_adr,
 	},
 	{
 		.mask = BIT(1),
 		.num_adr = ARRAY_SIZE(rt1308_1_adr),
-		.adr = rt1308_1_adr,
+		.adr_d = rt1308_1_adr,
 	},
 	{
 		.mask = BIT(3),
 		.num_adr = ARRAY_SIZE(rt715_3_adr),
-		.adr = rt715_3_adr,
+		.adr_d = rt715_3_adr,
 	},
 	{}
 };
···
 	{
 		.link_mask = 0xF, /* 4 active links required */
 		.links = icl_3_in_1_default,
-		.drv_name = "sdw_rt711_rt1308_rt715",
+		.drv_name = "sof_sdw",
 		.sof_fw_filename = "sof-icl.ri",
 		.sof_tplg_filename = "sof-icl-rt711-rt1308-rt715.tplg",
 	},
 	{
 		.link_mask = 0xB, /* 3 active links required */
 		.links = icl_3_in_1_mono_amp,
-		.drv_name = "sdw_rt711_rt1308_rt715",
+		.drv_name = "sof_sdw",
 		.sof_fw_filename = "sof-icl.ri",
 		.sof_tplg_filename = "sof-icl-rt711-rt1308-rt715-mono.tplg",
 	},
 	{
 		.link_mask = 0x1, /* rt700 connected on link0 */
 		.links = icl_rvp,
-		.drv_name = "sdw_rt700",
+		.drv_name = "sof_sdw",
 		.sof_fw_filename = "sof-icl.ri",
 		.sof_tplg_filename = "sof-icl-rt700.tplg",
 	},

sound/soc/intel/common/soc-acpi-intel-jsl-match.c

···
 /*
  * soc-apci-intel-jsl-match.c - tables and support for JSL ACPI enumeration.
  *
- * Copyright (c) 2019, Intel Corporation.
+ * Copyright (c) 2019-2020, Intel Corporation.
  *
  */
 
 #include <sound/soc-acpi.h>
 #include <sound/soc-acpi-intel-match.h>
 
+static struct snd_soc_acpi_codecs jsl_7219_98373_codecs = {
+	.num_codecs = 1,
+	.codecs = {"MX98373"}
+};
+
+static struct snd_soc_acpi_codecs rt1015_spk = {
+	.num_codecs = 1,
+	.codecs = {"10EC1015"}
+};
+
+/*
+ * When adding new entry to the snd_soc_acpi_intel_jsl_machines array,
+ * use .quirk_data member to distinguish different machine driver,
+ * and keep ACPI .id field unchanged for the common codec.
+ */
 struct snd_soc_acpi_mach snd_soc_acpi_intel_jsl_machines[] = {
 	{
 		.id = "DLGS7219",
 		.drv_name = "sof_da7219_max98373",
-		.machine_quirk = snd_soc_acpi_codec_list,
 		.sof_fw_filename = "sof-jsl.ri",
 		.sof_tplg_filename = "sof-jsl-da7219.tplg",
+		.machine_quirk = snd_soc_acpi_codec_list,
+		.quirk_data = &jsl_7219_98373_codecs,
+	},
+	{
+		.id = "DLGS7219",
+		.drv_name = "sof_da7219_max98360a",
+		.sof_fw_filename = "sof-jsl.ri",
+		.sof_tplg_filename = "sof-jsl-da7219-mx98360a.tplg",
+	},
+	{
+		.id = "10EC5682",
+		.drv_name = "jsl_rt5682_rt1015",
+		.sof_fw_filename = "sof-jsl.ri",
+		.machine_quirk = snd_soc_acpi_codec_list,
+		.quirk_data = &rt1015_spk,
+		.sof_tplg_filename = "sof-jsl-rt5682-rt1015.tplg",
 	},
 	{},
 };

sound/soc/intel/common/soc-acpi-intel-tgl-match.c

···
 	.codecs = {"MX98357A"}
 };
 
-static const u64 rt711_0_adr[] = {
-	0x000010025D071100
+static const struct snd_soc_acpi_endpoint single_endpoint = {
+	.num = 0,
+	.aggregated = 0,
+	.group_position = 0,
+	.group_id = 0,
 };
 
-static const u64 rt1308_1_adr[] = {
-	0x000120025D130800,
-	0x000122025D130800
+static const struct snd_soc_acpi_endpoint spk_l_endpoint = {
+	.num = 0,
+	.aggregated = 1,
+	.group_position = 0,
+	.group_id = 1,
+};
+
+static const struct snd_soc_acpi_endpoint spk_r_endpoint = {
+	.num = 0,
+	.aggregated = 1,
+	.group_position = 1,
+	.group_id = 1,
+};
+
+static const struct snd_soc_acpi_adr_device rt711_0_adr[] = {
+	{
+		.adr = 0x000010025D071100,
+		.num_endpoints = 1,
+		.endpoints = &single_endpoint,
+	}
+};
+
+static const struct snd_soc_acpi_adr_device rt1308_1_adr[] = {
+	{
+		.adr = 0x000120025D130800,
+		.num_endpoints = 1,
+		.endpoints = &spk_l_endpoint,
+	},
+	{
+		.adr = 0x000122025D130800,
+		.num_endpoints = 1,
+		.endpoints = &spk_r_endpoint,
+	}
+};
+
+static const struct snd_soc_acpi_adr_device rt5682_0_adr[] = {
+	{
+		.adr = 0x000021025D568200,
+		.num_endpoints = 1,
+		.endpoints = &single_endpoint,
+	}
 };
 
 static const struct snd_soc_acpi_link_adr tgl_i2s_rt1308[] = {
 	{
 		.mask = BIT(0),
 		.num_adr = ARRAY_SIZE(rt711_0_adr),
-		.adr = rt711_0_adr,
+		.adr_d = rt711_0_adr,
 	},
 	{}
 };
···
 	{
 		.mask = BIT(0),
 		.num_adr = ARRAY_SIZE(rt711_0_adr),
-		.adr = rt711_0_adr,
+		.adr_d = rt711_0_adr,
 	},
 	{
 		.mask = BIT(1),
 		.num_adr = ARRAY_SIZE(rt1308_1_adr),
-		.adr = rt1308_1_adr,
+		.adr_d = rt1308_1_adr,
 	},
 	{}
+};
+
+static const struct snd_soc_acpi_link_adr tgl_chromebook_base[] = {
+	{
+		.mask = BIT(0),
+		.num_adr = ARRAY_SIZE(rt5682_0_adr),
+		.adr_d = rt5682_0_adr,
+	},
+	{}
+};
+
+static struct snd_soc_acpi_codecs tgl_max98373_amp = {
+	.num_codecs = 1,
+	.codecs = {"MX98373"}
 };
 
 struct snd_soc_acpi_mach snd_soc_acpi_intel_tgl_machines[] = {
 	{
 		.id = "10EC1308",
-		.drv_name = "rt711_rt1308",
+		.drv_name = "sof_sdw",
 		.link_mask = 0x1, /* RT711 on SoundWire link0 */
 		.links = tgl_i2s_rt1308,
 		.sof_fw_filename = "sof-tgl.ri",
-		.sof_tplg_filename = "sof-tgl-rt711-rt1308.tplg",
+		.sof_tplg_filename = "sof-tgl-rt711-i2s-rt1308.tplg",
 	},
 	{
 		.id = "10EC5682",
···
 		.quirk_data = &tgl_codecs,
 		.sof_fw_filename = "sof-tgl.ri",
 		.sof_tplg_filename = "sof-tgl-max98357a-rt5682.tplg",
+	},
+	{
+		.id = "10EC5682",
+		.drv_name = "tgl_max98373_rt5682",
+		.machine_quirk = snd_soc_acpi_codec_list,
+		.quirk_data = &tgl_max98373_amp,
+		.sof_fw_filename = "sof-tgl.ri",
+		.sof_tplg_filename = "sof-tgl-max98373-rt5682.tplg",
 	},
 	{},
 };
···
 	{
 		.link_mask = 0x3, /* rt711 on link 0 and 2 rt1308s on link 1 */
 		.links = tgl_rvp,
-		.drv_name = "sdw_rt711_rt1308_rt715",
+		.drv_name = "sof_sdw",
 		.sof_fw_filename = "sof-tgl.ri",
 		.sof_tplg_filename = "sof-tgl-rt711-rt1308.tplg",
+	},
+	{
+		.link_mask = 0x1, /* this will only enable rt5682 for now */
+		.links = tgl_chromebook_base,
+		.drv_name = "sof_sdw",
+		.sof_fw_filename = "sof-tgl.ri",
+		.sof_tplg_filename = "sof-tgl-rt5682.tplg",
 	},
 	{},
 };

sound/soc/intel/haswell/sst-haswell-pcm.c

···
 	u8 channels;
 	int ret, dai;
 
-	dai = mod_map[rtd->cpu_dai->id].dai_id;
+	dai = mod_map[asoc_rtd_to_cpu(rtd, 0)->id].dai_id;
 	pcm_data = &pdata->pcm[dai][substream->stream];
 
 	/* check if we are being called a subsequent time */
···
 		}
 		pcm_data->allocated = false;
 
-		pcm_data->stream = sst_hsw_stream_new(hsw, rtd->cpu_dai->id,
+		pcm_data->stream = sst_hsw_stream_new(hsw, asoc_rtd_to_cpu(rtd, 0)->id,
 			hsw_notify_pointer, pcm_data);
 		if (pcm_data->stream == NULL) {
 			dev_err(rtd->dev, "error: failed to create stream\n");
···
 		path_id = SST_HSW_STREAM_PATH_SSP0_IN;
 
 	/* DSP stream type depends on DAI ID */
-	switch (rtd->cpu_dai->id) {
+	switch (asoc_rtd_to_cpu(rtd, 0)->id) {
 	case 0:
 		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 			stream_type = SST_HSW_STREAM_TYPE_SYSTEM;
···
 		break;
 	default:
 		dev_err(rtd->dev, "error: invalid DAI ID %d\n",
-			rtd->cpu_dai->id);
+			asoc_rtd_to_cpu(rtd, 0)->id);
 		return -EINVAL;
 	}
 
···
 	dmab = snd_pcm_get_dma_buf(substream);
 
 	ret = create_adsp_page_table(substream, pdata, rtd, runtime->dma_area,
-		runtime->dma_bytes, rtd->cpu_dai->id);
+		runtime->dma_bytes, asoc_rtd_to_cpu(rtd, 0)->id);
 	if (ret < 0)
 		return ret;
 
···
 		pages = runtime->dma_bytes / PAGE_SIZE;
 
 	ret = sst_hsw_stream_buffer(hsw, pcm_data->stream,
-		pdata->dmab[rtd->cpu_dai->id][substream->stream].addr,
+		pdata->dmab[asoc_rtd_to_cpu(rtd, 0)->id][substream->stream].addr,
 		pages, runtime->dma_bytes, 0,
 		snd_sgbuf_get_addr(dmab, 0) >> PAGE_SHIFT);
 	if (ret < 0) {
···
 	snd_pcm_uframes_t pos;
 	int dai;
 
-	dai = mod_map[rtd->cpu_dai->id].dai_id;
+	dai = mod_map[asoc_rtd_to_cpu(rtd, 0)->id].dai_id;
 	pcm_data = &pdata->pcm[dai][substream->stream];
 	sst_stream = pcm_data->stream;
 
···
 	u32 position;
 	int dai;
 
-	dai = mod_map[rtd->cpu_dai->id].dai_id;
+	dai = mod_map[asoc_rtd_to_cpu(rtd, 0)->id].dai_id;
 	pcm_data = &pdata->pcm[dai][substream->stream];
 	position = sst_hsw_get_dsp_position(hsw, pcm_data->stream);
 
···
 	struct sst_hsw *hsw = pdata->hsw;
 	int dai;
 
-	dai = mod_map[rtd->cpu_dai->id].dai_id;
+	dai = mod_map[asoc_rtd_to_cpu(rtd, 0)->id].dai_id;
 	pcm_data = &pdata->pcm[dai][substream->stream];
 
 	mutex_lock(&pcm_data->mutex);
···
 
 	snd_soc_set_runtime_hwparams(substream, &hsw_pcm_hardware);
 
-	pcm_data->stream = sst_hsw_stream_new(hsw, rtd->cpu_dai->id,
+	pcm_data->stream = sst_hsw_stream_new(hsw, asoc_rtd_to_cpu(rtd, 0)->id,
 		hsw_notify_pointer, pcm_data);
 	if (pcm_data->stream == NULL) {
 		dev_err(rtd->dev, "error: failed to create stream\n");
···
 	struct sst_hsw *hsw = pdata->hsw;
 	int ret, dai;
 
-	dai = mod_map[rtd->cpu_dai->id].dai_id;
+	dai = mod_map[asoc_rtd_to_cpu(rtd, 0)->id].dai_id;
 	pcm_data = &pdata->pcm[dai][substream->stream];
 
 	mutex_lock(&pcm_data->mutex);
···
 			hsw_pcm_hardware.buffer_bytes_max);
 	}
 	if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream)
-		priv_data->pcm[rtd->cpu_dai->id][SNDRV_PCM_STREAM_PLAYBACK].hsw_pcm = pcm;
+		priv_data->pcm[asoc_rtd_to_cpu(rtd, 0)->id][SNDRV_PCM_STREAM_PLAYBACK].hsw_pcm = pcm;
 	if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream)
-		priv_data->pcm[rtd->cpu_dai->id][SNDRV_PCM_STREAM_CAPTURE].hsw_pcm = pcm;
+		priv_data->pcm[asoc_rtd_to_cpu(rtd, 0)->id][SNDRV_PCM_STREAM_CAPTURE].hsw_pcm = pcm;
 
 	return 0;
 }

sound/soc/intel/skylake/bxt-sst.c

···
 #include "skl.h"
 
 #define BXT_BASEFW_TIMEOUT	3000
-#define BXT_INIT_TIMEOUT	300
 #define BXT_ROM_INIT_TIMEOUT	70
 #define BXT_IPC_PURGE_FW	0x01004000
 
···
 
 /* Delay before scheduling D0i3 entry */
 #define BXT_D0I3_DELAY 5000
-
-#define BXT_FW_ROM_INIT_RETRY 3
 
 static unsigned int bxt_get_errorcode(struct sst_dsp *ctx)
 {

sound/soc/intel/skylake/cnl-sst.c

···
 	ctx->dsp_ops.stream_tag = stream_tag;
 	memcpy(ctx->dmab.area, fwdata, fwsize);
 
+	ret = skl_dsp_core_power_up(ctx, SKL_DSP_CORE0_MASK);
+	if (ret < 0) {
+		dev_err(ctx->dev, "dsp core0 power up failed\n");
+		ret = -EIO;
+		goto base_fw_load_failed;
+	}
+
 	/* purge FW request */
 	sst_dsp_shim_write(ctx, CNL_ADSP_REG_HIPCIDR,
 			   CNL_ADSP_REG_HIPCIDR_BUSY | (CNL_IPC_PURGE |
 			   ((stream_tag - 1) << CNL_ROM_CTRL_DMA_ID)));
 
-	ret = cnl_dsp_enable_core(ctx, SKL_DSP_CORE0_MASK);
+	ret = skl_dsp_start_core(ctx, SKL_DSP_CORE0_MASK);
 	if (ret < 0) {
-		dev_err(ctx->dev, "dsp boot core failed ret: %d\n", ret);
+		dev_err(ctx->dev, "Start dsp core failed ret: %d\n", ret);
 		ret = -EIO;
+		goto base_fw_load_failed;
+	}
+
+	ret = sst_dsp_register_poll(ctx, CNL_ADSP_REG_HIPCIDA,
+				    CNL_ADSP_REG_HIPCIDA_DONE,
+				    CNL_ADSP_REG_HIPCIDA_DONE,
+				    BXT_INIT_TIMEOUT, "HIPCIDA Done");
+	if (ret < 0) {
+		dev_err(ctx->dev, "timeout for purge request: %d\n", ret);
 		goto base_fw_load_failed;
 	}
 
···
 {
 	struct firmware stripped_fw;
 	struct skl_dev *cnl = ctx->thread_context;
-	int ret;
+	int ret, i;
 
 	if (!ctx->fw) {
 		ret = request_firmware(&ctx->fw, ctx->fw_name, ctx->dev);
···
 	stripped_fw.size = ctx->fw->size;
 	skl_dsp_strip_extended_manifest(&stripped_fw);
 
-	ret = cnl_prepare_fw(ctx, stripped_fw.data, stripped_fw.size);
-	if (ret < 0) {
-		dev_err(ctx->dev, "prepare firmware failed: %d\n", ret);
-		goto cnl_load_base_firmware_failed;
+	for (i = 0; i < BXT_FW_ROM_INIT_RETRY; i++) {
+		ret = cnl_prepare_fw(ctx, stripped_fw.data, stripped_fw.size);
+		if (!ret)
+			break;
+		dev_dbg(ctx->dev, "prepare firmware failed: %d\n", ret);
 	}
+
+	if (ret < 0)
+		goto cnl_load_base_firmware_failed;
 
 	ret = sst_transfer_fw_host_dma(ctx);
 	if (ret < 0) {
···
 	return 0;
 
 cnl_load_base_firmware_failed:
+	dev_err(ctx->dev, "firmware load failed: %d\n", ret);
 	release_firmware(ctx->fw);
 	ctx->fw = NULL;
 

sound/soc/intel/skylake/skl-nhlt.c

···
 {
 	struct device *dev = &skl->pci->dev;
 
-	sysfs_remove_file(&dev->kobj, &dev_attr_platform_id.attr);
+	if (skl->nhlt)
+		sysfs_remove_file(&dev->kobj, &dev_attr_platform_id.attr);
 }
 
 /*

sound/soc/intel/skylake/skl-pcm.c

···
 	struct snd_soc_dapm_widget *w;
 	struct skl_dev *skl = bus_to_skl(bus);
 
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		w = dai->playback_widget;
-	else
-		w = dai->capture_widget;
+	w = snd_soc_dai_get_widget(dai, substream->stream);
 
 	if (w->ignore_suspend && enable)
 		skl->supend_active++;
···
 	if (!mconfig)
 		return -EIO;
 
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		w = dai->playback_widget;
-	else
-		w = dai->capture_widget;
+	w = snd_soc_dai_get_widget(dai, substream->stream);
 
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_RESUME:
···
 	struct hdac_bus *bus = dev_get_drvdata(dai->dev);
 	struct hdac_ext_stream *link_dev;
 	struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream);
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	struct skl_pipe_params p_params = {0};
 	struct hdac_ext_link *link;
 	int stream_tag;
···
 
 	link_dev->link_prepared = 0;
 
-	link = snd_hdac_ext_bus_get_link(bus, rtd->codec_dai->component->name);
+	link = snd_hdac_ext_bus_get_link(bus, asoc_rtd_to_codec(rtd, 0)->component->name);
 	if (!link)
 		return -EINVAL;
 
···
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai_link *dai_link = rtd->dai_link;
 
-	dev_dbg(rtd->cpu_dai->dev, "In %s:%s\n", __func__,
+	dev_dbg(asoc_rtd_to_cpu(rtd, 0)->dev, "In %s:%s\n", __func__,
 					dai_link->cpus->dai_name);
 
 	snd_soc_set_runtime_hwparams(substream, &azx_pcm_hw);
···
 				u64 nsec)
 {
 	struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream);
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	u64 codec_frames, codec_nsecs;
 
 	if (!codec_dai->driver->ops->delay)
···
 static int skl_platform_soc_new(struct snd_soc_component *component,
 				struct snd_soc_pcm_runtime *rtd)
 {
-	struct snd_soc_dai *dai = rtd->cpu_dai;
+	struct snd_soc_dai *dai = asoc_rtd_to_cpu(rtd, 0);
 	struct hdac_bus *bus = dev_get_drvdata(dai->dev);
 	struct snd_pcm *pcm = rtd->pcm;
 	unsigned int size;

sound/soc/intel/skylake/skl-sst-dsp.h

···
 
 #define SKL_FW_INIT			0x1
 #define SKL_FW_RFW_START		0xf
+#define BXT_FW_ROM_INIT_RETRY		3
+#define BXT_INIT_TIMEOUT		300
 
 #define SKL_ADSPIC_IPC			1
 #define SKL_ADSPIS_IPC			1

sound/soc/intel/skylake/skl.c

···
 	struct hdac_ext_link *hlink;
 	int ret;
 
+	snd_hdac_set_codec_wakeup(bus, true);
 	skl_enable_miscbdcge(bus->dev, false);
 	ret = snd_hdac_bus_init_chip(bus, full_reset);
 
···
 		writel(0, hlink->ml_addr + AZX_REG_ML_LOSIDV);
 
 	skl_enable_miscbdcge(bus->dev, true);
+	snd_hdac_set_codec_wakeup(bus, false);
 
 	return ret;
 }
···
 	struct pci_dev *pci = to_pci_dev(dev);
 	struct hdac_bus *bus = pci_get_drvdata(pci);
 	struct skl_dev *skl  = bus_to_skl(bus);
-	struct hdac_ext_link *hlink = NULL;
+	struct hdac_ext_link *hlink;
 	int ret;
 
 	/*
···
 static struct snd_soc_acpi_mach *skl_find_hda_machine(struct skl_dev *skl,
 					struct snd_soc_acpi_mach *machines)
 {
-	struct hdac_bus *bus = skl_to_bus(skl);
 	struct snd_soc_acpi_mach *mach;
-
-	/* check if we have any codecs detected on bus */
-	if (bus->codec_mask == 0)
-		return NULL;
 
 	/* point to common table */
 	mach = snd_soc_acpi_intel_hda_machines;
···
 {
 	struct platform_device_info pdevinfo = {NULL};
 	struct skl_clk_pdata *clk_pdata;
+
+	if (!skl->nhlt)
+		return 0;
 
 	clk_pdata = devm_kzalloc(&skl->pci->dev, sizeof(*clk_pdata),
 							GFP_KERNEL);
···
 {
 	struct skl_dev *skl = container_of(work, struct skl_dev, probe_work);
 	struct hdac_bus *bus = skl_to_bus(skl);
-	struct hdac_ext_link *hlink = NULL;
+	struct hdac_ext_link *hlink;
 	int err;
 
 	if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) {
···
 		if (err < 0)
 			return;
 	}
+
+	skl_init_pci(skl);
+	skl_dum_set(bus);
 
 	err = skl_init_chip(bus, true);
 	if (err < 0) {
···
 		return -ENXIO;
 	}
 
-	snd_hdac_bus_reset_link(bus, true);
-
 	snd_hdac_bus_parse_capabilities(bus);
 
 	/* check if PPCAP exists */
···
 	if (err < 0)
 		return err;
 
-	/* initialize chip */
-	skl_init_pci(skl);
-	skl_dum_set(bus);
-
-	return skl_init_chip(bus, true);
+	return 0;
 }
 
 static int skl_probe(struct pci_dev *pci,
···
 	if (bus->mlcap)
 		snd_hdac_ext_bus_get_ml_capabilities(bus);
 
-	snd_hdac_bus_stop_chip(bus);
-
 	/* create device for soc dmic */
 	err = skl_dmic_device_register(skl);
 	if (err < 0) {
···
 out_clk_free:
 	skl_clock_device_unregister(skl);
 out_nhlt_free:
-	intel_nhlt_free(skl->nhlt);
+	if (skl->nhlt)
+		intel_nhlt_free(skl->nhlt);
 out_free:
 	skl_free(bus);
 
···
 	skl_dmic_device_unregister(skl);
 	skl_clock_device_unregister(skl);
 	skl_nhlt_remove_sysfs(skl);
-	intel_nhlt_free(skl->nhlt);
+	if (skl->nhlt)
+		intel_nhlt_free(skl->nhlt);
 	skl_free(bus);
 	dev_set_drvdata(&pci->dev, NULL);
 }

sound/soc/jz4740/jz4740-i2s.c

···
 
 #define JZ_AIC_CONF_FIFO_RX_THRESHOLD_OFFSET 12
 #define JZ_AIC_CONF_FIFO_TX_THRESHOLD_OFFSET 8
-#define JZ4780_AIC_CONF_FIFO_RX_THRESHOLD_OFFSET 24
-#define JZ4780_AIC_CONF_FIFO_TX_THRESHOLD_OFFSET 16
-#define JZ4780_AIC_CONF_FIFO_RX_THRESHOLD_MASK \
-			(0xf << JZ4780_AIC_CONF_FIFO_RX_THRESHOLD_OFFSET)
-#define JZ4780_AIC_CONF_FIFO_TX_THRESHOLD_MASK \
-			(0x1f <<  JZ4780_AIC_CONF_FIFO_TX_THRESHOLD_OFFSET)
+#define JZ4760_AIC_CONF_FIFO_RX_THRESHOLD_OFFSET 24
+#define JZ4760_AIC_CONF_FIFO_TX_THRESHOLD_OFFSET 16
 
 #define JZ_AIC_CTRL_OUTPUT_SAMPLE_SIZE_MASK (0x7 << 19)
 #define JZ_AIC_CTRL_INPUT_SAMPLE_SIZE_MASK (0x7 << 16)
···
 #define JZ_AIC_I2S_STATUS_BUSY BIT(2)
 
 #define JZ_AIC_CLK_DIV_MASK 0xf
-#define I2SDIV_DV_SHIFT 8
+#define I2SDIV_DV_SHIFT 0
 #define I2SDIV_DV_MASK (0xf << I2SDIV_DV_SHIFT)
 #define I2SDIV_IDV_SHIFT 8
 #define I2SDIV_IDV_MASK (0xf << I2SDIV_IDV_SHIFT)
 
 enum jz47xx_i2s_version {
 	JZ_I2S_JZ4740,
+	JZ_I2S_JZ4760,
+	JZ_I2S_JZ4770,
 	JZ_I2S_JZ4780,
+};
+
+struct i2s_soc_info {
+	enum jz47xx_i2s_version version;
+	struct snd_soc_dai_driver *dai;
 };
 
 struct jz4740_i2s {
···
 	struct snd_dmaengine_dai_dma_data playback_dma_data;
 	struct snd_dmaengine_dai_dma_data capture_dma_data;
 
-	enum jz47xx_i2s_version version;
+	const struct i2s_soc_info *soc_info;
 };
 
 static inline uint32_t jz4740_i2s_read(const struct jz4740_i2s *i2s,
···
 		ctrl &= ~JZ_AIC_CTRL_INPUT_SAMPLE_SIZE_MASK;
 		ctrl |= sample_size << JZ_AIC_CTRL_INPUT_SAMPLE_SIZE_OFFSET;
 
-		if (i2s->version >= JZ_I2S_JZ4780) {
+		if (i2s->soc_info->version >= JZ_I2S_JZ4770) {
 			div_reg &= ~I2SDIV_IDV_MASK;
 			div_reg |= (div - 1) << I2SDIV_IDV_SHIFT;
 		} else {
···
 	snd_soc_dai_init_dma_data(dai, &i2s->playback_dma_data,
 		&i2s->capture_dma_data);
 
-	if (i2s->version >= JZ_I2S_JZ4780) {
-		conf = (7 << JZ4780_AIC_CONF_FIFO_RX_THRESHOLD_OFFSET) |
-			(8 << JZ4780_AIC_CONF_FIFO_TX_THRESHOLD_OFFSET) |
+	if (i2s->soc_info->version >= JZ_I2S_JZ4760) {
+		conf = (7 << JZ4760_AIC_CONF_FIFO_RX_THRESHOLD_OFFSET) |
+			(8 << JZ4760_AIC_CONF_FIFO_TX_THRESHOLD_OFFSET) |
 			JZ_AIC_CONF_OVERFLOW_PLAY_LAST |
 			JZ_AIC_CONF_I2S |
 			JZ_AIC_CONF_INTERNAL_CODEC;
···
 	.ops = &jz4740_i2s_dai_ops,
 };
 
-static struct snd_soc_dai_driver jz4780_i2s_dai = {
+static const struct i2s_soc_info jz4740_i2s_soc_info = {
+	.version = JZ_I2S_JZ4740,
+	.dai = &jz4740_i2s_dai,
+};
+
+static const struct i2s_soc_info jz4760_i2s_soc_info = {
+	.version = JZ_I2S_JZ4760,
+	.dai = &jz4740_i2s_dai,
+};
+
+static struct snd_soc_dai_driver jz4770_i2s_dai = {
 	.probe = jz4740_i2s_dai_probe,
 	.remove = jz4740_i2s_dai_remove,
 	.playback = {
···
 	.ops = &jz4740_i2s_dai_ops,
 };
 
+static const struct i2s_soc_info jz4770_i2s_soc_info = {
+	.version = JZ_I2S_JZ4770,
+	.dai = &jz4770_i2s_dai,
+};
+
+static const struct i2s_soc_info jz4780_i2s_soc_info = {
+	.version = JZ_I2S_JZ4780,
+	.dai = &jz4770_i2s_dai,
+};
+
 static const struct snd_soc_component_driver jz4740_i2s_component = {
 	.name		= "jz4740-i2s",
 	.suspend	= jz4740_i2s_suspend,
···
 
 #ifdef CONFIG_OF
 static const struct of_device_id jz4740_of_matches[] = {
-	{ .compatible = "ingenic,jz4740-i2s", .data = (void *)JZ_I2S_JZ4740 },
-	{ .compatible = "ingenic,jz4780-i2s", .data = (void *)JZ_I2S_JZ4780 },
+	{ .compatible = "ingenic,jz4740-i2s", .data = &jz4740_i2s_soc_info },
+	{ .compatible = "ingenic,jz4760-i2s", .data = &jz4760_i2s_soc_info },
+	{ .compatible = "ingenic,jz4770-i2s", .data = &jz4770_i2s_soc_info },
+	{ .compatible = "ingenic,jz4780-i2s", .data = &jz4780_i2s_soc_info },
 	{ /* sentinel */ }
 };
 MODULE_DEVICE_TABLE(of, jz4740_of_matches);
···
 
 static int jz4740_i2s_dev_probe(struct platform_device *pdev)
 {
+	struct device *dev = &pdev->dev;
 	struct jz4740_i2s *i2s;
 	struct resource *mem;
 	int ret;
 
-	i2s = devm_kzalloc(&pdev->dev, sizeof(*i2s), GFP_KERNEL);
+	i2s = devm_kzalloc(dev, sizeof(*i2s), GFP_KERNEL);
 	if (!i2s)
 		return -ENOMEM;
 
-	i2s->version =
-		(enum jz47xx_i2s_version)of_device_get_match_data(&pdev->dev);
+	i2s->soc_info = device_get_match_data(dev);
 
 	mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	i2s->base = devm_ioremap_resource(&pdev->dev, mem);
+	i2s->base = devm_ioremap_resource(dev, mem);
 	if (IS_ERR(i2s->base))
 		return PTR_ERR(i2s->base);
 
 	i2s->phys_base = mem->start;
 
-	i2s->clk_aic = devm_clk_get(&pdev->dev, "aic");
+	i2s->clk_aic = devm_clk_get(dev, "aic");
 	if (IS_ERR(i2s->clk_aic))
 		return PTR_ERR(i2s->clk_aic);
 
-	i2s->clk_i2s = devm_clk_get(&pdev->dev, "i2s");
+	i2s->clk_i2s = devm_clk_get(dev, "i2s");
 	if (IS_ERR(i2s->clk_i2s))
 		return PTR_ERR(i2s->clk_i2s);
 
 	platform_set_drvdata(pdev, i2s);
 
-	if (i2s->version == JZ_I2S_JZ4780)
-		ret = devm_snd_soc_register_component(&pdev->dev,
-			&jz4740_i2s_component, &jz4780_i2s_dai, 1);
-	else
-		ret = devm_snd_soc_register_component(&pdev->dev,
-			&jz4740_i2s_component, &jz4740_i2s_dai, 1);
-
+	ret = devm_snd_soc_register_component(dev, &jz4740_i2s_component,
+					      i2s->soc_info->dai, 1);
 	if (ret)
 		return ret;
 
-	return devm_snd_dmaengine_pcm_register(&pdev->dev, NULL,
+	return devm_snd_dmaengine_pcm_register(dev, NULL,
 		SND_DMAENGINE_PCM_FLAG_COMPAT);
 }
 

sound/soc/kirkwood/armada-370-db.c

···
 			    struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	unsigned int freq;
 
 	switch (params_rate(params)) {

sound/soc/kirkwood/kirkwood-dma.c

···
 static struct kirkwood_dma_data *kirkwood_priv(struct snd_pcm_substream *subs)
 {
 	struct snd_soc_pcm_runtime *soc_runtime = subs->private_data;
-	return snd_soc_dai_get_drvdata(soc_runtime->cpu_dai);
+	return snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(soc_runtime, 0));
 }
 
 static const struct snd_pcm_hardware kirkwood_dma_snd_hw = {

sound/soc/mediatek/common/mtk-afe-fe-dai.c

···
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai);
 	struct snd_pcm_runtime *runtime = substream->runtime;
-	int memif_num = rtd->cpu_dai->id;
+	int memif_num = asoc_rtd_to_cpu(rtd, 0)->id;
 	struct mtk_base_afe_memif *memif = &afe->memif[memif_num];
 	const struct snd_pcm_hardware *mtk_afe_hardware = afe->mtk_afe_hardware;
 	int ret;
···
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai);
-	struct mtk_base_afe_memif *memif = &afe->memif[rtd->cpu_dai->id];
+	struct mtk_base_afe_memif *memif = &afe->memif[asoc_rtd_to_cpu(rtd, 0)->id];
 	int irq_id;
 
 	irq_id = memif->irq_usage;
···
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai);
-	int id = rtd->cpu_dai->id;
+	int id = asoc_rtd_to_cpu(rtd, 0)->id;
 	struct mtk_base_afe_memif *memif = &afe->memif[id];
 	int ret;
 	unsigned int channels = params_channels(params);
···
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_pcm_runtime * const runtime = substream->runtime;
 	struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai);
-	int id = rtd->cpu_dai->id;
+	int id = asoc_rtd_to_cpu(rtd, 0)->id;
 	struct mtk_base_afe_memif *memif = &afe->memif[id];
 	struct mtk_base_afe_irq *irqs = &afe->irqs[memif->irq_usage];
 	const struct mtk_base_irq_data *irq_data = irqs->irq_data;
···
 {
 	struct snd_soc_pcm_runtime *rtd  = substream->private_data;
 	struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai);
-	int id = rtd->cpu_dai->id;
+	int id = asoc_rtd_to_cpu(rtd, 0)->id;
 	int pbuf_size;
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {

sound/soc/mediatek/common/mtk-afe-platform-driver.c

···
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct mtk_base_afe *afe = snd_soc_component_get_drvdata(component);
-	struct mtk_base_afe_memif *memif = &afe->memif[rtd->cpu_dai->id];
+	struct mtk_base_afe_memif *memif = &afe->memif[asoc_rtd_to_cpu(rtd, 0)->id];
 	const struct mtk_base_memif_data *memif_data = memif->data;
 	struct regmap *regmap = afe->regmap;
 	struct device *dev = afe->dev;

sound/soc/mediatek/mt2701/mt2701-afe-pcm.c

···
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	int fs;
 
-	if (rtd->cpu_dai->id != MT2701_MEMIF_ULBT)
+	if (asoc_rtd_to_cpu(rtd, 0)->id != MT2701_MEMIF_ULBT)
 		fs = mt2701_afe_i2s_fs(rate);
 	else
 		fs = (rate == 16000 ? 1 : 0);

sound/soc/mediatek/mt2701/mt2701-cs42448.c

···
 					   struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	unsigned int mclk_rate;
 	unsigned int rate = params_rate(params);
 	unsigned int div_mclk_over_bck = rate > 192000 ? 2 : 4;

sound/soc/mediatek/mt2701/mt2701-wm8960.c

···
 					  struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	unsigned int mclk_rate;
 	unsigned int rate = params_rate(params);
 	unsigned int div_mclk_over_bck = rate > 192000 ? 2 : 4;

sound/soc/mediatek/mt6797/mt6797-afe-pcm.c

···
 	struct snd_soc_component *component =
 		snd_soc_rtdcom_lookup(rtd, AFE_PCM_NAME);
 	struct mtk_base_afe *afe = snd_soc_component_get_drvdata(component);
-	int id = rtd->cpu_dai->id;
+	int id = asoc_rtd_to_cpu(rtd, 0)->id;
 
 	return mt6797_rate_transform(afe->dev, rate, id);
 }

sound/soc/mediatek/mt8173/mt8173-afe-pcm.c

···
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_component *component = snd_soc_rtdcom_lookup(rtd, AFE_PCM_NAME);
 	struct mtk_base_afe *afe = snd_soc_component_get_drvdata(component);
-	struct mtk_base_afe_memif *memif = &afe->memif[rtd->cpu_dai->id];
+	struct mtk_base_afe_memif *memif = &afe->memif[asoc_rtd_to_cpu(rtd, 0)->id];
 	int fs;
 
 	if (memif->data->id == MT8173_AFE_MEMIF_DAI ||

sound/soc/mediatek/mt8173/mt8173-max98090.c

···
 				     struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 
 	return snd_soc_dai_set_sysclk(codec_dai, 0, params_rate(params) * 256,
 				      SND_SOC_CLOCK_IN);
···
 {
 	int ret;
 	struct snd_soc_card *card = runtime->card;
-	struct snd_soc_component *component = runtime->codec_dai->component;
+	struct snd_soc_component *component = asoc_rtd_to_codec(runtime, 0)->component;
 
 	/* enable jack detection */
 	ret = snd_soc_card_jack_new(card, "Headphone", SND_JACK_HEADPHONE,

sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c

···
 	struct snd_soc_dai *codec_dai;
 	int i, ret;
 
-	for_each_rtd_codec_dai(rtd, i, codec_dai) {
+	for_each_rtd_codec_dais(rtd, i, codec_dai) {
 		/* pll from mclk 12.288M */
 		ret = snd_soc_dai_set_pll(codec_dai, 0, 0, MCLK_FOR_CODECS,
 					  params_rate(params) * 512);
···
 static int mt8173_rt5650_rt5514_init(struct snd_soc_pcm_runtime *runtime)
 {
 	struct snd_soc_card *card = runtime->card;
-	struct snd_soc_component *component = runtime->codec_dais[0]->component;
+	struct snd_soc_component *component = asoc_rtd_to_codec(runtime, 0)->component;
 	int ret;
 
 	rt5645_sel_asrc_clk_src(component,

sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c

···
 	struct snd_soc_dai *codec_dai;
 	int i, ret;
 
-	for_each_rtd_codec_dai(rtd, i, codec_dai) {
+	for_each_rtd_codec_dais(rtd, i, codec_dai) {
 		/* pll from mclk 12.288M */
 		ret = snd_soc_dai_set_pll(codec_dai, 0, 0, MCLK_FOR_CODECS,
 					  params_rate(params) * 512);
···
 static int mt8173_rt5650_rt5676_init(struct snd_soc_pcm_runtime *runtime)
 {
 	struct snd_soc_card *card = runtime->card;
-	struct snd_soc_component *component = runtime->codec_dais[0]->component;
-	struct snd_soc_component *component_sub = runtime->codec_dais[1]->component;
+	struct snd_soc_component *component = asoc_rtd_to_codec(runtime, 0)->component;
+	struct snd_soc_component *component_sub = asoc_rtd_to_codec(runtime, 1)->component;
 	int ret;
 
 	rt5645_sel_asrc_clk_src(component,

sound/soc/mediatek/mt8173/mt8173-rt5650.c

···
 #include <linux/of_gpio.h>
 #include <sound/soc.h>
 #include <sound/jack.h>
+#include <sound/hdmi-codec.h>
 #include "../../codecs/rt5645.h"
 
 #define MCLK_FOR_CODECS		12288000
···
 		break;
 	}
 
-	for_each_rtd_codec_dai(rtd, i, codec_dai) {
+	for_each_rtd_codec_dais(rtd, i, codec_dai) {
 		/* pll from mclk */
 		ret = snd_soc_dai_set_pll(codec_dai, 0, 0, mclk_clock,
 					  params_rate(params) * 512);
···
 	.hw_params = mt8173_rt5650_hw_params,
 };
 
-static struct snd_soc_jack mt8173_rt5650_jack;
+static struct snd_soc_jack mt8173_rt5650_jack, mt8173_rt5650_hdmi_jack;
 
 static int mt8173_rt5650_init(struct snd_soc_pcm_runtime *runtime)
 {
 	struct snd_soc_card *card = runtime->card;
-	struct snd_soc_component *component = runtime->codec_dais[0]->component;
-	const char *codec_capture_dai = runtime->codec_dais[1]->name;
+	struct snd_soc_component *component = asoc_rtd_to_codec(runtime, 0)->component;
+	const char *codec_capture_dai = asoc_rtd_to_codec(runtime, 1)->name;
 	int ret;
 
 	rt5645_sel_asrc_clk_src(component,
···
 				      &mt8173_rt5650_jack,
 				      &mt8173_rt5650_jack,
 				      &mt8173_rt5650_jack);
+}
+
+static int mt8173_rt5650_hdmi_init(struct snd_soc_pcm_runtime *rtd)
+{
+	int ret;
+
+	ret = snd_soc_card_jack_new(rtd->card, "HDMI Jack", SND_JACK_LINEOUT,
+				    &mt8173_rt5650_hdmi_jack, NULL, 0);
+	if (ret)
+		return ret;
+
+	return hdmi_codec_set_jack_detect(asoc_rtd_to_codec(rtd, 0)->component,
+					  &mt8173_rt5650_hdmi_jack);
 }
 
 enum {
···
 		.name = "HDMI BE",
 		.no_pcm = 1,
 		.dpcm_playback = 1,
+		.init = mt8173_rt5650_hdmi_init,
 		SND_SOC_DAILINK_REG(hdmi_be),
 	},
 };

sound/soc/mediatek/mt8183/mt8183-afe-pcm.c

···
 	struct snd_soc_component *component =
 		snd_soc_rtdcom_lookup(rtd, AFE_PCM_NAME);
 	struct mtk_base_afe *afe = snd_soc_component_get_drvdata(component);
-	int id = rtd->cpu_dai->id;
+	int id = asoc_rtd_to_cpu(rtd, 0)->id;
 
 	return mt8183_rate_transform(afe->dev, rate, id);
 }

sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c

···
 #include "../../codecs/da7219-aad.h"
 #include "../../codecs/da7219.h"
 
-static struct snd_soc_jack headset_jack;
+struct mt8183_da7219_max98357_priv {
+	struct snd_soc_jack headset_jack;
+};
 
 static int mt8183_mt6358_i2s_hw_params(struct snd_pcm_substream *substream,
 				       struct snd_pcm_hw_params *params)
···
 	unsigned int mclk_fs_ratio = 128;
 	unsigned int mclk_fs = rate * mclk_fs_ratio;
 
-	return snd_soc_dai_set_sysclk(rtd->cpu_dai,
+	return snd_soc_dai_set_sysclk(asoc_rtd_to_cpu(rtd, 0),
 				      0, mclk_fs, SND_SOC_CLOCK_OUT);
 }
 
···
 				       struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai;
 	unsigned int rate = params_rate(params);
 	unsigned int mclk_fs_ratio = 256;
 	unsigned int mclk_fs = rate * mclk_fs_ratio;
 	unsigned int freq;
 	int ret = 0, j;
 
-	ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, 0,
+	ret = snd_soc_dai_set_sysclk(asoc_rtd_to_cpu(rtd, 0), 0,
 				     mclk_fs, SND_SOC_CLOCK_OUT);
 	if (ret < 0)
 		dev_err(rtd->dev, "failed to set cpu dai sysclk\n");
 
-	for (j = 0; j < rtd->num_codecs; j++) {
-		struct snd_soc_dai *codec_dai = rtd->codec_dais[j];
+	for_each_rtd_codec_dais(rtd, j, codec_dai) {
 
 		if (!strcmp(codec_dai->component->name, "da7219.5-001a")) {
 			ret = snd_soc_dai_set_sysclk(codec_dai,
···
 static int mt8183_da7219_hw_free(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai;
 	int ret = 0, j;
 
-	for (j = 0; j < rtd->num_codecs; j++) {
-		struct snd_soc_dai *codec_dai = rtd->codec_dais[j];
+	for_each_rtd_codec_dais(rtd, j, codec_dai) {
 
 		if (!strcmp(codec_dai->component->name, "da7219.5-001a")) {
 			ret = snd_soc_dai_set_pll(codec_dai,
···
 
 	return 0;
 }
+
+static int
+mt8183_da7219_max98357_bt_sco_startup(
+	struct snd_pcm_substream *substream)
+{
+	static const unsigned int rates[] = {
+		8000, 16000
+	};
+	static const struct snd_pcm_hw_constraint_list constraints_rates = {
+		.count = ARRAY_SIZE(rates),
+		.list  = rates,
+		.mask = 0,
+	};
+	static const unsigned int channels[] = {
+		1,
+	};
+	static const struct snd_pcm_hw_constraint_list constraints_channels = {
+		.count = ARRAY_SIZE(channels),
+		.list = channels,
+		.mask = 0,
+	};
+
+	struct snd_pcm_runtime *runtime = substream->runtime;
+
+	snd_pcm_hw_constraint_list(runtime, 0,
+			SNDRV_PCM_HW_PARAM_RATE, &constraints_rates);
+	runtime->hw.channels_max = 1;
+	snd_pcm_hw_constraint_list(runtime, 0,
+			SNDRV_PCM_HW_PARAM_CHANNELS,
+			&constraints_channels);
+
+	runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE;
+	snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16);
+
+	return 0;
+}
+
+static const struct snd_soc_ops mt8183_da7219_max98357_bt_sco_ops = {
+	.startup = mt8183_da7219_max98357_bt_sco_startup,
+};
 
 /* FE */
 SND_SOC_DAILINK_DEFS(playback1,
···
 			    SND_SOC_DPCM_TRIGGER_PRE},
 		.dynamic = 1,
 		.dpcm_playback = 1,
+		.ops = &mt8183_da7219_max98357_bt_sco_ops,
 		SND_SOC_DAILINK_REG(playback2),
 	},
 	{
···
 			    SND_SOC_DPCM_TRIGGER_PRE},
 		.dynamic = 1,
 		.dpcm_capture = 1,
+		.ops = &mt8183_da7219_max98357_bt_sco_ops,
 		SND_SOC_DAILINK_REG(capture1),
 	},
 	{
···
 	{
 		.name = "TDM",
 		.no_pcm = 1,
+		.dai_fmt = SND_SOC_DAIFMT_I2S |
+			   SND_SOC_DAIFMT_IB_IF |
+			   SND_SOC_DAIFMT_CBM_CFM,
 		.dpcm_playback = 1,
 		.ignore_suspend = 1,
+		.be_hw_params_fixup = mt8183_i2s_hw_params_fixup,
 		SND_SOC_DAILINK_REG(tdm),
 	},
 };
···
 	},
 };
 
+static const struct snd_kcontrol_new mt8183_da7219_max98357_snd_controls[] = {
+	SOC_DAPM_PIN_SWITCH("Speakers"),
+};
+
+static const
+struct snd_soc_dapm_widget mt8183_da7219_max98357_dapm_widgets[] = {
+	SND_SOC_DAPM_SPK("Speakers", NULL),
+	SND_SOC_DAPM_PINCTRL("TDM_OUT_PINCTRL",
+			     "aud_tdm_out_on", "aud_tdm_out_off"),
+};
+
+static const struct snd_soc_dapm_route mt8183_da7219_max98357_dapm_routes[] = {
+	{"Speakers", NULL, "Speaker"},
+	{"I2S Playback", NULL, "TDM_OUT_PINCTRL"},
+};
+
 static struct snd_soc_card mt8183_da7219_max98357_card = {
 	.name = "mt8183_da7219_max98357",
 	.owner = THIS_MODULE,
+	.controls = mt8183_da7219_max98357_snd_controls,
+	.num_controls = ARRAY_SIZE(mt8183_da7219_max98357_snd_controls),
+	.dapm_widgets = mt8183_da7219_max98357_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(mt8183_da7219_max98357_dapm_widgets),
+	.dapm_routes = mt8183_da7219_max98357_dapm_routes,
+	.num_dapm_routes = ARRAY_SIZE(mt8183_da7219_max98357_dapm_routes),
 	.dai_link = mt8183_da7219_max98357_dai_links,
 	.num_links = ARRAY_SIZE(mt8183_da7219_max98357_dai_links),
 	.aux_dev = &mt8183_da7219_max98357_headset_dev,
···
 mt8183_da7219_max98357_headset_init(struct snd_soc_component *component)
 {
 	int ret;
+	struct mt8183_da7219_max98357_priv *priv =
+			snd_soc_card_get_drvdata(component->card);
 
 	/* Enable Headset and 4 Buttons Jack detection */
 	ret = snd_soc_card_jack_new(&mt8183_da7219_max98357_card,
···
 				    SND_JACK_HEADSET |
 				    SND_JACK_BTN_0 | SND_JACK_BTN_1 |
 				    SND_JACK_BTN_2 | SND_JACK_BTN_3,
-				    &headset_jack,
+				    &priv->headset_jack,
 				    NULL, 0);
 	if (ret)
 		return ret;
 
-	da7219_aad_jack_det(component, &headset_jack);
+	da7219_aad_jack_det(component, &priv->headset_jack);
 
 	return ret;
 }
···
 	struct snd_soc_card *card = &mt8183_da7219_max98357_card;
 	struct device_node *platform_node;
 	struct snd_soc_dai_link *dai_link;
-	struct pinctrl *default_pins;
+	struct mt8183_da7219_max98357_priv *priv;
+	struct pinctrl *pinctrl;
 	int ret, i;
 
 	card->dev = &pdev->dev;
···
 		return -EINVAL;
 	}
 
-	ret = devm_snd_soc_register_card(&pdev->dev, card);
-	if (ret) {
-		dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n",
+	priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL);
+	if (!priv)
+		return -ENOMEM;
+
+	snd_soc_card_set_drvdata(card, priv);
+
+	pinctrl = devm_pinctrl_get_select(&pdev->dev, PINCTRL_STATE_DEFAULT);
+	if (IS_ERR(pinctrl)) {
+		ret = PTR_ERR(pinctrl);
+		dev_err(&pdev->dev, "%s failed to select default state %d\n",
 			__func__, ret);
 		return ret;
 	}
 
-	default_pins =
-		devm_pinctrl_get_select(&pdev->dev, PINCTRL_STATE_DEFAULT);
-	if (IS_ERR(default_pins)) {
-		dev_err(&pdev->dev, "%s set pins failed\n",
-			__func__);
-		return PTR_ERR(default_pins);
-	}
-
-	return ret;
+	return devm_snd_soc_register_card(&pdev->dev, card);
 }
 
 #ifdef CONFIG_OF
···
 MODULE_AUTHOR("Shunli Wang <[e-mail]>");
 MODULE_LICENSE("GPL v2");
 MODULE_ALIAS("mt8183_da7219_max98357 soc card");
-

sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c

···
 	unsigned int mclk_fs_ratio = 128;
 	unsigned int mclk_fs = rate * mclk_fs_ratio;
 
-	return snd_soc_dai_set_sysclk(rtd->cpu_dai,
+	return snd_soc_dai_set_sysclk(asoc_rtd_to_cpu(rtd, 0),
 				      0, mclk_fs, SND_SOC_CLOCK_OUT);
 }
 

sound/soc/meson/Kconfig

···
 menu "ASoC support for Amlogic platforms"
 	depends on ARCH_MESON || COMPILE_TEST
 
+config SND_MESON_AIU
+	tristate "Amlogic AIU"
+	select SND_MESON_CODEC_GLUE
+	select SND_PCM_IEC958
+	imply SND_SOC_MESON_T9015
+	imply SND_SOC_HDMI_CODEC if DRM_MESON_DW_HDMI
+	help
+	  Select Y or M to add support for the Audio output subsystem found
+	  in the Amlogic Meson8, Meson8b and GX SoC families
+
 config SND_MESON_AXG_FIFO
 	tristate
 	select REGMAP_MMIO
···
 config SND_MESON_AXG_SOUND_CARD
 	tristate "Amlogic AXG Sound Card Support"
 	select SND_MESON_AXG_TDM_INTERFACE
+	select SND_MESON_CARD_UTILS
 	imply SND_MESON_AXG_FRDDR
 	imply SND_MESON_AXG_TODDR
 	imply SND_MESON_AXG_TDMIN
···
 	  Select Y or M to add support for PDM input embedded
 	  in the Amlogic AXG SoC family
 
+config SND_MESON_CARD_UTILS
+       tristate
+
+config SND_MESON_CODEC_GLUE
+	tristate
+
+config SND_MESON_GX_SOUND_CARD
+	tristate "Amlogic GX Sound Card Support"
+	select SND_MESON_CARD_UTILS
+	imply SND_MESON_AIU
+	help
+	  Select Y or M to add support for the GXBB/GXL SoC sound card
+
+config SND_MESON_G12A_TOACODEC
+	tristate "Amlogic G12A To Internal DAC Control Support"
+	select SND_MESON_CODEC_GLUE
+	select REGMAP_MMIO
+	imply SND_SOC_MESON_T9015
+	help
+	  Select Y or M to add support for the internal audio DAC on the
+	  g12a SoC family
+
 config SND_MESON_G12A_TOHDMITX
 	tristate "Amlogic G12A To HDMI TX Control Support"
 	select REGMAP_MMIO
+	select SND_MESON_CODEC_GLUE
 	imply SND_SOC_HDMI_CODEC
 	help
 	  Select Y or M to add support for HDMI audio on the g12a SoC
 	  family
+
+config SND_SOC_MESON_T9015
+	tristate "Amlogic T9015 DAC"
+	select REGMAP_MMIO
+	help
+	  Say Y or M if you want to add support for the internal DAC found
+	  on GXL, G12 and SM1 SoC family.
 endmenu

sound/soc/meson/Makefile

···
 # SPDX-License-Identifier: (GPL-2.0 OR MIT)
 
+snd-soc-meson-aiu-objs := aiu.o
+snd-soc-meson-aiu-objs += aiu-acodec-ctrl.o
+snd-soc-meson-aiu-objs += aiu-codec-ctrl.o
+snd-soc-meson-aiu-objs += aiu-encoder-i2s.o
+snd-soc-meson-aiu-objs += aiu-encoder-spdif.o
+snd-soc-meson-aiu-objs += aiu-fifo.o
+snd-soc-meson-aiu-objs += aiu-fifo-i2s.o
+snd-soc-meson-aiu-objs += aiu-fifo-spdif.o
 snd-soc-meson-axg-fifo-objs := axg-fifo.o
 snd-soc-meson-axg-frddr-objs := axg-frddr.o
 snd-soc-meson-axg-toddr-objs := axg-toddr.o
···
 snd-soc-meson-axg-spdifin-objs := axg-spdifin.o
 snd-soc-meson-axg-spdifout-objs := axg-spdifout.o
 snd-soc-meson-axg-pdm-objs := axg-pdm.o
+snd-soc-meson-card-utils-objs := meson-card-utils.o
+snd-soc-meson-codec-glue-objs := meson-codec-glue.o
+snd-soc-meson-gx-sound-card-objs := gx-card.o
+snd-soc-meson-g12a-toacodec-objs := g12a-toacodec.o
 snd-soc-meson-g12a-tohdmitx-objs := g12a-tohdmitx.o
+snd-soc-meson-t9015-objs := t9015.o
 
+obj-$(CONFIG_SND_MESON_AIU) += snd-soc-meson-aiu.o
 obj-$(CONFIG_SND_MESON_AXG_FIFO) += snd-soc-meson-axg-fifo.o
 obj-$(CONFIG_SND_MESON_AXG_FRDDR) += snd-soc-meson-axg-frddr.o
 obj-$(CONFIG_SND_MESON_AXG_TODDR) += snd-soc-meson-axg-toddr.o
···
 obj-$(CONFIG_SND_MESON_AXG_SPDIFIN) += snd-soc-meson-axg-spdifin.o
 obj-$(CONFIG_SND_MESON_AXG_SPDIFOUT) += snd-soc-meson-axg-spdifout.o
 obj-$(CONFIG_SND_MESON_AXG_PDM) += snd-soc-meson-axg-pdm.o
+obj-$(CONFIG_SND_MESON_CARD_UTILS) += snd-soc-meson-card-utils.o
+obj-$(CONFIG_SND_MESON_CODEC_GLUE) += snd-soc-meson-codec-glue.o
+obj-$(CONFIG_SND_MESON_GX_SOUND_CARD) += snd-soc-meson-gx-sound-card.o
+obj-$(CONFIG_SND_MESON_G12A_TOACODEC) += snd-soc-meson-g12a-toacodec.o
 obj-$(CONFIG_SND_MESON_G12A_TOHDMITX) += snd-soc-meson-g12a-tohdmitx.o
+obj-$(CONFIG_SND_SOC_MESON_T9015) += snd-soc-meson-t9015.o

sound/soc/meson/aiu-acodec-ctrl.c

···
+// SPDX-License-Identifier: GPL-2.0
+//
+// Copyright (c) 2020 BayLibre, SAS.
+// Author: Jerome Brunet <[e-mail]>
+
+#include <linux/bitfield.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+
+#include <dt-bindings/sound/meson-aiu.h>
+#include "aiu.h"
+#include "meson-codec-glue.h"
+
+#define CTRL_DIN_EN			15
+#define CTRL_CLK_INV			BIT(14)
+#define CTRL_LRCLK_INV			BIT(13)
+#define CTRL_I2S_IN_BCLK_SRC		BIT(11)
+#define CTRL_DIN_LRCLK_SRC_SHIFT	6
+#define CTRL_DIN_LRCLK_SRC		(0x3 << CTRL_DIN_LRCLK_SRC_SHIFT)
+#define CTRL_BCLK_MCLK_SRC		GENMASK(5, 4)
+#define CTRL_DIN_SKEW			GENMASK(3, 2)
+#define CTRL_I2S_OUT_LANE_SRC		0
+
+#define AIU_ACODEC_OUT_CHMAX		2
+
+static const char * const aiu_acodec_ctrl_mux_texts[] = {
+	"DISABLED", "I2S", "PCM",
+};
+
+static int aiu_acodec_ctrl_mux_put_enum(struct snd_kcontrol *kcontrol,
+					struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_component *component =
+		snd_soc_dapm_kcontrol_component(kcontrol);
+	struct snd_soc_dapm_context *dapm =
+		snd_soc_dapm_kcontrol_dapm(kcontrol);
+	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+	unsigned int mux, changed;
+
+	mux = snd_soc_enum_item_to_val(e, ucontrol->value.enumerated.item[0]);
+	changed = snd_soc_component_test_bits(component, e->reg,
+					      CTRL_DIN_LRCLK_SRC,
+					      FIELD_PREP(CTRL_DIN_LRCLK_SRC,
+							 mux));
+
+	if (!changed)
+		return 0;
+
+	/* Force disconnect of the mux while updating */
+	snd_soc_dapm_mux_update_power(dapm, kcontrol, 0, NULL, NULL);
+
+	snd_soc_component_update_bits(component, e->reg,
+				      CTRL_DIN_LRCLK_SRC |
+				      CTRL_BCLK_MCLK_SRC,
+				      FIELD_PREP(CTRL_DIN_LRCLK_SRC, mux) |
+				      FIELD_PREP(CTRL_BCLK_MCLK_SRC, mux));
+
+	snd_soc_dapm_mux_update_power(dapm, kcontrol, mux, e, NULL);
+
+	return 0;
+}
+
+static SOC_ENUM_SINGLE_DECL(aiu_acodec_ctrl_mux_enum, AIU_ACODEC_CTRL,
+			    CTRL_DIN_LRCLK_SRC_SHIFT,
+			    aiu_acodec_ctrl_mux_texts);
+
+static const struct snd_kcontrol_new aiu_acodec_ctrl_mux =
+	SOC_DAPM_ENUM_EXT("ACodec Source", aiu_acodec_ctrl_mux_enum,
+			  snd_soc_dapm_get_enum_double,
+			  aiu_acodec_ctrl_mux_put_enum);
+
+static const struct snd_kcontrol_new aiu_acodec_ctrl_out_enable =
+	SOC_DAPM_SINGLE_AUTODISABLE("Switch", AIU_ACODEC_CTRL,
+				    CTRL_DIN_EN, 1, 0);
+
+static const struct snd_soc_dapm_widget aiu_acodec_ctrl_widgets[] = {
+	SND_SOC_DAPM_MUX("ACODEC SRC", SND_SOC_NOPM, 0, 0,
+			 &aiu_acodec_ctrl_mux),
+	SND_SOC_DAPM_SWITCH("ACODEC OUT EN", SND_SOC_NOPM, 0, 0,
+			    &aiu_acodec_ctrl_out_enable),
+};
+
+static int aiu_acodec_ctrl_input_hw_params(struct snd_pcm_substream *substream,
+					   struct snd_pcm_hw_params *params,
+					   struct snd_soc_dai *dai)
+{
+	struct meson_codec_glue_input *data;
+	int ret;
+
+	ret = meson_codec_glue_input_hw_params(substream, params, dai);
+	if (ret)
+		return ret;
+
+	/* The glue will provide 1 lane out of the 4 to the output */
+	data = meson_codec_glue_input_get_data(dai);
+	data->params.channels_min = min_t(unsigned int, AIU_ACODEC_OUT_CHMAX,
+					  data->params.channels_min);
+	data->params.channels_max = min_t(unsigned int, AIU_ACODEC_OUT_CHMAX,
+					  data->params.channels_max);
+
+	return 0;
+}
+
+static const struct snd_soc_dai_ops aiu_acodec_ctrl_input_ops = {
+	.hw_params	= aiu_acodec_ctrl_input_hw_params,
+	.set_fmt	= meson_codec_glue_input_set_fmt,
+};
+
+static const struct snd_soc_dai_ops aiu_acodec_ctrl_output_ops = {
+	.startup	= meson_codec_glue_output_startup,
+};
+
+#define AIU_ACODEC_CTRL_FORMATS					\
+	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |	\
+	 SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE |	\
+	 SNDRV_PCM_FMTBIT_S32_LE)
+
+#define AIU_ACODEC_STREAM(xname, xsuffix, xchmax)		\
+{								\
+	.stream_name	= xname " " xsuffix,			\
+	.channels_min	= 1,					\
+	.channels_max	= (xchmax),				\
+	.rate_min       = 5512,					\
+	.rate_max	= 192000,				\
+	.formats	= AIU_ACODEC_CTRL_FORMATS,		\
+}
+
+#define AIU_ACODEC_INPUT(xname) {				\
+	.name = "ACODEC CTRL " xname,				\
+	.playback = AIU_ACODEC_STREAM(xname, "Playback", 8),	\
+	.ops = &aiu_acodec_ctrl_input_ops,			\
+	.probe = meson_codec_glue_input_dai_probe,		\
+	.remove = meson_codec_glue_input_dai_remove,		\
+}
+
+#define AIU_ACODEC_OUTPUT(xname) {				\
+	.name = "ACODEC CTRL " xname,				\
+	.capture = AIU_ACODEC_STREAM(xname, "Capture", AIU_ACODEC_OUT_CHMAX), \
+	.ops = &aiu_acodec_ctrl_output_ops,			\
+}
+
+static struct snd_soc_dai_driver aiu_acodec_ctrl_dai_drv[] = {
+	[CTRL_I2S] = AIU_ACODEC_INPUT("ACODEC I2S IN"),
+	[CTRL_PCM] = AIU_ACODEC_INPUT("ACODEC PCM IN"),
+	[CTRL_OUT] = AIU_ACODEC_OUTPUT("ACODEC OUT"),
+};
+
+static const struct snd_soc_dapm_route aiu_acodec_ctrl_routes[] = {
+	{ "ACODEC SRC", "I2S", "ACODEC I2S IN Playback" },
+	{ "ACODEC SRC", "PCM", "ACODEC PCM IN Playback" },
+	{ "ACODEC OUT EN", "Switch", "ACODEC SRC" },
+	{ "ACODEC OUT Capture", NULL, "ACODEC OUT EN" },
+};
+
+static const struct snd_kcontrol_new aiu_acodec_ctrl_controls[] = {
+	SOC_SINGLE("ACODEC I2S Lane Select", AIU_ACODEC_CTRL,
+		   CTRL_I2S_OUT_LANE_SRC, 3, 0),
+};
+
+static int aiu_acodec_of_xlate_dai_name(struct snd_soc_component *component,
+					struct of_phandle_args *args,
+					const char **dai_name)
+{
+	return aiu_of_xlate_dai_name(component, args, dai_name, AIU_ACODEC);
+}
+
+static int aiu_acodec_ctrl_component_probe(struct snd_soc_component *component)
+{
+	/*
+	 * NOTE: Din Skew setting
+	 * According to the documentation, the following update adds one delay
+	 * to the din line. Without this, the output saturates. This happens
+	 * regardless of the link format (i2s or left_j) so it is not clear what
+	 * it actually does but it seems to be required
+	 */
+	snd_soc_component_update_bits(component, AIU_ACODEC_CTRL,
+				      CTRL_DIN_SKEW,
+				      FIELD_PREP(CTRL_DIN_SKEW, 2));
+
+	return 0;
+}
+
+static const struct snd_soc_component_driver aiu_acodec_ctrl_component = {
+	.name			= "AIU Internal DAC Codec Control",
+	.probe			= aiu_acodec_ctrl_component_probe,
+	.controls		= aiu_acodec_ctrl_controls,
+	.num_controls		= ARRAY_SIZE(aiu_acodec_ctrl_controls),
+	.dapm_widgets		= aiu_acodec_ctrl_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(aiu_acodec_ctrl_widgets),
+	.dapm_routes		= aiu_acodec_ctrl_routes,
+	.num_dapm_routes	= ARRAY_SIZE(aiu_acodec_ctrl_routes),
+	.of_xlate_dai_name	= aiu_acodec_of_xlate_dai_name,
+	.endianness		= 1,
+	.non_legacy_dai_naming	= 1,
+};
+
+int aiu_acodec_ctrl_register_component(struct device *dev)
+{
+	return snd_soc_register_component(dev, &aiu_acodec_ctrl_component,
+					  aiu_acodec_ctrl_dai_drv,
+					  ARRAY_SIZE(aiu_acodec_ctrl_dai_drv));
+}

sound/soc/meson/aiu-codec-ctrl.c

···
+// SPDX-License-Identifier: GPL-2.0
+//
+// Copyright (c) 2020 BayLibre, SAS.
+// Author: Jerome Brunet <[e-mail]>
+
+#include <linux/bitfield.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+
+#include <dt-bindings/sound/meson-aiu.h>
+#include "aiu.h"
+#include "meson-codec-glue.h"
+
+#define CTRL_CLK_SEL		GENMASK(1, 0)
+#define CTRL_DATA_SEL_SHIFT	4
+#define CTRL_DATA_SEL		(0x3 << CTRL_DATA_SEL_SHIFT)
+
+static const char * const aiu_codec_ctrl_mux_texts[] = {
+	"DISABLED", "PCM", "I2S",
+};
+
+static int aiu_codec_ctrl_mux_put_enum(struct snd_kcontrol *kcontrol,
+				       struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_component *component =
+		snd_soc_dapm_kcontrol_component(kcontrol);
+	struct snd_soc_dapm_context *dapm =
+		snd_soc_dapm_kcontrol_dapm(kcontrol);
+	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+	unsigned int mux, changed;
+
+	mux = snd_soc_enum_item_to_val(e, ucontrol->value.enumerated.item[0]);
+	changed = snd_soc_component_test_bits(component, e->reg,
+					      CTRL_DATA_SEL,
+					      FIELD_PREP(CTRL_DATA_SEL, mux));
+
+	if (!changed)
+		return 0;
+
+	/* Force disconnect of the mux while updating */
+	snd_soc_dapm_mux_update_power(dapm, kcontrol, 0, NULL, NULL);
+
+	/* Reset the source first */
+	snd_soc_component_update_bits(component, e->reg,
+				      CTRL_CLK_SEL |
+				      CTRL_DATA_SEL,
+				      FIELD_PREP(CTRL_CLK_SEL, 0) |
+				      FIELD_PREP(CTRL_DATA_SEL, 0));
+
+	/* Set the appropriate source */
+	snd_soc_component_update_bits(component, e->reg,
+				      CTRL_CLK_SEL |
+				      CTRL_DATA_SEL,
+				      FIELD_PREP(CTRL_CLK_SEL, mux) |
+				      FIELD_PREP(CTRL_DATA_SEL, mux));
+
+	snd_soc_dapm_mux_update_power(dapm, kcontrol, mux, e, NULL);
+
+	return 0;
+}
+
+static SOC_ENUM_SINGLE_DECL(aiu_hdmi_ctrl_mux_enum, AIU_HDMI_CLK_DATA_CTRL,
+			    CTRL_DATA_SEL_SHIFT,
+			    aiu_codec_ctrl_mux_texts);
+
+static const struct snd_kcontrol_new aiu_hdmi_ctrl_mux =
+	SOC_DAPM_ENUM_EXT("HDMI Source", aiu_hdmi_ctrl_mux_enum,
+			  snd_soc_dapm_get_enum_double,
+			  aiu_codec_ctrl_mux_put_enum);
+
+static const struct snd_soc_dapm_widget aiu_hdmi_ctrl_widgets[] = {
+	SND_SOC_DAPM_MUX("HDMI CTRL SRC", SND_SOC_NOPM, 0, 0,
+			 &aiu_hdmi_ctrl_mux),
+};
+
+static const struct snd_soc_dai_ops aiu_codec_ctrl_input_ops = {
+	.hw_params	= meson_codec_glue_input_hw_params,
+	.set_fmt	= meson_codec_glue_input_set_fmt,
+};
+
+static const struct snd_soc_dai_ops aiu_codec_ctrl_output_ops = {
+	.startup	= meson_codec_glue_output_startup,
+};
+
+#define AIU_CODEC_CTRL_FORMATS					\
+	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |	\
+	 SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE |	\
+	 SNDRV_PCM_FMTBIT_S32_LE)
+
+#define AIU_CODEC_CTRL_STREAM(xname, xsuffix)			\
+{								\
+	.stream_name	= xname " " xsuffix,			\
+	.channels_min	= 1,					\
+	.channels_max	= 8,					\
+	.rate_min       = 5512,					\
+	.rate_max	= 192000,				\
+	.formats	= AIU_CODEC_CTRL_FORMATS,		\
+}
+
+#define AIU_CODEC_CTRL_INPUT(xname) {				\
+	.name = "CODEC CTRL " xname,				\
+	.playback = AIU_CODEC_CTRL_STREAM(xname, "Playback"),	\
+	.ops = &aiu_codec_ctrl_input_ops,			\
+	.probe = meson_codec_glue_input_dai_probe,		\
+	.remove = meson_codec_glue_input_dai_remove,		\
+}
+
+#define AIU_CODEC_CTRL_OUTPUT(xname) {				\
+	.name = "CODEC CTRL " xname,				\
+	.capture = AIU_CODEC_CTRL_STREAM(xname, "Capture"),	\
+	.ops = &aiu_codec_ctrl_output_ops,			\
+}
+
+static struct snd_soc_dai_driver aiu_hdmi_ctrl_dai_drv[] = {
+	[CTRL_I2S] = AIU_CODEC_CTRL_INPUT("HDMI I2S IN"),
+	[CTRL_PCM] = AIU_CODEC_CTRL_INPUT("HDMI PCM IN"),
+	[CTRL_OUT] = AIU_CODEC_CTRL_OUTPUT("HDMI OUT"),
+};
+
+static const struct snd_soc_dapm_route aiu_hdmi_ctrl_routes[] = {
+	{ "HDMI CTRL SRC", "I2S", "HDMI I2S IN Playback" },
+	{ "HDMI CTRL SRC", "PCM", "HDMI PCM IN Playback" },
+	{ "HDMI OUT Capture", NULL, "HDMI CTRL SRC" },
+};
+
+static int aiu_hdmi_of_xlate_dai_name(struct snd_soc_component *component,
+				      struct of_phandle_args *args,
+				      const char **dai_name)
+{
+	return aiu_of_xlate_dai_name(component, args, dai_name, AIU_HDMI);
+}
+
+static const struct snd_soc_component_driver aiu_hdmi_ctrl_component = {
+	.name			= "AIU HDMI Codec Control",
+	.dapm_widgets		= aiu_hdmi_ctrl_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(aiu_hdmi_ctrl_widgets),
+	.dapm_routes		= aiu_hdmi_ctrl_routes,
+	.num_dapm_routes	= ARRAY_SIZE(aiu_hdmi_ctrl_routes),
+	.of_xlate_dai_name	= aiu_hdmi_of_xlate_dai_name,
+	.endianness		= 1,
+	.non_legacy_dai_naming	= 1,
+};
+
+int aiu_hdmi_ctrl_register_component(struct device *dev)
+{
+	return snd_soc_register_component(dev, &aiu_hdmi_ctrl_component,
+					  aiu_hdmi_ctrl_dai_drv,
+					  ARRAY_SIZE(aiu_hdmi_ctrl_dai_drv));
+}
+

sound/soc/meson/aiu-encoder-i2s.c

···
+// SPDX-License-Identifier: GPL-2.0
+//
+// Copyright (c) 2020 BayLibre, SAS.
+// Author: Jerome Brunet <[e-mail]>
+
+#include <linux/bitfield.h>
+#include <linux/clk.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+
+#include "aiu.h"
+
+#define AIU_I2S_SOURCE_DESC_MODE_8CH	BIT(0)
+#define AIU_I2S_SOURCE_DESC_MODE_24BIT	BIT(5)
+#define AIU_I2S_SOURCE_DESC_MODE_32BIT	BIT(9)
+#define AIU_I2S_SOURCE_DESC_MODE_SPLIT	BIT(11)
+#define AIU_RST_SOFT_I2S_FAST		BIT(0)
+
+#define AIU_I2S_DAC_CFG_MSB_FIRST	BIT(2)
+#define AIU_I2S_MISC_HOLD_EN		BIT(2)
+#define AIU_CLK_CTRL_I2S_DIV_EN		BIT(0)
+#define AIU_CLK_CTRL_I2S_DIV		GENMASK(3, 2)
+#define AIU_CLK_CTRL_AOCLK_INVERT	BIT(6)
+#define AIU_CLK_CTRL_LRCLK_INVERT	BIT(7)
+#define AIU_CLK_CTRL_LRCLK_SKEW		GENMASK(9, 8)
+#define AIU_CLK_CTRL_MORE_HDMI_AMCLK	BIT(6)
+#define AIU_CLK_CTRL_MORE_I2S_DIV	GENMASK(5, 0)
+#define AIU_CODEC_DAC_LRCLK_CTRL_DIV	GENMASK(11, 0)
+
+static void aiu_encoder_i2s_divider_enable(struct snd_soc_component *component,
+					   bool enable)
+{
+	snd_soc_component_update_bits(component, AIU_CLK_CTRL,
+				      AIU_CLK_CTRL_I2S_DIV_EN,
+				      enable ? AIU_CLK_CTRL_I2S_DIV_EN : 0);
+}
+
+static void aiu_encoder_i2s_hold(struct snd_soc_component *component,
+				 bool enable)
+{
+	snd_soc_component_update_bits(component, AIU_I2S_MISC,
+				      AIU_I2S_MISC_HOLD_EN,
+				      enable ? AIU_I2S_MISC_HOLD_EN : 0);
+}
+
+static int aiu_encoder_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
+				   struct snd_soc_dai *dai)
+{
+	struct snd_soc_component *component = dai->component;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		aiu_encoder_i2s_hold(component, false);
+		return 0;
+
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		aiu_encoder_i2s_hold(component, true);
+		return 0;
+
+	default:
+		return -EINVAL;
+	}
+}
+
+static int aiu_encoder_i2s_setup_desc(struct snd_soc_component *component,
+				      struct snd_pcm_hw_params *params)
+{
+	/* Always operate in split (classic interleaved) mode */
+	unsigned int desc = AIU_I2S_SOURCE_DESC_MODE_SPLIT;
+	unsigned int val;
+
+	/* Reset required to update the pipeline */
+	snd_soc_component_write(component, AIU_RST_SOFT, AIU_RST_SOFT_I2S_FAST);
+	snd_soc_component_read(component, AIU_I2S_SYNC, &val);
+
+	switch (params_physical_width(params)) {
+	case 16: /* Nothing to do */
+		break;
+
+	case 32:
+		desc |= (AIU_I2S_SOURCE_DESC_MODE_24BIT |
+			 AIU_I2S_SOURCE_DESC_MODE_32BIT);
+		break;
+
+	default:
+		return -EINVAL;
+	}
+
+	switch (params_channels(params)) {
+	case 2: /* Nothing to do */
+		break;
+	case 8:
+		desc |= AIU_I2S_SOURCE_DESC_MODE_8CH;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	snd_soc_component_update_bits(component, AIU_I2S_SOURCE_DESC,
+				      AIU_I2S_SOURCE_DESC_MODE_8CH |
+				      AIU_I2S_SOURCE_DESC_MODE_24BIT |
+				      AIU_I2S_SOURCE_DESC_MODE_32BIT |
+				      AIU_I2S_SOURCE_DESC_MODE_SPLIT,
+				      desc);
+
+	return 0;
+}
+
+static int aiu_encoder_i2s_set_legacy_div(struct snd_soc_component *component,
+					  struct snd_pcm_hw_params *params,
+					  unsigned int bs)
+{
+	switch (bs) {
+	case 1:
+	case 2:
+	case 4:
+	case 8:
+		/* These are the only valid legacy dividers */
+		break;
+
+	default:
+		dev_err(component->dev, "Unsupported i2s divider: %u\n", bs);
+		return -EINVAL;
+	}
+
+	snd_soc_component_update_bits(component, AIU_CLK_CTRL,
+				      AIU_CLK_CTRL_I2S_DIV,
+				      FIELD_PREP(AIU_CLK_CTRL_I2S_DIV,
+						 __ffs(bs)));
+
+	snd_soc_component_update_bits(component, AIU_CLK_CTRL_MORE,
+				      AIU_CLK_CTRL_MORE_I2S_DIV,
+				      FIELD_PREP(AIU_CLK_CTRL_MORE_I2S_DIV,
+						 0));
+
+	return 0;
+}
+
+static int aiu_encoder_i2s_set_more_div(struct snd_soc_component *component,
+					struct snd_pcm_hw_params *params,
+					unsigned int bs)
+{
+	/*
+	 * NOTE: this HW is odd.
+	 * In most configuration, the i2s divider is 'mclk / blck'.
+	 * However, in 16 bits - 8ch mode, this factor needs to be
+	 * increased by 50% to get the correct output rate.
+	 * No idea why !
+	 */
+	if (params_width(params) == 16 && params_channels(params) == 8) {
+		if (bs % 2) {
+			dev_err(component->dev,
+				"Cannot increase i2s divider by 50%%\n");
+			return -EINVAL;
+		}
+		bs += bs / 2;
+	}
+
+	/* Use CLK_MORE for mclk to bclk divider */
+	snd_soc_component_update_bits(component, AIU_CLK_CTRL,
+				      AIU_CLK_CTRL_I2S_DIV,
+				      FIELD_PREP(AIU_CLK_CTRL_I2S_DIV, 0));
+
+	snd_soc_component_update_bits(component, AIU_CLK_CTRL_MORE,
+				      AIU_CLK_CTRL_MORE_I2S_DIV,
+				      FIELD_PREP(AIU_CLK_CTRL_MORE_I2S_DIV,
+						 bs - 1));
+
+	return 0;
+}
+
+static int aiu_encoder_i2s_set_clocks(struct snd_soc_component *component,
+				      struct snd_pcm_hw_params *params)
+{
+	struct aiu *aiu = snd_soc_component_get_drvdata(component);
+	unsigned int srate = params_rate(params);
+	unsigned int fs, bs;
+	int ret;
+
+	/* Get the oversampling factor */
+	fs = DIV_ROUND_CLOSEST(clk_get_rate(aiu->i2s.clks[MCLK].clk), srate);
+
+	if (fs % 64)
+		return -EINVAL;
+
+	/* Send data MSB first */
+	snd_soc_component_update_bits(component, AIU_I2S_DAC_CFG,
+				      AIU_I2S_DAC_CFG_MSB_FIRST,
+				      AIU_I2S_DAC_CFG_MSB_FIRST);
+
+	/* Set bclk to lrlck ratio */
+	snd_soc_component_update_bits(component, AIU_CODEC_DAC_LRCLK_CTRL,
+				      AIU_CODEC_DAC_LRCLK_CTRL_DIV,
+				      FIELD_PREP(AIU_CODEC_DAC_LRCLK_CTRL_DIV,
+						 64 - 1));
+
+	bs = fs / 64;
+
+	if (aiu->platform->has_clk_ctrl_more_i2s_div)
+		ret = aiu_encoder_i2s_set_more_div(component, params, bs);
+	else
+		ret = aiu_encoder_i2s_set_legacy_div(component, params, bs);
+
+	if (ret)
+		return ret;
+
+	/* Make sure amclk is used for HDMI i2s as well */
+	snd_soc_component_update_bits(component, AIU_CLK_CTRL_MORE,
+				      AIU_CLK_CTRL_MORE_HDMI_AMCLK,
+				      AIU_CLK_CTRL_MORE_HDMI_AMCLK);
+
+	return 0;
+}
+
+static int aiu_encoder_i2s_hw_params(struct snd_pcm_substream *substream,
+				     struct snd_pcm_hw_params *params,
+				     struct snd_soc_dai *dai)
+{
+	struct snd_soc_component *component = dai->component;
+	int ret;
+
+	/* Disable the clock while changing the settings */
+	aiu_encoder_i2s_divider_enable(component, false);
+
+	ret = aiu_encoder_i2s_setup_desc(component, params);
+	if (ret) {
+		dev_err(dai->dev, "setting i2s desc failed\n");
+		return ret;
+	}
+
+	ret = aiu_encoder_i2s_set_clocks(component, params);
+	if (ret) {
+		dev_err(dai->dev, "setting i2s clocks failed\n");
+		return ret;
+	}
+
+	aiu_encoder_i2s_divider_enable(component, true);
+
+	return 0;
+}
+
+static int aiu_encoder_i2s_hw_free(struct snd_pcm_substream *substream,
+				   struct snd_soc_dai *dai)
+{
+	struct snd_soc_component *component = dai->component;
+
+	aiu_encoder_i2s_divider_enable(component, false);
+
+	return 0;
+}
+
+static int aiu_encoder_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+	struct snd_soc_component *component = dai->component;
+	unsigned int inv = fmt & SND_SOC_DAIFMT_INV_MASK;
+	unsigned int val = 0;
+	unsigned int skew;
+
+	/* Only CPU Master / Codec Slave supported ATM */
+	if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS)
+		return -EINVAL;
+
+	if (inv == SND_SOC_DAIFMT_NB_IF ||
+	    inv == SND_SOC_DAIFMT_IB_IF)
+		val |= AIU_CLK_CTRL_LRCLK_INVERT;
+
+	if (inv == SND_SOC_DAIFMT_IB_NF ||
+	    inv == SND_SOC_DAIFMT_IB_IF)
+		val |= AIU_CLK_CTRL_AOCLK_INVERT;
+
+	/* Signal skew */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		/* Invert sample clock for i2s */
+		val ^= AIU_CLK_CTRL_LRCLK_INVERT;
+		skew = 1;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		skew = 0;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	val |= FIELD_PREP(AIU_CLK_CTRL_LRCLK_SKEW, skew);
+	snd_soc_component_update_bits(component, AIU_CLK_CTRL,
+				      AIU_CLK_CTRL_LRCLK_INVERT |
+				      AIU_CLK_CTRL_AOCLK_INVERT |
+				      AIU_CLK_CTRL_LRCLK_SKEW,
+				      val);
+
+	return 0;
+}
+
+static int aiu_encoder_i2s_set_sysclk(struct snd_soc_dai *dai, int clk_id,
+				      unsigned int freq, int dir)
+{
+	struct aiu *aiu = snd_soc_component_get_drvdata(dai->component);
+	int ret;
+
+	if (WARN_ON(clk_id != 0))
+		return -EINVAL;
+
+	if (dir == SND_SOC_CLOCK_IN)
+		return 0;
+
+	ret = clk_set_rate(aiu->i2s.clks[MCLK].clk, freq);
+	if (ret)
+		dev_err(dai->dev, "Failed to set sysclk to %uHz", freq);
+
+	return ret;
+}
+
+static const unsigned int hw_channels[] = {2, 8};
+static const struct snd_pcm_hw_constraint_list hw_channel_constraints = {
+	.list = hw_channels,
+	.count = ARRAY_SIZE(hw_channels),
+	.mask = 0,
+};
+
+static int aiu_encoder_i2s_startup(struct snd_pcm_substream *substream,
+				   struct snd_soc_dai *dai)
+{
+	struct aiu *aiu = snd_soc_component_get_drvdata(dai->component);
+	int ret;
+
+	/* Make sure the encoder gets either 2 or 8 channels */
+	ret = snd_pcm_hw_constraint_list(substream->runtime, 0,
+					 SNDRV_PCM_HW_PARAM_CHANNELS,
+					 &hw_channel_constraints);
+	if (ret) {
+		dev_err(dai->dev, "adding channels constraints failed\n");
+		return ret;
+	}
+
+	ret = clk_bulk_prepare_enable(aiu->i2s.clk_num, aiu->i2s.clks);
+	if (ret)
+		dev_err(dai->dev, "failed to enable i2s clocks\n");
+
+	return ret;
+}
+
+static void aiu_encoder_i2s_shutdown(struct snd_pcm_substream *substream,
+				     struct snd_soc_dai *dai)
+{
+	struct aiu *aiu = snd_soc_component_get_drvdata(dai->component);
+
+	clk_bulk_disable_unprepare(aiu->i2s.clk_num, aiu->i2s.clks);
+}
+
+const struct snd_soc_dai_ops aiu_encoder_i2s_dai_ops = {
+	.trigger	= aiu_encoder_i2s_trigger,
+	.hw_params	= aiu_encoder_i2s_hw_params,
+	.hw_free	= aiu_encoder_i2s_hw_free,
+	.set_fmt	= aiu_encoder_i2s_set_fmt,
+	.set_sysclk	= aiu_encoder_i2s_set_sysclk,
+	.startup	= aiu_encoder_i2s_startup,
+	.shutdown	= aiu_encoder_i2s_shutdown,
+};
+

sound/soc/meson/aiu-encoder-spdif.c

···
+// SPDX-License-Identifier: GPL-2.0
+//
+// Copyright (c) 2020 BayLibre, SAS.
+// Author: Jerome Brunet <[e-mail]>
+
+#include <linux/bitfield.h>
+#include <linux/clk.h>
+#include <sound/pcm_params.h>
+#include <sound/pcm_iec958.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+
+#include "aiu.h"
+
+#define AIU_958_MISC_NON_PCM		BIT(0)
+#define AIU_958_MISC_MODE_16BITS	BIT(1)
+#define AIU_958_MISC_16BITS_ALIGN	GENMASK(6, 5)
+#define AIU_958_MISC_MODE_32BITS	BIT(7)
+#define AIU_958_MISC_U_FROM_STREAM	BIT(12)
+#define AIU_958_MISC_FORCE_LR		BIT(13)
+#define AIU_958_CTRL_HOLD_EN		BIT(0)
+#define AIU_CLK_CTRL_958_DIV_EN		BIT(1)
+#define AIU_CLK_CTRL_958_DIV		GENMASK(5, 4)
+#define AIU_CLK_CTRL_958_DIV_MORE	BIT(12)
+
+#define AIU_CS_WORD_LEN			4
+#define AIU_958_INTERNAL_DIV		2
+
+static void
+aiu_encoder_spdif_divider_enable(struct snd_soc_component *component,
+				 bool enable)
+{
+	snd_soc_component_update_bits(component, AIU_CLK_CTRL,
+				      AIU_CLK_CTRL_958_DIV_EN,
+				      enable ? AIU_CLK_CTRL_958_DIV_EN : 0);
+}
+
+static void aiu_encoder_spdif_hold(struct snd_soc_component *component,
+				   bool enable)
+{
+	snd_soc_component_update_bits(component, AIU_958_CTRL,
+				      AIU_958_CTRL_HOLD_EN,
+				      enable ? AIU_958_CTRL_HOLD_EN : 0);
+}
+
+static int
+aiu_encoder_spdif_trigger(struct snd_pcm_substream *substream, int cmd,
+			  struct snd_soc_dai *dai)
+{
+	struct snd_soc_component *component = dai->component;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		aiu_encoder_spdif_hold(component, false);
+		return 0;
+
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		aiu_encoder_spdif_hold(component, true);
+		return 0;
+
+	default:
+		return -EINVAL;
+	}
+}
+
+static int aiu_encoder_spdif_setup_cs_word(struct snd_soc_component *component,
+					   struct snd_pcm_hw_params *params)
+{
+	u8 cs[AIU_CS_WORD_LEN];
+	unsigned int val;
+	int ret;
+
+	ret = snd_pcm_create_iec958_consumer_hw_params(params, cs,
+						       AIU_CS_WORD_LEN);
+	if (ret < 0)
+		return ret;
+
+	/* Write the 1st half word */
+	val = cs[1] | cs[0] << 8;
+	snd_soc_component_write(component, AIU_958_CHSTAT_L0, val);
+	snd_soc_component_write(component, AIU_958_CHSTAT_R0, val);
+
+	/* Write the 2nd half word */
+	val = cs[3] | cs[2] << 8;
+	snd_soc_component_write(component, AIU_958_CHSTAT_L1, val);
+	snd_soc_component_write(component, AIU_958_CHSTAT_R1, val);
+
+	return 0;
+}
+
+static int aiu_encoder_spdif_hw_params(struct snd_pcm_substream *substream,
+				       struct snd_pcm_hw_params *params,
+				       struct snd_soc_dai *dai)
+{
+	struct snd_soc_component *component = dai->component;
+	struct aiu *aiu = snd_soc_component_get_drvdata(component);
+	unsigned int val = 0, mrate;
+	int ret;
+
+	/* Disable the clock while changing the settings */
+	aiu_encoder_spdif_divider_enable(component, false);
+
+	switch (params_physical_width(params)) {
+	case 16:
+		val |= AIU_958_MISC_MODE_16BITS;
+		val |= FIELD_PREP(AIU_958_MISC_16BITS_ALIGN, 2);
+		break;
+	case 32:
+		val |= AIU_958_MISC_MODE_32BITS;
+		break;
+	default:
+		dev_err(dai->dev, "Unsupport physical width\n");
+		return -EINVAL;
+	}
+
+	snd_soc_component_update_bits(component, AIU_958_MISC,
+				      AIU_958_MISC_NON_PCM |
+				      AIU_958_MISC_MODE_16BITS |
+				      AIU_958_MISC_16BITS_ALIGN |
+				      AIU_958_MISC_MODE_32BITS |
+				      AIU_958_MISC_FORCE_LR |
+				      AIU_958_MISC_U_FROM_STREAM,
+				      val);
+
+	/* Set the stream channel status word */
+	ret = aiu_encoder_spdif_setup_cs_word(component, params);
+	if (ret) {
+		dev_err(dai->dev, "failed to set channel status word\n");
+		return ret;
+	}
+
+	snd_soc_component_update_bits(component, AIU_CLK_CTRL,
+				      AIU_CLK_CTRL_958_DIV |
+				      AIU_CLK_CTRL_958_DIV_MORE,
+				      FIELD_PREP(AIU_CLK_CTRL_958_DIV,
+						 __ffs(AIU_958_INTERNAL_DIV)));
+
+	/* 2 * 32bits per subframe * 2 channels = 128 */
+	mrate = params_rate(params) * 128 * AIU_958_INTERNAL_DIV;
+	ret = clk_set_rate(aiu->spdif.clks[MCLK].clk, mrate);
+	if (ret) {
+		dev_err(dai->dev, "failed to set mclk rate\n");
+		return ret;
+	}
+
+	aiu_encoder_spdif_divider_enable(component, true);
+
+	return 0;
+}
+
+static int aiu_encoder_spdif_hw_free(struct snd_pcm_substream *substream,
+				     struct snd_soc_dai *dai)
+{
+	struct snd_soc_component *component = dai->component;
+
+	aiu_encoder_spdif_divider_enable(component, false);
+
+	return 0;
+}
+
+static int aiu_encoder_spdif_startup(struct snd_pcm_substream *substream,
+				     struct snd_soc_dai *dai)
+{
+	struct aiu *aiu = snd_soc_component_get_drvdata(dai->component);
+	int ret;
+
+	/*
+	 * NOTE: Make sure the spdif block is on its own divider.
+	 *
+	 * The spdif can be clocked by the i2s master clock or its own
+	 * clock. We should (in theory) change the source depending on the
+	 * origin of the data.
+	 *
+	 * However, considering the clocking scheme used on these platforms,
+	 * the master clocks will pick the same PLL source when they are
+	 * playing from the same FIFO. The clock should be in sync so, it
+	 * should not be necessary to reparent the spdif master clock.
+	 */
+	ret = clk_set_parent(aiu->spdif.clks[MCLK].clk,
+			     aiu->spdif_mclk);
+	if (ret)
+		return ret;
+
+	ret = clk_bulk_prepare_enable(aiu->spdif.clk_num, aiu->spdif.clks);
+	if (ret)
+		dev_err(dai->dev, "failed to enable spdif clocks\n");
+
+	return ret;
+}
+
+static void aiu_encoder_spdif_shutdown(struct snd_pcm_substream *substream,
+				       struct snd_soc_dai *dai)
+{
+	struct aiu *aiu = snd_soc_component_get_drvdata(dai->component);
+
+	clk_bulk_disable_unprepare(aiu->spdif.clk_num, aiu->spdif.clks);
+}
+
+const struct snd_soc_dai_ops aiu_encoder_spdif_dai_ops = {
+	.trigger	= aiu_encoder_spdif_trigger,
+	.hw_params	= aiu_encoder_spdif_hw_params,
+	.hw_free	= aiu_encoder_spdif_hw_free,
+	.startup	= aiu_encoder_spdif_startup,
+	.shutdown	= aiu_encoder_spdif_shutdown,
+};

sound/soc/meson/aiu-fifo-i2s.c

···
+// SPDX-License-Identifier: GPL-2.0
+//
+// Copyright (c) 2020 BayLibre, SAS.
+// Author: Jerome Brunet <[e-mail]>
+
+#include <linux/bitfield.h>
+#include <linux/clk.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+
+#include "aiu.h"
+#include "aiu-fifo.h"
+
+#define AIU_I2S_SOURCE_DESC_MODE_8CH	BIT(0)
+#define AIU_I2S_SOURCE_DESC_MODE_24BIT	BIT(5)
+#define AIU_I2S_SOURCE_DESC_MODE_32BIT	BIT(9)
+#define AIU_I2S_SOURCE_DESC_MODE_SPLIT	BIT(11)
+#define AIU_MEM_I2S_MASKS_IRQ_BLOCK	GENMASK(31, 16)
+#define AIU_MEM_I2S_CONTROL_MODE_16BIT	BIT(6)
+#define AIU_MEM_I2S_BUF_CNTL_INIT	BIT(0)
+#define AIU_RST_SOFT_I2S_FAST		BIT(0)
+
+#define AIU_FIFO_I2S_BLOCK		256
+
+static struct snd_pcm_hardware fifo_i2s_pcm = {
+	.info = (SNDRV_PCM_INFO_INTERLEAVED |
+		 SNDRV_PCM_INFO_MMAP |
+		 SNDRV_PCM_INFO_MMAP_VALID |
+		 SNDRV_PCM_INFO_PAUSE),
+	.formats = AIU_FORMATS,
+	.rate_min = 5512,
+	.rate_max = 192000,
+	.channels_min = 2,
+	.channels_max = 8,
+	.period_bytes_min = AIU_FIFO_I2S_BLOCK,
+	.period_bytes_max = AIU_FIFO_I2S_BLOCK * USHRT_MAX,
+	.periods_min = 2,
+	.periods_max = UINT_MAX,
+
+	/* No real justification for this */
+	.buffer_bytes_max = 1 * 1024 * 1024,
+};
+
+static int aiu_fifo_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
+				struct snd_soc_dai *dai)
+{
+	struct snd_soc_component *component = dai->component;
+	unsigned int val;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		snd_soc_component_write(component, AIU_RST_SOFT,
+					AIU_RST_SOFT_I2S_FAST);
+		snd_soc_component_read(component, AIU_I2S_SYNC, &val);
+		break;
+	}
+
+	return aiu_fifo_trigger(substream, cmd, dai);
+}
+
+static int aiu_fifo_i2s_prepare(struct snd_pcm_substream *substream,
+				struct snd_soc_dai *dai)
+{
+	struct snd_soc_component *component = dai->component;
+	int ret;
+
+	ret = aiu_fifo_prepare(substream, dai);
+	if (ret)
+		return ret;
+
+	snd_soc_component_update_bits(component,
+				      AIU_MEM_I2S_BUF_CNTL,
+				      AIU_MEM_I2S_BUF_CNTL_INIT,
+				      AIU_MEM_I2S_BUF_CNTL_INIT);
+	snd_soc_component_update_bits(component,
+				      AIU_MEM_I2S_BUF_CNTL,
+				      AIU_MEM_I2S_BUF_CNTL_INIT, 0);
+
+	return 0;
+}
+
+static int aiu_fifo_i2s_hw_params(struct snd_pcm_substream *substream,
+				  struct snd_pcm_hw_params *params,
+				  struct snd_soc_dai *dai)
+{
+	struct snd_soc_component *component = dai->component;
+	struct aiu_fifo *fifo = dai->playback_dma_data;
+	unsigned int val;
+	int ret;
+
+	ret = aiu_fifo_hw_params(substream, params, dai);
+	if (ret)
+		return ret;
+
+	switch (params_physical_width(params)) {
+	case 16:
+		val = AIU_MEM_I2S_CONTROL_MODE_16BIT;
+		break;
+	case 32:
+		val = 0;
+		break;
+	default:
+		dev_err(dai->dev, "Unsupported physical width %u\n",
+			params_physical_width(params));
+		return -EINVAL;
+	}
+
+	snd_soc_component_update_bits(component, AIU_MEM_I2S_CONTROL,
+				      AIU_MEM_I2S_CONTROL_MODE_16BIT,
+				      val);
+
+	/* Setup the irq periodicity */
+	val = params_period_bytes(params) / fifo->fifo_block;
+	val = FIELD_PREP(AIU_MEM_I2S_MASKS_IRQ_BLOCK, val);
+	snd_soc_component_update_bits(component, AIU_MEM_I2S_MASKS,
+				      AIU_MEM_I2S_MASKS_IRQ_BLOCK, val);
+
+	return 0;
+}
+
+const struct snd_soc_dai_ops aiu_fifo_i2s_dai_ops = {
+	.trigger	= aiu_fifo_i2s_trigger,
+	.prepare	= aiu_fifo_i2s_prepare,
+	.hw_params	= aiu_fifo_i2s_hw_params,
+	.hw_free	= aiu_fifo_hw_free,
+	.startup	= aiu_fifo_startup,
+	.shutdown	= aiu_fifo_shutdown,
+};
+
+int aiu_fifo_i2s_dai_probe(struct snd_soc_dai *dai)
+{
+	struct snd_soc_component *component = dai->component;
+	struct aiu *aiu = snd_soc_component_get_drvdata(component);
+	struct aiu_fifo *fifo;
+	int ret;
+
+	ret = aiu_fifo_dai_probe(dai);
+	if (ret)
+		return ret;
+
+	fifo = dai->playback_dma_data;
+
+	fifo->pcm = &fifo_i2s_pcm;
+	fifo->mem_offset = AIU_MEM_I2S_START;
+	fifo->fifo_block = AIU_FIFO_I2S_BLOCK;
+	fifo->pclk = aiu->i2s.clks[PCLK].clk;
+	fifo->irq = aiu->i2s.irq;
+
+	return 0;
+}

sound/soc/meson/aiu-fifo-spdif.c

···
+// SPDX-License-Identifier: GPL-2.0
+//
+// Copyright (c) 2020 BayLibre, SAS.
+// Author: Jerome Brunet <[e-mail]>
+
+#include <linux/clk.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+
+#include "aiu.h"
+#include "aiu-fifo.h"
+
+#define AIU_IEC958_DCU_FF_CTRL_EN		BIT(0)
+#define AIU_IEC958_DCU_FF_CTRL_AUTO_DISABLE	BIT(1)
+#define AIU_IEC958_DCU_FF_CTRL_IRQ_MODE		GENMASK(3, 2)
+#define AIU_IEC958_DCU_FF_CTRL_IRQ_OUT_THD	BIT(2)
+#define AIU_IEC958_DCU_FF_CTRL_IRQ_FRAME_READ	BIT(3)
+#define AIU_IEC958_DCU_FF_CTRL_SYNC_HEAD_EN	BIT(4)
+#define AIU_IEC958_DCU_FF_CTRL_BYTE_SEEK	BIT(5)
+#define AIU_IEC958_DCU_FF_CTRL_CONTINUE		BIT(6)
+#define AIU_MEM_IEC958_CONTROL_ENDIAN		GENMASK(5, 3)
+#define AIU_MEM_IEC958_CONTROL_RD_DDR		BIT(6)
+#define AIU_MEM_IEC958_CONTROL_MODE_16BIT	BIT(7)
+#define AIU_MEM_IEC958_CONTROL_MODE_LINEAR	BIT(8)
+#define AIU_MEM_IEC958_BUF_CNTL_INIT		BIT(0)
+
+#define AIU_FIFO_SPDIF_BLOCK			8
+
+static struct snd_pcm_hardware fifo_spdif_pcm = {
+	.info = (SNDRV_PCM_INFO_INTERLEAVED |
+		 SNDRV_PCM_INFO_MMAP |
+		 SNDRV_PCM_INFO_MMAP_VALID |
+		 SNDRV_PCM_INFO_PAUSE),
+	.formats = AIU_FORMATS,
+	.rate_min = 5512,
+	.rate_max = 192000,
+	.channels_min = 2,
+	.channels_max = 2,
+	.period_bytes_min = AIU_FIFO_SPDIF_BLOCK,
+	.period_bytes_max = AIU_FIFO_SPDIF_BLOCK * USHRT_MAX,
+	.periods_min = 2,
+	.periods_max = UINT_MAX,
+
+	/* No real justification for this */
+	.buffer_bytes_max = 1 * 1024 * 1024,
+};
+
+static void fifo_spdif_dcu_enable(struct snd_soc_component *component,
+				  bool enable)
+{
+	snd_soc_component_update_bits(component, AIU_IEC958_DCU_FF_CTRL,
+				      AIU_IEC958_DCU_FF_CTRL_EN,
+				      enable ? AIU_IEC958_DCU_FF_CTRL_EN : 0);
+}
+
+static int fifo_spdif_trigger(struct snd_pcm_substream *substream, int cmd,
+			      struct snd_soc_dai *dai)
+{
+	struct snd_soc_component *component = dai->component;
+	int ret;
+
+	ret = aiu_fifo_trigger(substream, cmd, dai);
+	if (ret)
+		return ret;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		fifo_spdif_dcu_enable(component, true);
+		break;
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+	case SNDRV_PCM_TRIGGER_STOP:
+		fifo_spdif_dcu_enable(component, false);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int fifo_spdif_prepare(struct snd_pcm_substream *substream,
+			      struct snd_soc_dai *dai)
+{
+	struct snd_soc_component *component = dai->component;
+	int ret;
+
+	ret = aiu_fifo_prepare(substream, dai);
+	if (ret)
+		return ret;
+
+	snd_soc_component_update_bits(component,
+				      AIU_MEM_IEC958_BUF_CNTL,
+				      AIU_MEM_IEC958_BUF_CNTL_INIT,
+				      AIU_MEM_IEC958_BUF_CNTL_INIT);
+	snd_soc_component_update_bits(component,
+				      AIU_MEM_IEC958_BUF_CNTL,
+				      AIU_MEM_IEC958_BUF_CNTL_INIT, 0);
+
+	return 0;
+}
+
+static int fifo_spdif_hw_params(struct snd_pcm_substream *substream,
+				struct snd_pcm_hw_params *params,
+				struct snd_soc_dai *dai)
+{
+	struct snd_soc_component *component = dai->component;
+	unsigned int val;
+	int ret;
+
+	ret = aiu_fifo_hw_params(substream, params, dai);
+	if (ret)
+		return ret;
+
+	val = AIU_MEM_IEC958_CONTROL_RD_DDR |
+	      AIU_MEM_IEC958_CONTROL_MODE_LINEAR;
+
+	switch (params_physical_width(params)) {
+	case 16:
+		val |= AIU_MEM_IEC958_CONTROL_MODE_16BIT;
+		break;
+	case 32:
+		break;
+	default:
+		dev_err(dai->dev, "Unsupported physical width %u\n",
+			params_physical_width(params));
+		return -EINVAL;
+	}
+
+	snd_soc_component_update_bits(component, AIU_MEM_IEC958_CONTROL,
+				      AIU_MEM_IEC958_CONTROL_ENDIAN |
+				      AIU_MEM_IEC958_CONTROL_RD_DDR |
+				      AIU_MEM_IEC958_CONTROL_MODE_LINEAR |
+				      AIU_MEM_IEC958_CONTROL_MODE_16BIT,
+				      val);
+
+	/* Number bytes read by the FIFO between each IRQ */
+	snd_soc_component_write(component, AIU_IEC958_BPF,
+				params_period_bytes(params));
+
+	/*
+	 * AUTO_DISABLE and SYNC_HEAD are enabled by default but
+	 * this should be disabled in PCM (uncompressed) mode
+	 */
+	snd_soc_component_update_bits(component, AIU_IEC958_DCU_FF_CTRL,
+				      AIU_IEC958_DCU_FF_CTRL_AUTO_DISABLE |
+				      AIU_IEC958_DCU_FF_CTRL_IRQ_MODE |
+				      AIU_IEC958_DCU_FF_CTRL_SYNC_HEAD_EN,
+				      AIU_IEC958_DCU_FF_CTRL_IRQ_FRAME_READ);
+
+	return 0;
+}
+
+const struct snd_soc_dai_ops aiu_fifo_spdif_dai_ops = {
+	.trigger	= fifo_spdif_trigger,
+	.prepare	= fifo_spdif_prepare,
+	.hw_params	= fifo_spdif_hw_params,
+	.hw_free	= aiu_fifo_hw_free,
+	.startup	= aiu_fifo_startup,
+	.shutdown	= aiu_fifo_shutdown,
+};
+
+int aiu_fifo_spdif_dai_probe(struct snd_soc_dai *dai)
+{
+	struct snd_soc_component *component = dai->component;
+	struct aiu *aiu = snd_soc_component_get_drvdata(component);
+	struct aiu_fifo *fifo;
+	int ret;
+
+	ret = aiu_fifo_dai_probe(dai);
+	if (ret)
+		return ret;
+
+	fifo = dai->playback_dma_data;
+
+	fifo->pcm = &fifo_spdif_pcm;
+	fifo->mem_offset = AIU_MEM_IEC958_START;
+	fifo->fifo_block = 1;
+	fifo->pclk = aiu->spdif.clks[PCLK].clk;
+	fifo->irq = aiu->spdif.irq;
+
+	return 0;
+}

sound/soc/meson/aiu-fifo.c

···
+// SPDX-License-Identifier: GPL-2.0
+//
+// Copyright (c) 2020 BayLibre, SAS.
+// Author: Jerome Brunet <[e-mail]>
+
+#include <linux/bitfield.h>
+#include <linux/clk.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+
+#include "aiu-fifo.h"
+
+#define AIU_MEM_START	0x00
+#define AIU_MEM_RD	0x04
+#define AIU_MEM_END	0x08
+#define AIU_MEM_MASKS	0x0c
+#define  AIU_MEM_MASK_CH_RD GENMASK(7, 0)
+#define  AIU_MEM_MASK_CH_MEM GENMASK(15, 8)
+#define AIU_MEM_CONTROL	0x10
+#define  AIU_MEM_CONTROL_INIT BIT(0)
+#define  AIU_MEM_CONTROL_FILL_EN BIT(1)
+#define  AIU_MEM_CONTROL_EMPTY_EN BIT(2)
+
+static struct snd_soc_dai *aiu_fifo_dai(struct snd_pcm_substream *ss)
+{
+	struct snd_soc_pcm_runtime *rtd = ss->private_data;
+
+	return asoc_rtd_to_cpu(rtd, 0);
+}
+
+snd_pcm_uframes_t aiu_fifo_pointer(struct snd_soc_component *component,
+				   struct snd_pcm_substream *substream)
+{
+	struct snd_soc_dai *dai = aiu_fifo_dai(substream);
+	struct aiu_fifo *fifo = dai->playback_dma_data;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	unsigned int addr;
+
+	snd_soc_component_read(component, fifo->mem_offset + AIU_MEM_RD,
+			       &addr);
+
+	return bytes_to_frames(runtime, addr - (unsigned int)runtime->dma_addr);
+}
+
+static void aiu_fifo_enable(struct snd_soc_dai *dai, bool enable)
+{
+	struct snd_soc_component *component = dai->component;
+	struct aiu_fifo *fifo = dai->playback_dma_data;
+	unsigned int en_mask = (AIU_MEM_CONTROL_FILL_EN |
+				AIU_MEM_CONTROL_EMPTY_EN);
+
+	snd_soc_component_update_bits(component,
+				      fifo->mem_offset + AIU_MEM_CONTROL,
+				      en_mask, enable ? en_mask : 0);
+}
+
+int aiu_fifo_trigger(struct snd_pcm_substream *substream, int cmd,
+		     struct snd_soc_dai *dai)
+{
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		aiu_fifo_enable(dai, true);
+		break;
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+	case SNDRV_PCM_TRIGGER_STOP:
+		aiu_fifo_enable(dai, false);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+int aiu_fifo_prepare(struct snd_pcm_substream *substream,
+		     struct snd_soc_dai *dai)
+{
+	struct snd_soc_component *component = dai->component;
+	struct aiu_fifo *fifo = dai->playback_dma_data;
+
+	snd_soc_component_update_bits(component,
+				      fifo->mem_offset + AIU_MEM_CONTROL,
+				      AIU_MEM_CONTROL_INIT,
+				      AIU_MEM_CONTROL_INIT);
+	snd_soc_component_update_bits(component,
+				      fifo->mem_offset + AIU_MEM_CONTROL,
+				      AIU_MEM_CONTROL_INIT, 0);
+	return 0;
+}
+
+int aiu_fifo_hw_params(struct snd_pcm_substream *substream,
+		       struct snd_pcm_hw_params *params,
+		       struct snd_soc_dai *dai)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_soc_component *component = dai->component;
+	struct aiu_fifo *fifo = dai->playback_dma_data;
+	dma_addr_t end;
+	int ret;
+
+	ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
+	if (ret < 0)
+		return ret;
+
+	/* Setup the fifo boundaries */
+	end = runtime->dma_addr + runtime->dma_bytes - fifo->fifo_block;
+	snd_soc_component_write(component, fifo->mem_offset + AIU_MEM_START,
+				runtime->dma_addr);
+	snd_soc_component_write(component, fifo->mem_offset + AIU_MEM_RD,
+				runtime->dma_addr);
+	snd_soc_component_write(component, fifo->mem_offset + AIU_MEM_END,
+				end);
+
+	/* Setup the fifo to read all the memory - no skip */
+	snd_soc_component_update_bits(component,
+				      fifo->mem_offset + AIU_MEM_MASKS,
+				      AIU_MEM_MASK_CH_RD | AIU_MEM_MASK_CH_MEM,
+				      FIELD_PREP(AIU_MEM_MASK_CH_RD, 0xff) |
+				      FIELD_PREP(AIU_MEM_MASK_CH_MEM, 0xff));
+
+	return 0;
+}
+
+int aiu_fifo_hw_free(struct snd_pcm_substream *substream,
+		     struct snd_soc_dai *dai)
+{
+	return snd_pcm_lib_free_pages(substream);
+}
+
+static irqreturn_t aiu_fifo_isr(int irq, void *dev_id)
+{
+	struct snd_pcm_substream *playback = dev_id;
+
+	snd_pcm_period_elapsed(playback);
+
+	return IRQ_HANDLED;
+}
+
+int aiu_fifo_startup(struct snd_pcm_substream *substream,
+		     struct snd_soc_dai *dai)
+{
+	struct aiu_fifo *fifo = dai->playback_dma_data;
+	int ret;
+
+	snd_soc_set_runtime_hwparams(substream, fifo->pcm);
+
+	/*
+	 * Make sure the buffer and period size are multiple of the fifo burst
+	 * size
+	 */
+	ret = snd_pcm_hw_constraint_step(substream->runtime, 0,
+					 SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
+					 fifo->fifo_block);
+	if (ret)
+		return ret;
+
+	ret = snd_pcm_hw_constraint_step(substream->runtime, 0,
+					 SNDRV_PCM_HW_PARAM_PERIOD_BYTES,
+					 fifo->fifo_block);
+	if (ret)
+		return ret;
+
+	ret = clk_prepare_enable(fifo->pclk);
+	if (ret)
+		return ret;
+
+	ret = request_irq(fifo->irq, aiu_fifo_isr, 0, dev_name(dai->dev),
+			  substream);
+	if (ret)
+		clk_disable_unprepare(fifo->pclk);
+
+	return ret;
+}
+
+void aiu_fifo_shutdown(struct snd_pcm_substream *substream,
+		       struct snd_soc_dai *dai)
+{
+	struct aiu_fifo *fifo = dai->playback_dma_data;
+
+	free_irq(fifo->irq, substream);
+	clk_disable_unprepare(fifo->pclk);
+}
+
+int aiu_fifo_pcm_new(struct snd_soc_pcm_runtime *rtd,
+		     struct snd_soc_dai *dai)
+{
+	struct snd_pcm_substream *substream =
+		rtd->pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
+	struct snd_card *card = rtd->card->snd_card;
+	struct aiu_fifo *fifo = dai->playback_dma_data;
+	size_t size = fifo->pcm->buffer_bytes_max;
+
+	snd_pcm_lib_preallocate_pages(substream,
+				      SNDRV_DMA_TYPE_DEV,
+				      card->dev, size, size);
+
+	return 0;
+}
+
+int aiu_fifo_dai_probe(struct snd_soc_dai *dai)
+{
+	struct aiu_fifo *fifo;
+
+	fifo = kzalloc(sizeof(*fifo), GFP_KERNEL);
+	if (!fifo)
+		return -ENOMEM;
+
+	dai->playback_dma_data = fifo;
+
+	return 0;
+}
+
+int aiu_fifo_dai_remove(struct snd_soc_dai *dai)
+{
+	kfree(dai->playback_dma_data);
+
+	return 0;
+}
+

sound/soc/meson/aiu-fifo.h

···
+/* SPDX-License-Identifier: (GPL-2.0 OR MIT) */
+/*
+ * Copyright (c) 2020 BayLibre, SAS.
+ * Author: Jerome Brunet <[e-mail]>
+ */
+
+#ifndef _MESON_AIU_FIFO_H
+#define _MESON_AIU_FIFO_H
+
+struct snd_pcm_hardware;
+struct snd_soc_component_driver;
+struct snd_soc_dai_driver;
+struct clk;
+struct snd_pcm_ops;
+struct snd_pcm_substream;
+struct snd_soc_dai;
+struct snd_pcm_hw_params;
+struct platform_device;
+
+struct aiu_fifo {
+	struct snd_pcm_hardware *pcm;
+	unsigned int mem_offset;
+	unsigned int fifo_block;
+	struct clk *pclk;
+	int irq;
+};
+
+int aiu_fifo_dai_probe(struct snd_soc_dai *dai);
+int aiu_fifo_dai_remove(struct snd_soc_dai *dai);
+
+snd_pcm_uframes_t aiu_fifo_pointer(struct snd_soc_component *component,
+				   struct snd_pcm_substream *substream);
+
+int aiu_fifo_trigger(struct snd_pcm_substream *substream, int cmd,
+		     struct snd_soc_dai *dai);
+int aiu_fifo_prepare(struct snd_pcm_substream *substream,
+		     struct snd_soc_dai *dai);
+int aiu_fifo_hw_params(struct snd_pcm_substream *substream,
+		       struct snd_pcm_hw_params *params,
+		       struct snd_soc_dai *dai);
+int aiu_fifo_hw_free(struct snd_pcm_substream *substream,
+		     struct snd_soc_dai *dai);
+int aiu_fifo_startup(struct snd_pcm_substream *substream,
+		     struct snd_soc_dai *dai);
+void aiu_fifo_shutdown(struct snd_pcm_substream *substream,
+		       struct snd_soc_dai *dai);
+int aiu_fifo_pcm_new(struct snd_soc_pcm_runtime *rtd,
+		     struct snd_soc_dai *dai);
+
+#endif /* _MESON_AIU_FIFO_H */

sound/soc/meson/aiu.c

···
+// SPDX-License-Identifier: GPL-2.0
+//
+// Copyright (c) 2020 BayLibre, SAS.
+// Author: Jerome Brunet <[e-mail]>
+
+#include <linux/bitfield.h>
+#include <linux/clk.h>
+#include <linux/module.h>
+#include <linux/of_platform.h>
+#include <linux/regmap.h>
+#include <linux/reset.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+
+#include <dt-bindings/sound/meson-aiu.h>
+#include "aiu.h"
+#include "aiu-fifo.h"
+
+#define AIU_I2S_MISC_958_SRC_SHIFT 3
+
+static const char * const aiu_spdif_encode_sel_texts[] = {
+	"SPDIF", "I2S",
+};
+
+static SOC_ENUM_SINGLE_DECL(aiu_spdif_encode_sel_enum, AIU_I2S_MISC,
+			    AIU_I2S_MISC_958_SRC_SHIFT,
+			    aiu_spdif_encode_sel_texts);
+
+static const struct snd_kcontrol_new aiu_spdif_encode_mux =
+	SOC_DAPM_ENUM("SPDIF Buffer Src", aiu_spdif_encode_sel_enum);
+
+static const struct snd_soc_dapm_widget aiu_cpu_dapm_widgets[] = {
+	SND_SOC_DAPM_MUX("SPDIF SRC SEL", SND_SOC_NOPM, 0, 0,
+			 &aiu_spdif_encode_mux),
+};
+
+static const struct snd_soc_dapm_route aiu_cpu_dapm_routes[] = {
+	{ "I2S Encoder Playback", NULL, "I2S FIFO Playback" },
+	{ "SPDIF SRC SEL", "SPDIF", "SPDIF FIFO Playback" },
+	{ "SPDIF SRC SEL", "I2S", "I2S FIFO Playback" },
+	{ "SPDIF Encoder Playback", NULL, "SPDIF SRC SEL" },
+};
+
+int aiu_of_xlate_dai_name(struct snd_soc_component *component,
+			  struct of_phandle_args *args,
+			  const char **dai_name,
+			  unsigned int component_id)
+{
+	struct snd_soc_dai *dai;
+	int id;
+
+	if (args->args_count != 2)
+		return -EINVAL;
+
+	if (args->args[0] != component_id)
+		return -EINVAL;
+
+	id = args->args[1];
+
+	if (id < 0 || id >= component->num_dai)
+		return -EINVAL;
+
+	for_each_component_dais(component, dai) {
+		if (id == 0)
+			break;
+		id--;
+	}
+
+	*dai_name = dai->driver->name;
+
+	return 0;
+}
+
+static int aiu_cpu_of_xlate_dai_name(struct snd_soc_component *component,
+				     struct of_phandle_args *args,
+				     const char **dai_name)
+{
+	return aiu_of_xlate_dai_name(component, args, dai_name, AIU_CPU);
+}
+
+static int aiu_cpu_component_probe(struct snd_soc_component *component)
+{
+	struct aiu *aiu = snd_soc_component_get_drvdata(component);
+
+	/* Required for the SPDIF Source control operation */
+	return clk_prepare_enable(aiu->i2s.clks[PCLK].clk);
+}
+
+static void aiu_cpu_component_remove(struct snd_soc_component *component)
+{
+	struct aiu *aiu = snd_soc_component_get_drvdata(component);
+
+	clk_disable_unprepare(aiu->i2s.clks[PCLK].clk);
+}
+
+static const struct snd_soc_component_driver aiu_cpu_component = {
+	.name			= "AIU CPU",
+	.dapm_widgets		= aiu_cpu_dapm_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(aiu_cpu_dapm_widgets),
+	.dapm_routes		= aiu_cpu_dapm_routes,
+	.num_dapm_routes	= ARRAY_SIZE(aiu_cpu_dapm_routes),
+	.of_xlate_dai_name	= aiu_cpu_of_xlate_dai_name,
+	.pointer		= aiu_fifo_pointer,
+	.probe			= aiu_cpu_component_probe,
+	.remove			= aiu_cpu_component_remove,
+};
+
+static struct snd_soc_dai_driver aiu_cpu_dai_drv[] = {
+	[CPU_I2S_FIFO] = {
+		.name = "I2S FIFO",
+		.playback = {
+			.stream_name	= "I2S FIFO Playback",
+			.channels_min	= 2,
+			.channels_max	= 8,
+			.rates		= SNDRV_PCM_RATE_CONTINUOUS,
+			.rate_min	= 5512,
+			.rate_max	= 192000,
+			.formats	= AIU_FORMATS,
+		},
+		.ops		= &aiu_fifo_i2s_dai_ops,
+		.pcm_new	= aiu_fifo_pcm_new,
+		.probe		= aiu_fifo_i2s_dai_probe,
+		.remove		= aiu_fifo_dai_remove,
+	},
+	[CPU_SPDIF_FIFO] = {
+		.name = "SPDIF FIFO",
+		.playback = {
+			.stream_name	= "SPDIF FIFO Playback",
+			.channels_min	= 2,
+			.channels_max	= 2,
+			.rates		= SNDRV_PCM_RATE_CONTINUOUS,
+			.rate_min	= 5512,
+			.rate_max	= 192000,
+			.formats	= AIU_FORMATS,
+		},
+		.ops		= &aiu_fifo_spdif_dai_ops,
+		.pcm_new	= aiu_fifo_pcm_new,
+		.probe		= aiu_fifo_spdif_dai_probe,
+		.remove		= aiu_fifo_dai_remove,
+	},
+	[CPU_I2S_ENCODER] = {
+		.name = "I2S Encoder",
+		.playback = {
+			.stream_name = "I2S Encoder Playback",
+			.channels_min = 2,
+			.channels_max = 8,
+			.rates = SNDRV_PCM_RATE_8000_192000,
+			.formats = AIU_FORMATS,
+		},
+		.ops = &aiu_encoder_i2s_dai_ops,
+	},
+	[CPU_SPDIF_ENCODER] = {
+		.name = "SPDIF Encoder",
+		.playback = {
+			.stream_name = "SPDIF Encoder Playback",
+			.channels_min = 2,
+			.channels_max = 2,
+			.rates = (SNDRV_PCM_RATE_32000  |
+				  SNDRV_PCM_RATE_44100  |
+				  SNDRV_PCM_RATE_48000  |
+				  SNDRV_PCM_RATE_88200  |
+				  SNDRV_PCM_RATE_96000  |
+				  SNDRV_PCM_RATE_176400 |
+				  SNDRV_PCM_RATE_192000),
+			.formats = AIU_FORMATS,
+		},
+		.ops = &aiu_encoder_spdif_dai_ops,
+	}
+};
+
+static const struct regmap_config aiu_regmap_cfg = {
+	.reg_bits	= 32,
+	.val_bits	= 32,
+	.reg_stride	= 4,
+	.max_register	= 0x2ac,
+};
+
+static int aiu_clk_bulk_get(struct device *dev,
+			    const char * const *ids,
+			    unsigned int num,
+			    struct aiu_interface *interface)
+{
+	struct clk_bulk_data *clks;
+	int i, ret;
+
+	clks = devm_kcalloc(dev, num, sizeof(*clks), GFP_KERNEL);
+	if (!clks)
+		return -ENOMEM;
+
+	for (i = 0; i < num; i++)
+		clks[i].id = ids[i];
+
+	ret = devm_clk_bulk_get(dev, num, clks);
+	if (ret < 0)
+		return ret;
+
+	interface->clks = clks;
+	interface->clk_num = num;
+	return 0;
+}
+
+static const char * const aiu_i2s_ids[] = {
+	[PCLK]	= "i2s_pclk",
+	[AOCLK]	= "i2s_aoclk",
+	[MCLK]	= "i2s_mclk",
+	[MIXER]	= "i2s_mixer",
+};
+
+static const char * const aiu_spdif_ids[] = {
+	[PCLK]	= "spdif_pclk",
+	[AOCLK]	= "spdif_aoclk",
+	[MCLK]	= "spdif_mclk_sel"
+};
+
+static int aiu_clk_get(struct device *dev)
+{
+	struct aiu *aiu = dev_get_drvdata(dev);
+	int ret;
+
+	aiu->pclk = devm_clk_get(dev, "pclk");
+	if (IS_ERR(aiu->pclk)) {
+		if (PTR_ERR(aiu->pclk) != -EPROBE_DEFER)
+			dev_err(dev, "Can't get the aiu pclk\n");
+		return PTR_ERR(aiu->pclk);
+	}
+
+	aiu->spdif_mclk = devm_clk_get(dev, "spdif_mclk");
+	if (IS_ERR(aiu->spdif_mclk)) {
+		if (PTR_ERR(aiu->spdif_mclk) != -EPROBE_DEFER)
+			dev_err(dev, "Can't get the aiu spdif master clock\n");
+		return PTR_ERR(aiu->spdif_mclk);
+	}
+
+	ret = aiu_clk_bulk_get(dev, aiu_i2s_ids, ARRAY_SIZE(aiu_i2s_ids),
+			       &aiu->i2s);
+	if (ret) {
+		if (ret != -EPROBE_DEFER)
+			dev_err(dev, "Can't get the i2s clocks\n");
+		return ret;
+	}
+
+	ret = aiu_clk_bulk_get(dev, aiu_spdif_ids, ARRAY_SIZE(aiu_spdif_ids),
+			       &aiu->spdif);
+	if (ret) {
+		if (ret != -EPROBE_DEFER)
+			dev_err(dev, "Can't get the spdif clocks\n");
+		return ret;
+	}
+
+	ret = clk_prepare_enable(aiu->pclk);
+	if (ret) {
+		dev_err(dev, "peripheral clock enable failed\n");
+		return ret;
+	}
+
+	ret = devm_add_action_or_reset(dev,
+				       (void(*)(void *))clk_disable_unprepare,
+				       aiu->pclk);
+	if (ret)
+		dev_err(dev, "failed to add reset action on pclk");
+
+	return ret;
+}
+
+static int aiu_probe(struct platform_device *pdev)
+{
+	struct device *dev = &pdev->dev;
+	void __iomem *regs;
+	struct regmap *map;
+	struct aiu *aiu;
+	int ret;
+
+	aiu = devm_kzalloc(dev, sizeof(*aiu), GFP_KERNEL);
+	if (!aiu)
+		return -ENOMEM;
+
+	aiu->platform = device_get_match_data(dev);
+	if (!aiu->platform)
+		return -ENODEV;
+
+	platform_set_drvdata(pdev, aiu);
+
+	ret = device_reset(dev);
+	if (ret) {
+		if (ret != -EPROBE_DEFER)
+			dev_err(dev, "Failed to reset device\n");
+		return ret;
+	}
+
+	regs = devm_platform_ioremap_resource(pdev, 0);
+	if (IS_ERR(regs))
+		return PTR_ERR(regs);
+
+	map = devm_regmap_init_mmio(dev, regs, &aiu_regmap_cfg);
+	if (IS_ERR(map)) {
+		dev_err(dev, "failed to init regmap: %ld\n",
+			PTR_ERR(map));
+		return PTR_ERR(map);
+	}
+
+	aiu->i2s.irq = platform_get_irq_byname(pdev, "i2s");
+	if (aiu->i2s.irq < 0)
+		return aiu->i2s.irq;
+
+	aiu->spdif.irq = platform_get_irq_byname(pdev, "spdif");
+	if (aiu->spdif.irq < 0)
+		return aiu->spdif.irq;
+
+	ret = aiu_clk_get(dev);
+	if (ret)
+		return ret;
+
+	/* Register the cpu component of the aiu */
+	ret = snd_soc_register_component(dev, &aiu_cpu_component,
+					 aiu_cpu_dai_drv,
+					 ARRAY_SIZE(aiu_cpu_dai_drv));
+	if (ret) {
+		dev_err(dev, "Failed to register cpu component\n");
+		return ret;
+	}
+
+	/* Register the hdmi codec control component */
+	ret = aiu_hdmi_ctrl_register_component(dev);
+	if (ret) {
+		dev_err(dev, "Failed to register hdmi control component\n");
+		goto err;
+	}
+
+	/* Register the internal dac control component on gxl */
+	if (aiu->platform->has_acodec) {
+		ret = aiu_acodec_ctrl_register_component(dev);
+		if (ret) {
+			dev_err(dev,
+			    "Failed to register acodec control component\n");
+			goto err;
+		}
+	}
+
+	return 0;
+err:
+	snd_soc_unregister_component(dev);
+	return ret;
+}
+
+static int aiu_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_component(&pdev->dev);
+
+	return 0;
+}
+
+static const struct aiu_platform_data aiu_gxbb_pdata = {
+	.has_acodec = false,
+	.has_clk_ctrl_more_i2s_div = true,
+};
+
+static const struct aiu_platform_data aiu_gxl_pdata = {
+	.has_acodec = true,
+	.has_clk_ctrl_more_i2s_div = true,
+};
+
+static const struct aiu_platform_data aiu_meson8_pdata = {
+	.has_acodec = false,
+	.has_clk_ctrl_more_i2s_div = false,
+};
+
+static const struct of_device_id aiu_of_match[] = {
+	{ .compatible = "amlogic,aiu-gxbb", .data = &aiu_gxbb_pdata },
+	{ .compatible = "amlogic,aiu-gxl", .data = &aiu_gxl_pdata },
+	{ .compatible = "amlogic,aiu-meson8", .data = &aiu_meson8_pdata },
+	{ .compatible = "amlogic,aiu-meson8b", .data = &aiu_meson8_pdata },
+	{}
+};
+MODULE_DEVICE_TABLE(of, aiu_of_match);
+
+static struct platform_driver aiu_pdrv = {
+	.probe = aiu_probe,
+	.remove = aiu_remove,
+	.driver = {
+		.name = "meson-aiu",
+		.of_match_table = aiu_of_match,
+	},
+};
+module_platform_driver(aiu_pdrv);
+
+MODULE_DESCRIPTION("Meson AIU Driver");
+MODULE_AUTHOR("Jerome Brunet <[e-mail]>");
+MODULE_LICENSE("GPL v2");

sound/soc/meson/aiu.h

···
+/* SPDX-License-Identifier: (GPL-2.0 OR MIT) */
+/*
+ * Copyright (c) 2018 BayLibre, SAS.
+ * Author: Jerome Brunet <[e-mail]>
+ */
+
+#ifndef _MESON_AIU_H
+#define _MESON_AIU_H
+
+struct clk;
+struct clk_bulk_data;
+struct device;
+struct of_phandle_args;
+struct snd_soc_dai;
+struct snd_soc_dai_ops;
+
+enum aiu_clk_ids {
+	PCLK = 0,
+	AOCLK,
+	MCLK,
+	MIXER
+};
+
+struct aiu_interface {
+	struct clk_bulk_data *clks;
+	unsigned int clk_num;
+	int irq;
+};
+
+struct aiu_platform_data {
+	bool has_acodec;
+	bool has_clk_ctrl_more_i2s_div;
+};
+
+struct aiu {
+	struct clk *pclk;
+	struct clk *spdif_mclk;
+	struct aiu_interface i2s;
+	struct aiu_interface spdif;
+	const struct aiu_platform_data *platform;
+};
+
+#define AIU_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |	\
+		     SNDRV_PCM_FMTBIT_S20_LE |	\
+		     SNDRV_PCM_FMTBIT_S24_LE)
+
+int aiu_of_xlate_dai_name(struct snd_soc_component *component,
+			  struct of_phandle_args *args,
+			  const char **dai_name,
+			  unsigned int component_id);
+
+int aiu_hdmi_ctrl_register_component(struct device *dev);
+int aiu_acodec_ctrl_register_component(struct device *dev);
+
+int aiu_fifo_i2s_dai_probe(struct snd_soc_dai *dai);
+int aiu_fifo_spdif_dai_probe(struct snd_soc_dai *dai);
+
+extern const struct snd_soc_dai_ops aiu_fifo_i2s_dai_ops;
+extern const struct snd_soc_dai_ops aiu_fifo_spdif_dai_ops;
+extern const struct snd_soc_dai_ops aiu_encoder_i2s_dai_ops;
+extern const struct snd_soc_dai_ops aiu_encoder_spdif_dai_ops;
+
+#define AIU_IEC958_BPF			0x000
+#define AIU_958_MISC			0x010
+#define AIU_IEC958_DCU_FF_CTRL		0x01c
+#define AIU_958_CHSTAT_L0		0x020
+#define AIU_958_CHSTAT_L1		0x024
+#define AIU_958_CTRL			0x028
+#define AIU_I2S_SOURCE_DESC		0x034
+#define AIU_I2S_DAC_CFG			0x040
+#define AIU_I2S_SYNC			0x044
+#define AIU_I2S_MISC			0x048
+#define AIU_RST_SOFT			0x054
+#define AIU_CLK_CTRL			0x058
+#define AIU_CLK_CTRL_MORE		0x064
+#define AIU_CODEC_DAC_LRCLK_CTRL	0x0a0
+#define AIU_HDMI_CLK_DATA_CTRL		0x0a8
+#define AIU_ACODEC_CTRL			0x0b0
+#define AIU_958_CHSTAT_R0		0x0c0
+#define AIU_958_CHSTAT_R1		0x0c4
+#define AIU_MEM_I2S_START		0x180
+#define AIU_MEM_I2S_MASKS		0x18c
+#define AIU_MEM_I2S_CONTROL		0x190
+#define AIU_MEM_IEC958_START		0x194
+#define AIU_MEM_IEC958_CONTROL		0x1a4
+#define AIU_MEM_I2S_BUF_CNTL		0x1d8
+#define AIU_MEM_IEC958_BUF_CNTL		0x1fc
+
+#endif /* _MESON_AIU_H */

sound/soc/meson/axg-card.c

···
 #include <sound/soc-dai.h>
 
 #include "axg-tdm.h"
-
-struct axg_card {
-	struct snd_soc_card card;
-	void **link_data;
-};
+#include "meson-card.h"
 
 struct axg_dai_link_tdm_mask {
 	u32 tx;
···
 	.channels_max = 8,
 };
 
-#define PREFIX "amlogic,"
-
-static int axg_card_reallocate_links(struct axg_card *priv,
-				     unsigned int num_links)
-{
-	struct snd_soc_dai_link *links;
-	void **ldata;
-
-	links = krealloc(priv->card.dai_link,
-			 num_links * sizeof(*priv->card.dai_link),
-			 GFP_KERNEL | __GFP_ZERO);
-	ldata = krealloc(priv->link_data,
-			 num_links * sizeof(*priv->link_data),
-			 GFP_KERNEL | __GFP_ZERO);
-
-	if (!links || !ldata) {
-		dev_err(priv->card.dev, "failed to allocate links\n");
-		return -ENOMEM;
-	}
-
-	priv->card.dai_link = links;
-	priv->link_data = ldata;
-	priv->card.num_links = num_links;
-	return 0;
-}
-
-static int axg_card_parse_dai(struct snd_soc_card *card,
-			      struct device_node *node,
-			      struct device_node **dai_of_node,
-			      const char **dai_name)
-{
-	struct of_phandle_args args;
-	int ret;
-
-	if (!dai_name || !dai_of_node || !node)
-		return -EINVAL;
-
-	ret = of_parse_phandle_with_args(node, "sound-dai",
-					 "#sound-dai-cells", 0, &args);
-	if (ret) {
-		if (ret != -EPROBE_DEFER)
-			dev_err(card->dev, "can't parse dai %d\n", ret);
-		return ret;
-	}
-	*dai_of_node = args.np;
-
-	return snd_soc_get_dai_name(&args, dai_name);
-}
-
-static int axg_card_set_link_name(struct snd_soc_card *card,
-				  struct snd_soc_dai_link *link,
-				  struct device_node *node,
-				  const char *prefix)
-{
-	char *name = devm_kasprintf(card->dev, GFP_KERNEL, "%s.%s",
-				    prefix, node->full_name);
-	if (!name)
-		return -ENOMEM;
-
-	link->name = name;
-	link->stream_name = name;
-
-	return 0;
-}
-
-static void axg_card_clean_references(struct axg_card *priv)
-{
-	struct snd_soc_card *card = &priv->card;
-	struct snd_soc_dai_link *link;
-	struct snd_soc_dai_link_component *codec;
-	struct snd_soc_aux_dev *aux;
-	int i, j;
-
-	if (card->dai_link) {
-		for_each_card_prelinks(card, i, link) {
-			if (link->cpus)
-				of_node_put(link->cpus->of_node);
-			for_each_link_codecs(link, j, codec)
-				of_node_put(codec->of_node);
-		}
-	}
-
-	if (card->aux_dev) {
-		for_each_card_pre_auxs(card, i, aux)
-			of_node_put(aux->dlc.of_node);
-	}
-
-	kfree(card->dai_link);
-	kfree(priv->link_data);
-}
-
-static int axg_card_add_aux_devices(struct snd_soc_card *card)
-{
-	struct device_node *node = card->dev->of_node;
-	struct snd_soc_aux_dev *aux;
-	int num, i;
-
-	num = of_count_phandle_with_args(node, "audio-aux-devs", NULL);
-	if (num == -ENOENT) {
-		/*
-		 * It is ok to have no auxiliary devices but for this card it
-		 * is a strange situtation. Let's warn the about it.
-		 */
-		dev_warn(card->dev, "card has no auxiliary devices\n");
-		return 0;
-	} else if (num < 0) {
-		dev_err(card->dev, "error getting auxiliary devices: %d\n",
-			num);
-		return num;
-	}
-
-	aux = devm_kcalloc(card->dev, num, sizeof(*aux), GFP_KERNEL);
-	if (!aux)
-		return -ENOMEM;
-	card->aux_dev = aux;
-	card->num_aux_devs = num;
-
-	for_each_card_pre_auxs(card, i, aux) {
-		aux->dlc.of_node =
-			of_parse_phandle(node, "audio-aux-devs", i);
-		if (!aux->dlc.of_node)
-			return -EINVAL;
-	}
-
-	return 0;
-}
-
 static int axg_card_tdm_be_hw_params(struct snd_pcm_substream *substream,
 				     struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct axg_card *priv = snd_soc_card_get_drvdata(rtd->card);
+	struct meson_card *priv = snd_soc_card_get_drvdata(rtd->card);
 	struct axg_dai_link_tdm_data *be =
 		(struct axg_dai_link_tdm_data *)priv->link_data[rtd->num];
-	struct snd_soc_dai *codec_dai;
-	unsigned int mclk;
-	int ret, i;
 
-	if (be->mclk_fs) {
-		mclk = params_rate(params) * be->mclk_fs;
-
-		for_each_rtd_codec_dai(rtd, i, codec_dai) {
-			ret = snd_soc_dai_set_sysclk(codec_dai, 0, mclk,
-						     SND_SOC_CLOCK_IN);
-			if (ret && ret != -ENOTSUPP)
-				return ret;
-		}
-
-		ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, 0, mclk,
-					     SND_SOC_CLOCK_OUT);
-		if (ret && ret != -ENOTSUPP)
-			return ret;
-	}
-
-	return 0;
+	return meson_card_i2s_set_sysclk(substream, params, be->mclk_fs);
 }
 
 static const struct snd_soc_ops axg_card_tdm_be_ops = {
···
 
 static int axg_card_tdm_dai_init(struct snd_soc_pcm_runtime *rtd)
 {
-	struct axg_card *priv = snd_soc_card_get_drvdata(rtd->card);
+	struct meson_card *priv = snd_soc_card_get_drvdata(rtd->card);
 	struct axg_dai_link_tdm_data *be =
 		(struct axg_dai_link_tdm_data *)priv->link_data[rtd->num];
 	struct snd_soc_dai *codec_dai;
 	int ret, i;
 
-	for_each_rtd_codec_dai(rtd, i, codec_dai) {
+	for_each_rtd_codec_dais(rtd, i, codec_dai) {
 		ret = snd_soc_dai_set_tdm_slot(codec_dai,
 					       be->codec_masks[i].tx,
 					       be->codec_masks[i].rx,
···
 		}
 	}
 
-	ret = axg_tdm_set_tdm_slots(rtd->cpu_dai, be->tx_mask, be->rx_mask,
+	ret = axg_tdm_set_tdm_slots(asoc_rtd_to_cpu(rtd, 0), be->tx_mask, be->rx_mask,
 				    be->slots, be->slot_width);
 	if (ret) {
-		dev_err(rtd->cpu_dai->dev, "setting tdm link slots failed\n");
+		dev_err(asoc_rtd_to_cpu(rtd, 0)->dev, "setting tdm link slots failed\n");
 		return ret;
 	}
 
···
 
 static int axg_card_tdm_dai_lb_init(struct snd_soc_pcm_runtime *rtd)
 {
-	struct axg_card *priv = snd_soc_card_get_drvdata(rtd->card);
+	struct meson_card *priv = snd_soc_card_get_drvdata(rtd->card);
 	struct axg_dai_link_tdm_data *be =
 		(struct axg_dai_link_tdm_data *)priv->link_data[rtd->num];
 	int ret;
 
 	/* The loopback rx_mask is the pad tx_mask */
-	ret = axg_tdm_set_tdm_slots(rtd->cpu_dai, NULL, be->tx_mask,
+	ret = axg_tdm_set_tdm_slots(asoc_rtd_to_cpu(rtd, 0), NULL, be->tx_mask,
 				    be->slots, be->slot_width);
 	if (ret) {
-		dev_err(rtd->cpu_dai->dev, "setting tdm link slots failed\n");
+		dev_err(asoc_rtd_to_cpu(rtd, 0)->dev, "setting tdm link slots failed\n");
 		return ret;
 	}
 
···
 static int axg_card_add_tdm_loopback(struct snd_soc_card *card,
 				     int *index)
 {
-	struct axg_card *priv = snd_soc_card_get_drvdata(card);
+	struct meson_card *priv = snd_soc_card_get_drvdata(card);
 	struct snd_soc_dai_link *pad = &card->dai_link[*index];
 	struct snd_soc_dai_link *lb;
 	struct snd_soc_dai_link_component *dlc;
 	int ret;
 
 	/* extend links */
-	ret = axg_card_reallocate_links(priv, card->num_links + 1);
+	ret = meson_card_reallocate_links(card, card->num_links + 1);
 	if (ret)
 		return ret;
 
···
 	*index += 1;
 
 	return 0;
-}
-
-static unsigned int axg_card_parse_daifmt(struct device_node *node,
-					  struct device_node *cpu_node)
-{
-	struct device_node *bitclkmaster = NULL;
-	struct device_node *framemaster = NULL;
-	unsigned int daifmt;
-
-	daifmt = snd_soc_of_parse_daifmt(node, PREFIX,
-					 &bitclkmaster, &framemaster);
-	daifmt &= ~SND_SOC_DAIFMT_MASTER_MASK;
-
-	/* If no master is provided, default to cpu master */
-	if (!bitclkmaster || bitclkmaster == cpu_node) {
-		daifmt |= (!framemaster || framemaster == cpu_node) ?
-			SND_SOC_DAIFMT_CBS_CFS : SND_SOC_DAIFMT_CBS_CFM;
-	} else {
-		daifmt |= (!framemaster || framemaster == cpu_node) ?
-			SND_SOC_DAIFMT_CBM_CFS : SND_SOC_DAIFMT_CBM_CFM;
-	}
-
-	of_node_put(bitclkmaster);
-	of_node_put(framemaster);
-
-	return daifmt;
 }
 
 static int axg_card_parse_cpu_tdm_slots(struct snd_soc_card *card,
···
 			      struct device_node *node,
 			      int *index)
 {
-	struct axg_card *priv = snd_soc_card_get_drvdata(card);
+	struct meson_card *priv = snd_soc_card_get_drvdata(card);
 	struct snd_soc_dai_link *link = &card->dai_link[*index];
 	struct axg_dai_link_tdm_data *be;
 	int ret;
···
 	/* Setup tdm link */
 	link->ops = &axg_card_tdm_be_ops;
 	link->init = axg_card_tdm_dai_init;
-	link->dai_fmt = axg_card_parse_daifmt(node, link->cpus->of_node);
+	link->dai_fmt = meson_card_parse_daifmt(node, link->cpus->of_node);
 
 	of_property_read_u32(node, "mclk-fs", &be->mclk_fs);
 
···
 	return 0;
 }
 
-static int axg_card_set_be_link(struct snd_soc_card *card,
-				struct snd_soc_dai_link *link,
-				struct device_node *node)
-{
-	struct snd_soc_dai_link_component *codec;
-	struct device_node *np;
-	int ret, num_codecs;
-
-	link->no_pcm = 1;
-	link->dpcm_playback = 1;
-	link->dpcm_capture = 1;
-
-	num_codecs = of_get_child_count(node);
-	if (!num_codecs) {
-		dev_err(card->dev, "be link %s has no codec\n",
-			node->full_name);
-		return -EINVAL;
-	}
-
-	codec = devm_kcalloc(card->dev, num_codecs, sizeof(*codec), GFP_KERNEL);
-	if (!codec)
-		return -ENOMEM;
-
-	link->codecs = codec;
-	link->num_codecs = num_codecs;
-
-	for_each_child_of_node(node, np) {
-		ret = axg_card_parse_dai(card, np, &codec->of_node,
-					 &codec->dai_name);
-		if (ret) {
-			of_node_put(np);
-			return ret;
-		}
-
-		codec++;
-	}
-
-	ret = axg_card_set_link_name(card, link, node, "be");
-	if (ret)
-		dev_err(card->dev, "error setting %pOFn link name\n", np);
-
-	return ret;
-}
-
-static int axg_card_set_fe_link(struct snd_soc_card *card,
-				struct snd_soc_dai_link *link,
-				struct device_node *node,
-				bool is_playback)
-{
-	struct snd_soc_dai_link_component *codec;
-
-	codec = devm_kzalloc(card->dev, sizeof(*codec), GFP_KERNEL);
-	if (!codec)
-		return -ENOMEM;
-
-	link->codecs = codec;
-	link->num_codecs = 1;
-
-	link->dynamic = 1;
-	link->dpcm_merged_format = 1;
-	link->dpcm_merged_chan = 1;
-	link->dpcm_merged_rate = 1;
-	link->codecs->dai_name = "snd-soc-dummy-dai";
-	link->codecs->name = "snd-soc-dummy";
-
-	if (is_playback)
-		link->dpcm_playback = 1;
-	else
-		link->dpcm_capture = 1;
-
-	return axg_card_set_link_name(card, link, node, "fe");
-}
-
 static int axg_card_cpu_is_capture_fe(struct device_node *np)
 {
-	return of_device_is_compatible(np, PREFIX "axg-toddr");
+	return of_device_is_compatible(np, DT_PREFIX "axg-toddr");
 }
 
 static int axg_card_cpu_is_playback_fe(struct device_node *np)
 {
-	return of_device_is_compatible(np, PREFIX "axg-frddr");
+	return of_device_is_compatible(np, DT_PREFIX "axg-frddr");
 }
 
 static int axg_card_cpu_is_tdm_iface(struct device_node *np)
 {
-	return of_device_is_compatible(np, PREFIX "axg-tdm-iface");
+	return of_device_is_compatible(np, DT_PREFIX "axg-tdm-iface");
 }
 
 static int axg_card_cpu_is_codec(struct device_node *np)
 {
-	return of_device_is_compatible(np, PREFIX "g12a-tohdmitx");
+	return of_device_is_compatible(np, DT_PREFIX "g12a-tohdmitx") ||
+		of_device_is_compatible(np, DT_PREFIX "g12a-toacodec");
 }
 
 static int axg_card_add_link(struct snd_soc_card *card, struct device_node *np,
···
 	dai_link->cpus = cpu;
 	dai_link->num_cpus = 1;
 
-	ret = axg_card_parse_dai(card, np, &dai_link->cpus->of_node,
-				 &dai_link->cpus->dai_name);
+	ret = meson_card_parse_dai(card, np, &dai_link->cpus->of_node,
+				   &dai_link->cpus->dai_name);
 	if (ret)
 		return ret;
 
 	if (axg_card_cpu_is_playback_fe(dai_link->cpus->of_node))
-		ret = axg_card_set_fe_link(card, dai_link, np, true);
+		ret = meson_card_set_fe_link(card, dai_link, np, true);
 	else if (axg_card_cpu_is_capture_fe(dai_link->cpus->of_node))
-		ret = axg_card_set_fe_link(card, dai_link, np, false);
+		ret = meson_card_set_fe_link(card, dai_link, np, false);
 	else
-		ret = axg_card_set_be_link(card, dai_link, np);
+		ret = meson_card_set_be_link(card, dai_link, np);
 
 	if (ret)
 		return ret;
···
 	return ret;
 }
 
-static int axg_card_add_links(struct snd_soc_card *card)
-{
-	struct axg_card *priv = snd_soc_card_get_drvdata(card);
-	struct device_node *node = card->dev->of_node;
-	struct device_node *np;
-	int num, i, ret;
-
-	num = of_get_child_count(node);
-	if (!num) {
-		dev_err(card->dev, "card has no links\n");
-		return -EINVAL;
-	}
-
-	ret = axg_card_reallocate_links(priv, num);
-	if (ret)
-		return ret;
-
-	i = 0;
-	for_each_child_of_node(node, np) {
-		ret = axg_card_add_link(card, np, &i);
-		if (ret) {
-			of_node_put(np);
-			return ret;
-		}
-
-		i++;
-	}
-
-	return 0;
-}
-
-static int axg_card_parse_of_optional(struct snd_soc_card *card,
-				      const char *propname,
-				      int (*func)(struct snd_soc_card *c,
-						  const char *p))
-{
-	/* If property is not provided, don't fail ... */
-	if (!of_property_read_bool(card->dev->of_node, propname))
-		return 0;
-
-	/* ... but do fail if it is provided and the parsing fails */
-	return func(card, propname);
-}
+static const struct meson_card_match_data axg_card_match_data = {
+	.add_link = axg_card_add_link,
+};
 
 static const struct of_device_id axg_card_of_match[] = {
-	{ .compatible = "amlogic,axg-sound-card", },
-	{}
+	{
+		.compatible = "amlogic,axg-sound-card",
+		.data = &axg_card_match_data,
+	}, {}
 };
 MODULE_DEVICE_TABLE(of, axg_card_of_match);
 
-static int axg_card_probe(struct platform_device *pdev)
-{
-	struct device *dev = &pdev->dev;
-	struct axg_card *priv;
-	int ret;
-
-	priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
-	if (!priv)
-		return -ENOMEM;
-
-	platform_set_drvdata(pdev, priv);
-	snd_soc_card_set_drvdata(&priv->card, priv);
-
-	priv->card.owner = THIS_MODULE;
-	priv->card.dev = dev;
-
-	ret = snd_soc_of_parse_card_name(&priv->card, "model");
-	if (ret < 0)
-		return ret;
-
-	ret = axg_card_parse_of_optional(&priv->card, "audio-routing",
-					 snd_soc_of_parse_audio_routing);
-	if (ret) {
-		dev_err(dev, "error while parsing routing\n");
-		return ret;
-	}
-
-	ret = axg_card_parse_of_optional(&priv->card, "audio-widgets",
-					 snd_soc_of_parse_audio_simple_widgets);
-	if (ret) {
-		dev_err(dev, "error while parsing widgets\n");
-		return ret;
-	}
-
-	ret = axg_card_add_links(&priv->card);
-	if (ret)
-		goto out_err;
-
-	ret = axg_card_add_aux_devices(&priv->card);
-	if (ret)
-		goto out_err;
-
-	ret = devm_snd_soc_register_card(dev, &priv->card);
-	if (ret)
-		goto out_err;
-
-	return 0;
-
-out_err:
-	axg_card_clean_references(priv);
-	return ret;
-}
-
-static int axg_card_remove(struct platform_device *pdev)
-{
-	struct axg_card *priv = platform_get_drvdata(pdev);
-
-	axg_card_clean_references(priv);
-
-	return 0;
-}
-
 static struct platform_driver axg_card_pdrv = {
-	.probe = axg_card_probe,
-	.remove = axg_card_remove,
+	.probe = meson_card_probe,
+	.remove = meson_card_remove,
 	.driver = {
 		.name = "axg-sound-card",
 		.of_match_table = axg_card_of_match,

sound/soc/meson/axg-fifo.c

···
 {
 	struct snd_soc_pcm_runtime *rtd = ss->private_data;
 
-	return rtd->cpu_dai;
+	return asoc_rtd_to_cpu(rtd, 0);
 }
 
 static struct axg_fifo *axg_fifo_data(struct snd_pcm_substream *ss)

sound/soc/meson/g12a-toacodec.c

···
+// SPDX-License-Identifier: GPL-2.0
+//
+// Copyright (c) 2020 BayLibre, SAS.
+// Author: Jerome Brunet <[e-mail]>
+
+#include <linux/bitfield.h>
+#include <linux/clk.h>
+#include <linux/module.h>
+#include <sound/pcm_params.h>
+#include <linux/regmap.h>
+#include <linux/regulator/consumer.h>
+#include <linux/reset.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+
+#include <dt-bindings/sound/meson-g12a-toacodec.h>
+#include "axg-tdm.h"
+#include "meson-codec-glue.h"
+
+#define G12A_TOACODEC_DRV_NAME "g12a-toacodec"
+
+#define TOACODEC_CTRL0			0x0
+#define  CTRL0_ENABLE_SHIFT		31
+#define  CTRL0_DAT_SEL_SHIFT		14
+#define  CTRL0_DAT_SEL			(0x3 << CTRL0_DAT_SEL_SHIFT)
+#define  CTRL0_LANE_SEL			12
+#define  CTRL0_LRCLK_SEL		GENMASK(9, 8)
+#define  CTRL0_BLK_CAP_INV		BIT(7)
+#define  CTRL0_BCLK_O_INV		BIT(6)
+#define  CTRL0_BCLK_SEL			GENMASK(5, 4)
+#define  CTRL0_MCLK_SEL			GENMASK(2, 0)
+
+#define TOACODEC_OUT_CHMAX		2
+
+static const char * const g12a_toacodec_mux_texts[] = {
+	"I2S A", "I2S B", "I2S C",
+};
+
+static int g12a_toacodec_mux_put_enum(struct snd_kcontrol *kcontrol,
+				      struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_component *component =
+		snd_soc_dapm_kcontrol_component(kcontrol);
+	struct snd_soc_dapm_context *dapm =
+		snd_soc_dapm_kcontrol_dapm(kcontrol);
+	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+	unsigned int mux, changed;
+
+	mux = snd_soc_enum_item_to_val(e, ucontrol->value.enumerated.item[0]);
+	changed = snd_soc_component_test_bits(component, e->reg,
+					      CTRL0_DAT_SEL,
+					      FIELD_PREP(CTRL0_DAT_SEL, mux));
+
+	if (!changed)
+		return 0;
+
+	/* Force disconnect of the mux while updating */
+	snd_soc_dapm_mux_update_power(dapm, kcontrol, 0, NULL, NULL);
+
+	snd_soc_component_update_bits(component, e->reg,
+				      CTRL0_DAT_SEL |
+				      CTRL0_LRCLK_SEL |
+				      CTRL0_BCLK_SEL,
+				      FIELD_PREP(CTRL0_DAT_SEL, mux) |
+				      FIELD_PREP(CTRL0_LRCLK_SEL, mux) |
+				      FIELD_PREP(CTRL0_BCLK_SEL, mux));
+
+	/*
+	 * FIXME:
+	 * On this soc, the glue gets the MCLK directly from the clock
+	 * controller instead of going the through the TDM interface.
+	 *
+	 * Here we assume interface A uses clock A, etc ... While it is
+	 * true for now, it could be different. Instead the glue should
+	 * find out the clock used by the interface and select the same
+	 * source. For that, we will need regmap backed clock mux which
+	 * is a work in progress
+	 */
+	snd_soc_component_update_bits(component, e->reg,
+				      CTRL0_MCLK_SEL,
+				      FIELD_PREP(CTRL0_MCLK_SEL, mux));
+
+	snd_soc_dapm_mux_update_power(dapm, kcontrol, mux, e, NULL);
+
+	return 0;
+}
+
+static SOC_ENUM_SINGLE_DECL(g12a_toacodec_mux_enum, TOACODEC_CTRL0,
+			    CTRL0_DAT_SEL_SHIFT,
+			    g12a_toacodec_mux_texts);
+
+static const struct snd_kcontrol_new g12a_toacodec_mux =
+	SOC_DAPM_ENUM_EXT("Source", g12a_toacodec_mux_enum,
+			  snd_soc_dapm_get_enum_double,
+			  g12a_toacodec_mux_put_enum);
+
+static const struct snd_kcontrol_new g12a_toacodec_out_enable =
+	SOC_DAPM_SINGLE_AUTODISABLE("Switch", TOACODEC_CTRL0,
+				    CTRL0_ENABLE_SHIFT, 1, 0);
+
+static const struct snd_soc_dapm_widget g12a_toacodec_widgets[] = {
+	SND_SOC_DAPM_MUX("SRC", SND_SOC_NOPM, 0, 0,
+			 &g12a_toacodec_mux),
+	SND_SOC_DAPM_SWITCH("OUT EN", SND_SOC_NOPM, 0, 0,
+			    &g12a_toacodec_out_enable),
+};
+
+static int g12a_toacodec_input_hw_params(struct snd_pcm_substream *substream,
+					 struct snd_pcm_hw_params *params,
+					 struct snd_soc_dai *dai)
+{
+	struct meson_codec_glue_input *data;
+	int ret;
+
+	ret = meson_codec_glue_input_hw_params(substream, params, dai);
+	if (ret)
+		return ret;
+
+	/* The glue will provide 1 lane out of the 4 to the output */
+	data = meson_codec_glue_input_get_data(dai);
+	data->params.channels_min = min_t(unsigned int, TOACODEC_OUT_CHMAX,
+					data->params.channels_min);
+	data->params.channels_max = min_t(unsigned int, TOACODEC_OUT_CHMAX,
+					data->params.channels_max);
+
+	return 0;
+}
+
+static const struct snd_soc_dai_ops g12a_toacodec_input_ops = {
+	.hw_params	= g12a_toacodec_input_hw_params,
+	.set_fmt	= meson_codec_glue_input_set_fmt,
+};
+
+static const struct snd_soc_dai_ops g12a_toacodec_output_ops = {
+	.startup	= meson_codec_glue_output_startup,
+};
+
+#define TOACODEC_STREAM(xname, xsuffix, xchmax)			\
+{								\
+	.stream_name	= xname " " xsuffix,			\
+	.channels_min	= 1,					\
+	.channels_max	= (xchmax),				\
+	.rate_min       = 5512,					\
+	.rate_max	= 192000,				\
+	.formats	= AXG_TDM_FORMATS,			\
+}
+
+#define TOACODEC_INPUT(xname, xid) {					\
+	.name = xname,							\
+	.id = (xid),							\
+	.playback = TOACODEC_STREAM(xname, "Playback", 8),		\
+	.ops = &g12a_toacodec_input_ops,				\
+	.probe = meson_codec_glue_input_dai_probe,			\
+	.remove = meson_codec_glue_input_dai_remove,			\
+}
+
+#define TOACODEC_OUTPUT(xname, xid) {					\
+	.name = xname,							\
+	.id = (xid),							\
+	.capture = TOACODEC_STREAM(xname, "Capture", TOACODEC_OUT_CHMAX), \
+	.ops = &g12a_toacodec_output_ops,				\
+}
+
+static struct snd_soc_dai_driver g12a_toacodec_dai_drv[] = {
+	TOACODEC_INPUT("IN A", TOACODEC_IN_A),
+	TOACODEC_INPUT("IN B", TOACODEC_IN_B),
+	TOACODEC_INPUT("IN C", TOACODEC_IN_C),
+	TOACODEC_OUTPUT("OUT", TOACODEC_OUT),
+};
+
+static int g12a_toacodec_component_probe(struct snd_soc_component *c)
+{
+	/* Initialize the static clock parameters */
+	return snd_soc_component_write(c, TOACODEC_CTRL0,
+				       CTRL0_BLK_CAP_INV);
+}
+
+static const struct snd_soc_dapm_route g12a_toacodec_routes[] = {
+	{ "SRC", "I2S A", "IN A Playback" },
+	{ "SRC", "I2S B", "IN B Playback" },
+	{ "SRC", "I2S C", "IN C Playback" },
+	{ "OUT EN", "Switch", "SRC" },
+	{ "OUT Capture", NULL, "OUT EN" },
+};
+
+static const struct snd_kcontrol_new g12a_toacodec_controls[] = {
+	SOC_SINGLE("Lane Select", TOACODEC_CTRL0, CTRL0_LANE_SEL, 3, 0),
+};
+
+static const struct snd_soc_component_driver g12a_toacodec_component_drv = {
+	.probe			= g12a_toacodec_component_probe,
+	.controls		= g12a_toacodec_controls,
+	.num_controls		= ARRAY_SIZE(g12a_toacodec_controls),
+	.dapm_widgets		= g12a_toacodec_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(g12a_toacodec_widgets),
+	.dapm_routes		= g12a_toacodec_routes,
+	.num_dapm_routes	= ARRAY_SIZE(g12a_toacodec_routes),
+	.endianness		= 1,
+	.non_legacy_dai_naming	= 1,
+};
+
+static const struct regmap_config g12a_toacodec_regmap_cfg = {
+	.reg_bits	= 32,
+	.val_bits	= 32,
+	.reg_stride	= 4,
+};
+
+static const struct of_device_id g12a_toacodec_of_match[] = {
+	{ .compatible = "amlogic,g12a-toacodec", },
+	{}
+};
+MODULE_DEVICE_TABLE(of, g12a_toacodec_of_match);
+
+static int g12a_toacodec_probe(struct platform_device *pdev)
+{
+	struct device *dev = &pdev->dev;
+	void __iomem *regs;
+	struct regmap *map;
+	int ret;
+
+	ret = device_reset(dev);
+	if (ret)
+		return ret;
+
+	regs = devm_platform_ioremap_resource(pdev, 0);
+	if (IS_ERR(regs))
+		return PTR_ERR(regs);
+
+	map = devm_regmap_init_mmio(dev, regs, &g12a_toacodec_regmap_cfg);
+	if (IS_ERR(map)) {
+		dev_err(dev, "failed to init regmap: %ld\n",
+			PTR_ERR(map));
+		return PTR_ERR(map);
+	}
+
+	return devm_snd_soc_register_component(dev,
+			&g12a_toacodec_component_drv, g12a_toacodec_dai_drv,
+			ARRAY_SIZE(g12a_toacodec_dai_drv));
+}
+
+static struct platform_driver g12a_toacodec_pdrv = {
+	.driver = {
+		.name = G12A_TOACODEC_DRV_NAME,
+		.of_match_table = g12a_toacodec_of_match,
+	},
+	.probe = g12a_toacodec_probe,
+};
+module_platform_driver(g12a_toacodec_pdrv);
+
+MODULE_AUTHOR("Jerome Brunet <[e-mail]>");
+MODULE_DESCRIPTION("Amlogic G12a To Internal DAC Codec Driver");
+MODULE_LICENSE("GPL v2");

sound/soc/meson/g12a-tohdmitx.c

···
 #include <sound/soc-dai.h>
 
 #include <dt-bindings/sound/meson-g12a-tohdmitx.h>
+#include "meson-codec-glue.h"
 
 #define G12A_TOHDMITX_DRV_NAME "g12a-tohdmitx"
 
 #define TOHDMITX_CTRL0			0x0
 #define  CTRL0_ENABLE_SHIFT		31
-#define  CTRL0_I2S_DAT_SEL		GENMASK(13, 12)
+#define  CTRL0_I2S_DAT_SEL_SHIFT	12
+#define  CTRL0_I2S_DAT_SEL		(0x3 << CTRL0_I2S_DAT_SEL_SHIFT)
 #define  CTRL0_I2S_LRCLK_SEL		GENMASK(9, 8)
 #define  CTRL0_I2S_BLK_CAP_INV		BIT(7)
 #define  CTRL0_I2S_BCLK_O_INV		BIT(6)
 #define  CTRL0_I2S_BCLK_SEL		GENMASK(5, 4)
 #define  CTRL0_SPDIF_CLK_CAP_INV	BIT(3)
 #define  CTRL0_SPDIF_CLK_O_INV		BIT(2)
-#define  CTRL0_SPDIF_SEL		BIT(1)
+#define  CTRL0_SPDIF_SEL_SHIFT		1
+#define  CTRL0_SPDIF_SEL		(0x1 << CTRL0_SPDIF_SEL_SHIFT)
 #define  CTRL0_SPDIF_CLK_SEL		BIT(0)
-
-struct g12a_tohdmitx_input {
-	struct snd_soc_pcm_stream params;
-	unsigned int fmt;
-};
-
-static struct snd_soc_dapm_widget *
-g12a_tohdmitx_get_input(struct snd_soc_dapm_widget *w)
-{
-	struct snd_soc_dapm_path *p = NULL;
-	struct snd_soc_dapm_widget *in;
-
-	snd_soc_dapm_widget_for_each_source_path(w, p) {
-		if (!p->connect)
-			continue;
-
-		/* Check that we still are in the same component */
-		if (snd_soc_dapm_to_component(w->dapm) !=
-		    snd_soc_dapm_to_component(p->source->dapm))
-			continue;
-
-		if (p->source->id == snd_soc_dapm_dai_in)
-			return p->source;
-
-		in = g12a_tohdmitx_get_input(p->source);
-		if (in)
-			return in;
-	}
-
-	return NULL;
-}
-
-static struct g12a_tohdmitx_input *
-g12a_tohdmitx_get_input_data(struct snd_soc_dapm_widget *w)
-{
-	struct snd_soc_dapm_widget *in =
-		g12a_tohdmitx_get_input(w);
-	struct snd_soc_dai *dai;
-
-	if (WARN_ON(!in))
-		return NULL;
-
-	dai = in->priv;
-
-	return dai->playback_dma_data;
-}
 
 static const char * const g12a_tohdmitx_i2s_mux_texts[] = {
 	"I2S A", "I2S B", "I2S C",
 };
 
-static SOC_ENUM_SINGLE_EXT_DECL(g12a_tohdmitx_i2s_mux_enum,
-				g12a_tohdmitx_i2s_mux_texts);
-
-static int g12a_tohdmitx_get_input_val(struct snd_soc_component *component,
-				       unsigned int mask)
-{
-	unsigned int val;
-
-	snd_soc_component_read(component, TOHDMITX_CTRL0, &val);
-	return (val & mask) >> __ffs(mask);
-}
-
-static int g12a_tohdmitx_i2s_mux_get_enum(struct snd_kcontrol *kcontrol,
-					  struct snd_ctl_elem_value *ucontrol)
-{
-	struct snd_soc_component *component =
-		snd_soc_dapm_kcontrol_component(kcontrol);
-
-	ucontrol->value.enumerated.item[0] =
-		g12a_tohdmitx_get_input_val(component, CTRL0_I2S_DAT_SEL);
-
-	return 0;
-}
-
 static int g12a_tohdmitx_i2s_mux_put_enum(struct snd_kcontrol *kcontrol,
-					  struct snd_ctl_elem_value *ucontrol)
+				   struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_component *component =
 		snd_soc_dapm_kcontrol_component(kcontrol);
 	struct snd_soc_dapm_context *dapm =
 		snd_soc_dapm_kcontrol_dapm(kcontrol);
 	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
-	unsigned int mux = ucontrol->value.enumerated.item[0];
-	unsigned int val = g12a_tohdmitx_get_input_val(component,
-						       CTRL0_I2S_DAT_SEL);
+	unsigned int mux, changed;
+
+	mux = snd_soc_enum_item_to_val(e, ucontrol->value.enumerated.item[0]);
+	changed = snd_soc_component_test_bits(component, e->reg,
+					      CTRL0_I2S_DAT_SEL,
+					      FIELD_PREP(CTRL0_I2S_DAT_SEL,
+							 mux));
+
+	if (!changed)
+		return 0;
 
 	/* Force disconnect of the mux while updating */
-	if (val != mux)
-		snd_soc_dapm_mux_update_power(dapm, kcontrol, 0, NULL, NULL);
+	snd_soc_dapm_mux_update_power(dapm, kcontrol, 0, NULL, NULL);
 
-	snd_soc_component_update_bits(component, TOHDMITX_CTRL0,
+	snd_soc_component_update_bits(component, e->reg,
 				      CTRL0_I2S_DAT_SEL |
 				      CTRL0_I2S_LRCLK_SEL |
 				      CTRL0_I2S_BCLK_SEL,
···
 	return 0;
 }
 
+static SOC_ENUM_SINGLE_DECL(g12a_tohdmitx_i2s_mux_enum, TOHDMITX_CTRL0,
+			    CTRL0_I2S_DAT_SEL_SHIFT,
+			    g12a_tohdmitx_i2s_mux_texts);
+
 static const struct snd_kcontrol_new g12a_tohdmitx_i2s_mux =
 	SOC_DAPM_ENUM_EXT("I2S Source", g12a_tohdmitx_i2s_mux_enum,
-			  g12a_tohdmitx_i2s_mux_get_enum,
+			  snd_soc_dapm_get_enum_double,
 			  g12a_tohdmitx_i2s_mux_put_enum);
 
 static const char * const g12a_tohdmitx_spdif_mux_texts[] = {
 	"SPDIF A", "SPDIF B",
 };
-
-static SOC_ENUM_SINGLE_EXT_DECL(g12a_tohdmitx_spdif_mux_enum,
-				g12a_tohdmitx_spdif_mux_texts);
-
-static int g12a_tohdmitx_spdif_mux_get_enum(struct snd_kcontrol *kcontrol,
-					    struct snd_ctl_elem_value *ucontrol)
-{
-	struct snd_soc_component *component =
-		snd_soc_dapm_kcontrol_component(kcontrol);
-
-	ucontrol->value.enumerated.item[0] =
-		g12a_tohdmitx_get_input_val(component, CTRL0_SPDIF_SEL);
-
-	return 0;
-}
 
 static int g12a_tohdmitx_spdif_mux_put_enum(struct snd_kcontrol *kcontrol,
 					    struct snd_ctl_elem_value *ucontrol)
···
 	struct snd_soc_dapm_context *dapm =
 		snd_soc_dapm_kcontrol_dapm(kcontrol);
 	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
-	unsigned int mux = ucontrol->value.enumerated.item[0];
-	unsigned int val = g12a_tohdmitx_get_input_val(component,
-						       CTRL0_SPDIF_SEL);
+	unsigned int mux, changed;
+
+	mux = snd_soc_enum_item_to_val(e, ucontrol->value.enumerated.item[0]);
+	changed = snd_soc_component_test_bits(component, TOHDMITX_CTRL0,
+					      CTRL0_SPDIF_SEL,
+					      FIELD_PREP(CTRL0_SPDIF_SEL, mux));
+
+	if (!changed)
+		return 0;
 
 	/* Force disconnect of the mux while updating */
-	if (val != mux)
-		snd_soc_dapm_mux_update_power(dapm, kcontrol, 0, NULL, NULL);
+	snd_soc_dapm_mux_update_power(dapm, kcontrol, 0, NULL, NULL);
 
 	snd_soc_component_update_bits(component, TOHDMITX_CTRL0,
 				      CTRL0_SPDIF_SEL |
···
 	return 0;
 }
 
+static SOC_ENUM_SINGLE_DECL(g12a_tohdmitx_spdif_mux_enum, TOHDMITX_CTRL0,
+			    CTRL0_SPDIF_SEL_SHIFT,
+			    g12a_tohdmitx_spdif_mux_texts);
+
 static const struct snd_kcontrol_new g12a_tohdmitx_spdif_mux =
 	SOC_DAPM_ENUM_EXT("SPDIF Source", g12a_tohdmitx_spdif_mux_enum,
-			  g12a_tohdmitx_spdif_mux_get_enum,
+			  snd_soc_dapm_get_enum_double,
 			  g12a_tohdmitx_spdif_mux_put_enum);
 
 static const struct snd_kcontrol_new g12a_tohdmitx_out_enable =
···
 			    &g12a_tohdmitx_out_enable),
 };
 
-static int g12a_tohdmitx_input_probe(struct snd_soc_dai *dai)
-{
-	struct g12a_tohdmitx_input *data;
-
-	data = kzalloc(sizeof(*data), GFP_KERNEL);
-	if (!data)
-		return -ENOMEM;
-
-	dai->playback_dma_data = data;
-	return 0;
-}
-
-static int g12a_tohdmitx_input_remove(struct snd_soc_dai *dai)
-{
-	kfree(dai->playback_dma_data);
-	return 0;
-}
-
-static int g12a_tohdmitx_input_hw_params(struct snd_pcm_substream *substream,
-					 struct snd_pcm_hw_params *params,
-					 struct snd_soc_dai *dai)
-{
-	struct g12a_tohdmitx_input *data = dai->playback_dma_data;
-
-	data->params.rates = snd_pcm_rate_to_rate_bit(params_rate(params));
-	data->params.rate_min = params_rate(params);
-	data->params.rate_max = params_rate(params);
-	data->params.formats = 1 << params_format(params);
-	data->params.channels_min = params_channels(params);
-	data->params.channels_max = params_channels(params);
-	data->params.sig_bits = dai->driver->playback.sig_bits;
-
-	return 0;
-}
-
-
-static int g12a_tohdmitx_input_set_fmt(struct snd_soc_dai *dai,
-				       unsigned int fmt)
-{
-	struct g12a_tohdmitx_input *data = dai->playback_dma_data;
-
-	/* Save the source stream format for the downstream link */
-	data->fmt = fmt;
-	return 0;
-}
-
-static int g12a_tohdmitx_output_startup(struct snd_pcm_substream *substream,
-					struct snd_soc_dai *dai)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct g12a_tohdmitx_input *in_data =
-		g12a_tohdmitx_get_input_data(dai->capture_widget);
-
-	if (!in_data)
-		return -ENODEV;
-
-	if (WARN_ON(!rtd->dai_link->params)) {
-		dev_warn(dai->dev, "codec2codec link expected\n");
-		return -EINVAL;
-	}
-
-	/* Replace link params with the input params */
-	rtd->dai_link->params = &in_data->params;
-
-	if (!in_data->fmt)
-		return 0;
-
-	return snd_soc_runtime_set_dai_fmt(rtd, in_data->fmt);
-}
-
 static const struct snd_soc_dai_ops g12a_tohdmitx_input_ops = {
-	.hw_params	= g12a_tohdmitx_input_hw_params,
-	.set_fmt	= g12a_tohdmitx_input_set_fmt,
+	.hw_params	= meson_codec_glue_input_hw_params,
+	.set_fmt	= meson_codec_glue_input_set_fmt,
 };
 
 static const struct snd_soc_dai_ops g12a_tohdmitx_output_ops = {
-	.startup	= g12a_tohdmitx_output_startup,
+	.startup	= meson_codec_glue_output_startup,
 };
 
 #define TOHDMITX_SPDIF_FORMATS					\
···
 	.id = (xid),							\
 	.playback = TOHDMITX_STREAM(xname, "Playback", xfmt, xchmax),	\
 	.ops = &g12a_tohdmitx_input_ops,				\
-	.probe = g12a_tohdmitx_input_probe,				\
-	.remove = g12a_tohdmitx_input_remove,				\
+	.probe = meson_codec_glue_input_dai_probe,			\
+	.remove = meson_codec_glue_input_dai_remove,			\
 }
 
 #define TOHDMITX_OUT(xname, xid, xfmt, xchmax) {			\

sound/soc/meson/gx-card.c

···
+// SPDX-License-Identifier: (GPL-2.0 OR MIT)
+//
+// Copyright (c) 2020 BayLibre, SAS.
+// Author: Jerome Brunet <[e-mail]>
+
+#include <linux/module.h>
+#include <linux/of_platform.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+
+#include "meson-card.h"
+
+struct gx_dai_link_i2s_data {
+	unsigned int mclk_fs;
+};
+
+/*
+ * Base params for the codec to codec links
+ * Those will be over-written by the CPU side of the link
+ */
+static const struct snd_soc_pcm_stream codec_params = {
+	.formats = SNDRV_PCM_FMTBIT_S24_LE,
+	.rate_min = 5525,
+	.rate_max = 192000,
+	.channels_min = 1,
+	.channels_max = 8,
+};
+
+static int gx_card_i2s_be_hw_params(struct snd_pcm_substream *substream,
+				    struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct meson_card *priv = snd_soc_card_get_drvdata(rtd->card);
+	struct gx_dai_link_i2s_data *be =
+		(struct gx_dai_link_i2s_data *)priv->link_data[rtd->num];
+
+	return meson_card_i2s_set_sysclk(substream, params, be->mclk_fs);
+}
+
+static const struct snd_soc_ops gx_card_i2s_be_ops = {
+	.hw_params = gx_card_i2s_be_hw_params,
+};
+
+static int gx_card_parse_i2s(struct snd_soc_card *card,
+			     struct device_node *node,
+			     int *index)
+{
+	struct meson_card *priv = snd_soc_card_get_drvdata(card);
+	struct snd_soc_dai_link *link = &card->dai_link[*index];
+	struct gx_dai_link_i2s_data *be;
+
+	/* Allocate i2s link parameters */
+	be = devm_kzalloc(card->dev, sizeof(*be), GFP_KERNEL);
+	if (!be)
+		return -ENOMEM;
+	priv->link_data[*index] = be;
+
+	/* Setup i2s link */
+	link->ops = &gx_card_i2s_be_ops;
+	link->dai_fmt = meson_card_parse_daifmt(node, link->cpus->of_node);
+
+	of_property_read_u32(node, "mclk-fs", &be->mclk_fs);
+
+	return 0;
+}
+
+static int gx_card_cpu_identify(struct snd_soc_dai_link_component *c,
+				char *match)
+{
+	if (of_device_is_compatible(c->of_node, DT_PREFIX "aiu")) {
+		if (strstr(c->dai_name, match))
+			return 1;
+	}
+
+	/* dai not matched */
+	return 0;
+}
+
+static int gx_card_add_link(struct snd_soc_card *card, struct device_node *np,
+			    int *index)
+{
+	struct snd_soc_dai_link *dai_link = &card->dai_link[*index];
+	struct snd_soc_dai_link_component *cpu;
+	int ret;
+
+	cpu = devm_kzalloc(card->dev, sizeof(*cpu), GFP_KERNEL);
+	if (!cpu)
+		return -ENOMEM;
+
+	dai_link->cpus = cpu;
+	dai_link->num_cpus = 1;
+
+	ret = meson_card_parse_dai(card, np, &dai_link->cpus->of_node,
+				   &dai_link->cpus->dai_name);
+	if (ret)
+		return ret;
+
+	if (gx_card_cpu_identify(dai_link->cpus, "FIFO"))
+		ret = meson_card_set_fe_link(card, dai_link, np, true);
+	else
+		ret = meson_card_set_be_link(card, dai_link, np);
+
+	if (ret)
+		return ret;
+
+	/* Check if the cpu is the i2s encoder and parse i2s data */
+	if (gx_card_cpu_identify(dai_link->cpus, "I2S Encoder"))
+		ret = gx_card_parse_i2s(card, np, index);
+
+	/* Or apply codec to codec params if necessary */
+	else if (gx_card_cpu_identify(dai_link->cpus, "CODEC CTRL"))
+		dai_link->params = &codec_params;
+
+	return ret;
+}
+
+static const struct meson_card_match_data gx_card_match_data = {
+	.add_link = gx_card_add_link,
+};
+
+static const struct of_device_id gx_card_of_match[] = {
+	{
+		.compatible = "amlogic,gx-sound-card",
+		.data = &gx_card_match_data,
+	}, {}
+};
+MODULE_DEVICE_TABLE(of, gx_card_of_match);
+
+static struct platform_driver gx_card_pdrv = {
+	.probe = meson_card_probe,
+	.remove = meson_card_remove,
+	.driver = {
+		.name = "gx-sound-card",
+		.of_match_table = gx_card_of_match,
+	},
+};
+module_platform_driver(gx_card_pdrv);
+
+MODULE_DESCRIPTION("Amlogic GX ALSA machine driver");
+MODULE_AUTHOR("Jerome Brunet <[e-mail]>");
+MODULE_LICENSE("GPL v2");

sound/soc/meson/meson-card-utils.c

···
+// SPDX-License-Identifier: GPL-2.0
+//
+// Copyright (c) 2020 BayLibre, SAS.
+// Author: Jerome Brunet <[e-mail]>
+
+#include <linux/module.h>
+#include <linux/of_platform.h>
+#include <sound/soc.h>
+
+#include "meson-card.h"
+
+int meson_card_i2s_set_sysclk(struct snd_pcm_substream *substream,
+			      struct snd_pcm_hw_params *params,
+			      unsigned int mclk_fs)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai;
+	unsigned int mclk;
+	int ret, i;
+
+	if (!mclk_fs)
+		return 0;
+
+	mclk = params_rate(params) * mclk_fs;
+
+	for_each_rtd_codec_dais(rtd, i, codec_dai) {
+		ret = snd_soc_dai_set_sysclk(codec_dai, 0, mclk,
+					     SND_SOC_CLOCK_IN);
+		if (ret && ret != -ENOTSUPP)
+			return ret;
+	}
+
+	ret = snd_soc_dai_set_sysclk(asoc_rtd_to_cpu(rtd, 0), 0, mclk,
+				     SND_SOC_CLOCK_OUT);
+	if (ret && ret != -ENOTSUPP)
+		return ret;
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(meson_card_i2s_set_sysclk);
+
+int meson_card_reallocate_links(struct snd_soc_card *card,
+				unsigned int num_links)
+{
+	struct meson_card *priv = snd_soc_card_get_drvdata(card);
+	struct snd_soc_dai_link *links;
+	void **ldata;
+
+	links = krealloc(priv->card.dai_link,
+			 num_links * sizeof(*priv->card.dai_link),
+			 GFP_KERNEL | __GFP_ZERO);
+	ldata = krealloc(priv->link_data,
+			 num_links * sizeof(*priv->link_data),
+			 GFP_KERNEL | __GFP_ZERO);
+
+	if (!links || !ldata) {
+		dev_err(priv->card.dev, "failed to allocate links\n");
+		return -ENOMEM;
+	}
+
+	priv->card.dai_link = links;
+	priv->link_data = ldata;
+	priv->card.num_links = num_links;
+	return 0;
+}
+EXPORT_SYMBOL_GPL(meson_card_reallocate_links);
+
+int meson_card_parse_dai(struct snd_soc_card *card,
+			 struct device_node *node,
+			 struct device_node **dai_of_node,
+			 const char **dai_name)
+{
+	struct of_phandle_args args;
+	int ret;
+
+	if (!dai_name || !dai_of_node || !node)
+		return -EINVAL;
+
+	ret = of_parse_phandle_with_args(node, "sound-dai",
+					 "#sound-dai-cells", 0, &args);
+	if (ret) {
+		if (ret != -EPROBE_DEFER)
+			dev_err(card->dev, "can't parse dai %d\n", ret);
+		return ret;
+	}
+	*dai_of_node = args.np;
+
+	return snd_soc_get_dai_name(&args, dai_name);
+}
+EXPORT_SYMBOL_GPL(meson_card_parse_dai);
+
+static int meson_card_set_link_name(struct snd_soc_card *card,
+				    struct snd_soc_dai_link *link,
+				    struct device_node *node,
+				    const char *prefix)
+{
+	char *name = devm_kasprintf(card->dev, GFP_KERNEL, "%s.%s",
+				    prefix, node->full_name);
+	if (!name)
+		return -ENOMEM;
+
+	link->name = name;
+	link->stream_name = name;
+
+	return 0;
+}
+
+unsigned int meson_card_parse_daifmt(struct device_node *node,
+				     struct device_node *cpu_node)
+{
+	struct device_node *bitclkmaster = NULL;
+	struct device_node *framemaster = NULL;
+	unsigned int daifmt;
+
+	daifmt = snd_soc_of_parse_daifmt(node, DT_PREFIX,
+					 &bitclkmaster, &framemaster);
+	daifmt &= ~SND_SOC_DAIFMT_MASTER_MASK;
+
+	/* If no master is provided, default to cpu master */
+	if (!bitclkmaster || bitclkmaster == cpu_node) {
+		daifmt |= (!framemaster || framemaster == cpu_node) ?
+			SND_SOC_DAIFMT_CBS_CFS : SND_SOC_DAIFMT_CBS_CFM;
+	} else {
+		daifmt |= (!framemaster || framemaster == cpu_node) ?
+			SND_SOC_DAIFMT_CBM_CFS : SND_SOC_DAIFMT_CBM_CFM;
+	}
+
+	of_node_put(bitclkmaster);
+	of_node_put(framemaster);
+
+	return daifmt;
+}
+EXPORT_SYMBOL_GPL(meson_card_parse_daifmt);
+
+int meson_card_set_be_link(struct snd_soc_card *card,
+			   struct snd_soc_dai_link *link,
+			   struct device_node *node)
+{
+	struct snd_soc_dai_link_component *codec;
+	struct device_node *np;
+	int ret, num_codecs;
+
+	link->no_pcm = 1;
+	link->dpcm_playback = 1;
+	link->dpcm_capture = 1;
+
+	num_codecs = of_get_child_count(node);
+	if (!num_codecs) {
+		dev_err(card->dev, "be link %s has no codec\n",
+			node->full_name);
+		return -EINVAL;
+	}
+
+	codec = devm_kcalloc(card->dev, num_codecs, sizeof(*codec), GFP_KERNEL);
+	if (!codec)
+		return -ENOMEM;
+
+	link->codecs = codec;
+	link->num_codecs = num_codecs;
+
+	for_each_child_of_node(node, np) {
+		ret = meson_card_parse_dai(card, np, &codec->of_node,
+					   &codec->dai_name);
+		if (ret) {
+			of_node_put(np);
+			return ret;
+		}
+
+		codec++;
+	}
+
+	ret = meson_card_set_link_name(card, link, node, "be");
+	if (ret)
+		dev_err(card->dev, "error setting %pOFn link name\n", np);
+
+	return ret;
+}
+EXPORT_SYMBOL_GPL(meson_card_set_be_link);
+
+int meson_card_set_fe_link(struct snd_soc_card *card,
+			   struct snd_soc_dai_link *link,
+			   struct device_node *node,
+			   bool is_playback)
+{
+	struct snd_soc_dai_link_component *codec;
+
+	codec = devm_kzalloc(card->dev, sizeof(*codec), GFP_KERNEL);
+	if (!codec)
+		return -ENOMEM;
+
+	link->codecs = codec;
+	link->num_codecs = 1;
+
+	link->dynamic = 1;
+	link->dpcm_merged_format = 1;
+	link->dpcm_merged_chan = 1;
+	link->dpcm_merged_rate = 1;
+	link->codecs->dai_name = "snd-soc-dummy-dai";
+	link->codecs->name = "snd-soc-dummy";
+
+	if (is_playback)
+		link->dpcm_playback = 1;
+	else
+		link->dpcm_capture = 1;
+
+	return meson_card_set_link_name(card, link, node, "fe");
+}
+EXPORT_SYMBOL_GPL(meson_card_set_fe_link);
+
+static int meson_card_add_links(struct snd_soc_card *card)
+{
+	struct meson_card *priv = snd_soc_card_get_drvdata(card);
+	struct device_node *node = card->dev->of_node;
+	struct device_node *np;
+	int num, i, ret;
+
+	num = of_get_child_count(node);
+	if (!num) {
+		dev_err(card->dev, "card has no links\n");
+		return -EINVAL;
+	}
+
+	ret = meson_card_reallocate_links(card, num);
+	if (ret)
+		return ret;
+
+	i = 0;
+	for_each_child_of_node(node, np) {
+		ret = priv->match_data->add_link(card, np, &i);
+		if (ret) {
+			of_node_put(np);
+			return ret;
+		}
+
+		i++;
+	}
+
+	return 0;
+}
+
+static int meson_card_parse_of_optional(struct snd_soc_card *card,
+					const char *propname,
+					int (*func)(struct snd_soc_card *c,
+						    const char *p))
+{
+	/* If property is not provided, don't fail ... */
+	if (!of_property_read_bool(card->dev->of_node, propname))
+		return 0;
+
+	/* ... but do fail if it is provided and the parsing fails */
+	return func(card, propname);
+}
+
+static int meson_card_add_aux_devices(struct snd_soc_card *card)
+{
+	struct device_node *node = card->dev->of_node;
+	struct snd_soc_aux_dev *aux;
+	int num, i;
+
+	num = of_count_phandle_with_args(node, "audio-aux-devs", NULL);
+	if (num == -ENOENT) {
+		return 0;
+	} else if (num < 0) {
+		dev_err(card->dev, "error getting auxiliary devices: %d\n",
+			num);
+		return num;
+	}
+
+	aux = devm_kcalloc(card->dev, num, sizeof(*aux), GFP_KERNEL);
+	if (!aux)
+		return -ENOMEM;
+	card->aux_dev = aux;
+	card->num_aux_devs = num;
+
+	for_each_card_pre_auxs(card, i, aux) {
+		aux->dlc.of_node =
+			of_parse_phandle(node, "audio-aux-devs", i);
+		if (!aux->dlc.of_node)
+			return -EINVAL;
+	}
+
+	return 0;
+}
+
+static void meson_card_clean_references(struct meson_card *priv)
+{
+	struct snd_soc_card *card = &priv->card;
+	struct snd_soc_dai_link *link;
+	struct snd_soc_dai_link_component *codec;
+	struct snd_soc_aux_dev *aux;
+	int i, j;
+
+	if (card->dai_link) {
+		for_each_card_prelinks(card, i, link) {
+			if (link->cpus)
+				of_node_put(link->cpus->of_node);
+			for_each_link_codecs(link, j, codec)
+				of_node_put(codec->of_node);
+		}
+	}
+
+	if (card->aux_dev) {
+		for_each_card_pre_auxs(card, i, aux)
+			of_node_put(aux->dlc.of_node);
+	}
+
+	kfree(card->dai_link);
+	kfree(priv->link_data);
+}
+
+int meson_card_probe(struct platform_device *pdev)
+{
+	const struct meson_card_match_data *data;
+	struct device *dev = &pdev->dev;
+	struct meson_card *priv;
+	int ret;
+
+	data = of_device_get_match_data(dev);
+	if (!data) {
+		dev_err(dev, "failed to match device\n");
+		return -ENODEV;
+	}
+
+	priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
+	if (!priv)
+		return -ENOMEM;
+
+	platform_set_drvdata(pdev, priv);
+	snd_soc_card_set_drvdata(&priv->card, priv);
+
+	priv->card.owner = THIS_MODULE;
+	priv->card.dev = dev;
+	priv->match_data = data;
+
+	ret = snd_soc_of_parse_card_name(&priv->card, "model");
+	if (ret < 0)
+		return ret;
+
+	ret = meson_card_parse_of_optional(&priv->card, "audio-routing",
+					   snd_soc_of_parse_audio_routing);
+	if (ret) {
+		dev_err(dev, "error while parsing routing\n");
+		return ret;
+	}
+
+	ret = meson_card_parse_of_optional(&priv->card, "audio-widgets",
+					   snd_soc_of_parse_audio_simple_widgets);
+	if (ret) {
+		dev_err(dev, "error while parsing widgets\n");
+		return ret;
+	}
+
+	ret = meson_card_add_links(&priv->card);
+	if (ret)
+		goto out_err;
+
+	ret = meson_card_add_aux_devices(&priv->card);
+	if (ret)
+		goto out_err;
+
+	ret = devm_snd_soc_register_card(dev, &priv->card);
+	if (ret)
+		goto out_err;
+
+	return 0;
+
+out_err:
+	meson_card_clean_references(priv);
+	return ret;
+}
+EXPORT_SYMBOL_GPL(meson_card_probe);
+
+int meson_card_remove(struct platform_device *pdev)
+{
+	struct meson_card *priv = platform_get_drvdata(pdev);
+
+	meson_card_clean_references(priv);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(meson_card_remove);
+
+MODULE_DESCRIPTION("Amlogic Sound Card Utils");
+MODULE_AUTHOR("Jerome Brunet <[e-mail]>");
+MODULE_LICENSE("GPL v2");

sound/soc/meson/meson-card.h

···
+/* SPDX-License-Identifier: GPL-2.0 */
+/*
+ * Copyright (c) 2020 BayLibre, SAS.
+ * Author: Jerome Brunet <[e-mail]>
+ */
+
+#ifndef _MESON_SND_CARD_H
+#define _MESON_SND_CARD_H
+
+struct device_node;
+struct platform_device;
+
+struct snd_soc_card;
+struct snd_pcm_substream;
+struct snd_pcm_hw_params;
+
+#define DT_PREFIX "amlogic,"
+
+struct meson_card_match_data {
+	int (*add_link)(struct snd_soc_card *card,
+			struct device_node *node,
+			int *index);
+};
+
+struct meson_card {
+	const struct meson_card_match_data *match_data;
+	struct snd_soc_card card;
+	void **link_data;
+};
+
+unsigned int meson_card_parse_daifmt(struct device_node *node,
+				     struct device_node *cpu_node);
+
+int meson_card_i2s_set_sysclk(struct snd_pcm_substream *substream,
+			      struct snd_pcm_hw_params *params,
+			      unsigned int mclk_fs);
+
+int meson_card_reallocate_links(struct snd_soc_card *card,
+				unsigned int num_links);
+int meson_card_parse_dai(struct snd_soc_card *card,
+			 struct device_node *node,
+			 struct device_node **dai_of_node,
+			 const char **dai_name);
+int meson_card_set_be_link(struct snd_soc_card *card,
+			   struct snd_soc_dai_link *link,
+			   struct device_node *node);
+int meson_card_set_fe_link(struct snd_soc_card *card,
+			   struct snd_soc_dai_link *link,
+			   struct device_node *node,
+			   bool is_playback);
+
+int meson_card_probe(struct platform_device *pdev);
+int meson_card_remove(struct platform_device *pdev);
+
+#endif /* _MESON_SND_CARD_H */

sound/soc/meson/meson-codec-glue.c

···
+// SPDX-License-Identifier: GPL-2.0
+//
+// Copyright (c) 2019 BayLibre, SAS.
+// Author: Jerome Brunet <[e-mail]>
+
+#include <linux/module.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+
+#include "meson-codec-glue.h"
+
+static struct snd_soc_dapm_widget *
+meson_codec_glue_get_input(struct snd_soc_dapm_widget *w)
+{
+	struct snd_soc_dapm_path *p = NULL;
+	struct snd_soc_dapm_widget *in;
+
+	snd_soc_dapm_widget_for_each_source_path(w, p) {
+		if (!p->connect)
+			continue;
+
+		/* Check that we still are in the same component */
+		if (snd_soc_dapm_to_component(w->dapm) !=
+		    snd_soc_dapm_to_component(p->source->dapm))
+			continue;
+
+		if (p->source->id == snd_soc_dapm_dai_in)
+			return p->source;
+
+		in = meson_codec_glue_get_input(p->source);
+		if (in)
+			return in;
+	}
+
+	return NULL;
+}
+
+static void meson_codec_glue_input_set_data(struct snd_soc_dai *dai,
+					    struct meson_codec_glue_input *data)
+{
+	dai->playback_dma_data = data;
+}
+
+struct meson_codec_glue_input *
+meson_codec_glue_input_get_data(struct snd_soc_dai *dai)
+{
+	return dai->playback_dma_data;
+}
+EXPORT_SYMBOL_GPL(meson_codec_glue_input_get_data);
+
+static struct meson_codec_glue_input *
+meson_codec_glue_output_get_input_data(struct snd_soc_dapm_widget *w)
+{
+	struct snd_soc_dapm_widget *in =
+		meson_codec_glue_get_input(w);
+	struct snd_soc_dai *dai;
+
+	if (WARN_ON(!in))
+		return NULL;
+
+	dai = in->priv;
+
+	return meson_codec_glue_input_get_data(dai);
+}
+
+int meson_codec_glue_input_hw_params(struct snd_pcm_substream *substream,
+				     struct snd_pcm_hw_params *params,
+				     struct snd_soc_dai *dai)
+{
+	struct meson_codec_glue_input *data =
+		meson_codec_glue_input_get_data(dai);
+
+	data->params.rates = snd_pcm_rate_to_rate_bit(params_rate(params));
+	data->params.rate_min = params_rate(params);
+	data->params.rate_max = params_rate(params);
+	data->params.formats = 1ULL << (__force int) params_format(params);
+	data->params.channels_min = params_channels(params);
+	data->params.channels_max = params_channels(params);
+	data->params.sig_bits = dai->driver->playback.sig_bits;
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(meson_codec_glue_input_hw_params);
+
+int meson_codec_glue_input_set_fmt(struct snd_soc_dai *dai,
+				   unsigned int fmt)
+{
+	struct meson_codec_glue_input *data =
+		meson_codec_glue_input_get_data(dai);
+
+	/* Save the source stream format for the downstream link */
+	data->fmt = fmt;
+	return 0;
+}
+EXPORT_SYMBOL_GPL(meson_codec_glue_input_set_fmt);
+
+int meson_codec_glue_output_startup(struct snd_pcm_substream *substream,
+				    struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct meson_codec_glue_input *in_data =
+		meson_codec_glue_output_get_input_data(dai->capture_widget);
+
+	if (!in_data)
+		return -ENODEV;
+
+	if (WARN_ON(!rtd->dai_link->params)) {
+		dev_warn(dai->dev, "codec2codec link expected\n");
+		return -EINVAL;
+	}
+
+	/* Replace link params with the input params */
+	rtd->dai_link->params = &in_data->params;
+
+	if (!in_data->fmt)
+		return 0;
+
+	return snd_soc_runtime_set_dai_fmt(rtd, in_data->fmt);
+}
+EXPORT_SYMBOL_GPL(meson_codec_glue_output_startup);
+
+int meson_codec_glue_input_dai_probe(struct snd_soc_dai *dai)
+{
+	struct meson_codec_glue_input *data;
+
+	data = kzalloc(sizeof(*data), GFP_KERNEL);
+	if (!data)
+		return -ENOMEM;
+
+	meson_codec_glue_input_set_data(dai, data);
+	return 0;
+}
+EXPORT_SYMBOL_GPL(meson_codec_glue_input_dai_probe);
+
+int meson_codec_glue_input_dai_remove(struct snd_soc_dai *dai)
+{
+	struct meson_codec_glue_input *data =
+		meson_codec_glue_input_get_data(dai);
+
+	kfree(data);
+	return 0;
+}
+EXPORT_SYMBOL_GPL(meson_codec_glue_input_dai_remove);
+
+MODULE_AUTHOR("Jerome Brunet <[e-mail]>");
+MODULE_DESCRIPTION("Amlogic Codec Glue Helpers");
+MODULE_LICENSE("GPL v2");
+

sound/soc/meson/meson-codec-glue.h

···
+/* SPDX-License-Identifier: GPL-2.0
+ *
+ * Copyright (c) 2018 Baylibre SAS.
+ * Author: Jerome Brunet <[e-mail]>
+ */
+
+#ifndef _MESON_CODEC_GLUE_H
+#define _MESON_CODEC_GLUE_H
+
+#include <sound/soc.h>
+
+struct meson_codec_glue_input {
+	struct snd_soc_pcm_stream params;
+	unsigned int fmt;
+};
+
+/* Input helpers */
+struct meson_codec_glue_input *
+meson_codec_glue_input_get_data(struct snd_soc_dai *dai);
+int meson_codec_glue_input_hw_params(struct snd_pcm_substream *substream,
+				     struct snd_pcm_hw_params *params,
+				     struct snd_soc_dai *dai);
+int meson_codec_glue_input_set_fmt(struct snd_soc_dai *dai,
+				   unsigned int fmt);
+int meson_codec_glue_input_dai_probe(struct snd_soc_dai *dai);
+int meson_codec_glue_input_dai_remove(struct snd_soc_dai *dai);
+
+/* Output helpers */
+int meson_codec_glue_output_startup(struct snd_pcm_substream *substream,
+				    struct snd_soc_dai *dai);
+
+#endif /* _MESON_CODEC_GLUE_H */

sound/soc/meson/t9015.c

···
+// SPDX-License-Identifier: GPL-2.0
+//
+// Copyright (c) 2020 BayLibre, SAS.
+// Author: Jerome Brunet <[e-mail]>
+
+#include <linux/clk.h>
+#include <linux/delay.h>
+#include <linux/module.h>
+#include <linux/regmap.h>
+#include <linux/regulator/consumer.h>
+#include <linux/reset.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+
+#define BLOCK_EN	0x00
+#define  LORN_EN	0
+#define  LORP_EN	1
+#define  LOLN_EN	2
+#define  LOLP_EN	3
+#define  DACR_EN	4
+#define  DACL_EN	5
+#define  DACR_INV	20
+#define  DACL_INV	21
+#define  DACR_SRC	22
+#define  DACL_SRC	23
+#define  REFP_BUF_EN	BIT(12)
+#define  BIAS_CURRENT_EN BIT(13)
+#define  VMID_GEN_FAST	BIT(14)
+#define  VMID_GEN_EN	BIT(15)
+#define  I2S_MODE	BIT(30)
+#define VOL_CTRL0	0x04
+#define  GAIN_H		31
+#define  GAIN_L		23
+#define VOL_CTRL1	0x08
+#define  DAC_MONO	8
+#define  RAMP_RATE	10
+#define  VC_RAMP_MODE	12
+#define  MUTE_MODE	13
+#define  UNMUTE_MODE	14
+#define  DAC_SOFT_MUTE	15
+#define  DACR_VC	16
+#define  DACL_VC	24
+#define LINEOUT_CFG	0x0c
+#define  LORN_POL	0
+#define  LORP_POL	4
+#define  LOLN_POL	8
+#define  LOLP_POL	12
+#define POWER_CFG	0x10
+
+struct t9015 {
+	struct clk *pclk;
+	struct regulator *avdd;
+};
+
+static int t9015_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+	struct snd_soc_component *component = dai->component;
+	unsigned int val;
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		val = I2S_MODE;
+		break;
+
+	case SND_SOC_DAIFMT_CBS_CFS:
+		val = 0;
+		break;
+
+	default:
+		return -EINVAL;
+	}
+
+	snd_soc_component_update_bits(component, BLOCK_EN, I2S_MODE, val);
+
+	if (((fmt & SND_SOC_DAIFMT_FORMAT_MASK) != SND_SOC_DAIFMT_I2S) &&
+	    ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) != SND_SOC_DAIFMT_LEFT_J))
+		return -EINVAL;
+
+	return 0;
+}
+
+static const struct snd_soc_dai_ops t9015_dai_ops = {
+	.set_fmt = t9015_dai_set_fmt,
+};
+
+static struct snd_soc_dai_driver t9015_dai = {
+	.name = "t9015-hifi",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_8000_96000,
+		.formats = (SNDRV_PCM_FMTBIT_S8 |
+			    SNDRV_PCM_FMTBIT_S16_LE |
+			    SNDRV_PCM_FMTBIT_S20_LE |
+			    SNDRV_PCM_FMTBIT_S24_LE),
+	},
+	.ops = &t9015_dai_ops,
+};
+
+static const DECLARE_TLV_DB_MINMAX_MUTE(dac_vol_tlv, -9525, 0);
+
+static const char * const ramp_rate_txt[] = { "Fast", "Slow" };
+static SOC_ENUM_SINGLE_DECL(ramp_rate_enum, VOL_CTRL1, RAMP_RATE,
+			    ramp_rate_txt);
+
+static const char * const dacr_in_txt[] = { "Right", "Left" };
+static SOC_ENUM_SINGLE_DECL(dacr_in_enum, BLOCK_EN, DACR_SRC, dacr_in_txt);
+
+static const char * const dacl_in_txt[] = { "Left", "Right" };
+static SOC_ENUM_SINGLE_DECL(dacl_in_enum, BLOCK_EN, DACL_SRC, dacl_in_txt);
+
+static const char * const mono_txt[] = { "Stereo", "Mono"};
+static SOC_ENUM_SINGLE_DECL(mono_enum, VOL_CTRL1, DAC_MONO, mono_txt);
+
+static const struct snd_kcontrol_new t9015_snd_controls[] = {
+	/* Volume Controls */
+	SOC_ENUM("Playback Channel Mode", mono_enum),
+	SOC_SINGLE("Playback Switch", VOL_CTRL1, DAC_SOFT_MUTE, 1, 1),
+	SOC_DOUBLE_TLV("Playback Volume", VOL_CTRL1, DACL_VC, DACR_VC,
+		       0xff, 0, dac_vol_tlv),
+
+	/* Ramp Controls */
+	SOC_ENUM("Ramp Rate", ramp_rate_enum),
+	SOC_SINGLE("Volume Ramp Switch", VOL_CTRL1, VC_RAMP_MODE, 1, 0),
+	SOC_SINGLE("Mute Ramp Switch", VOL_CTRL1, MUTE_MODE, 1, 0),
+	SOC_SINGLE("Unmute Ramp Switch", VOL_CTRL1, UNMUTE_MODE, 1, 0),
+};
+
+static const struct snd_kcontrol_new t9015_right_dac_mux =
+	SOC_DAPM_ENUM("Right DAC Source", dacr_in_enum);
+static const struct snd_kcontrol_new t9015_left_dac_mux =
+	SOC_DAPM_ENUM("Left DAC Source", dacl_in_enum);
+
+static const struct snd_soc_dapm_widget t9015_dapm_widgets[] = {
+	SND_SOC_DAPM_AIF_IN("Right IN", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("Left IN", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_MUX("Right DAC Sel", SND_SOC_NOPM, 0, 0,
+			 &t9015_right_dac_mux),
+	SND_SOC_DAPM_MUX("Left DAC Sel", SND_SOC_NOPM, 0, 0,
+			 &t9015_left_dac_mux),
+	SND_SOC_DAPM_DAC("Right DAC", NULL, BLOCK_EN, DACR_EN, 0),
+	SND_SOC_DAPM_DAC("Left DAC",  NULL, BLOCK_EN, DACL_EN, 0),
+	SND_SOC_DAPM_OUT_DRV("Right- Driver", BLOCK_EN, LORN_EN, 0,
+			 NULL, 0),
+	SND_SOC_DAPM_OUT_DRV("Right+ Driver", BLOCK_EN, LORP_EN, 0,
+			 NULL, 0),
+	SND_SOC_DAPM_OUT_DRV("Left- Driver",  BLOCK_EN, LOLN_EN, 0,
+			 NULL, 0),
+	SND_SOC_DAPM_OUT_DRV("Left+ Driver",  BLOCK_EN, LOLP_EN, 0,
+			 NULL, 0),
+	SND_SOC_DAPM_OUTPUT("LORN"),
+	SND_SOC_DAPM_OUTPUT("LORP"),
+	SND_SOC_DAPM_OUTPUT("LOLN"),
+	SND_SOC_DAPM_OUTPUT("LOLP"),
+};
+
+static const struct snd_soc_dapm_route t9015_dapm_routes[] = {
+	{ "Right IN", NULL, "Playback" },
+	{ "Left IN",  NULL, "Playback" },
+	{ "Right DAC Sel", "Right", "Right IN" },
+	{ "Right DAC Sel", "Left",  "Left IN" },
+	{ "Left DAC Sel",  "Right", "Right IN" },
+	{ "Left DAC Sel",  "Left",  "Left IN" },
+	{ "Right DAC", NULL, "Right DAC Sel" },
+	{ "Left DAC",  NULL, "Left DAC Sel" },
+	{ "Right- Driver", NULL, "Right DAC" },
+	{ "Right+ Driver", NULL, "Right DAC" },
+	{ "Left- Driver",  NULL, "Left DAC"  },
+	{ "Left+ Driver",  NULL, "Left DAC"  },
+	{ "LORN", NULL, "Right- Driver", },
+	{ "LORP", NULL, "Right+ Driver", },
+	{ "LOLN", NULL, "Left- Driver",  },
+	{ "LOLP", NULL, "Left+ Driver",  },
+};
+
+static int t9015_set_bias_level(struct snd_soc_component *component,
+				enum snd_soc_bias_level level)
+{
+	struct t9015 *priv = snd_soc_component_get_drvdata(component);
+	enum snd_soc_bias_level now =
+		snd_soc_component_get_bias_level(component);
+	int ret;
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		snd_soc_component_update_bits(component, BLOCK_EN,
+					      BIAS_CURRENT_EN,
+					      BIAS_CURRENT_EN);
+		break;
+	case SND_SOC_BIAS_PREPARE:
+		snd_soc_component_update_bits(component, BLOCK_EN,
+					      BIAS_CURRENT_EN,
+					      0);
+		break;
+	case SND_SOC_BIAS_STANDBY:
+		ret = regulator_enable(priv->avdd);
+		if (ret) {
+			dev_err(component->dev, "AVDD enable failed\n");
+			return ret;
+		}
+
+		if (now == SND_SOC_BIAS_OFF) {
+			snd_soc_component_update_bits(component, BLOCK_EN,
+				VMID_GEN_EN | VMID_GEN_FAST | REFP_BUF_EN,
+				VMID_GEN_EN | VMID_GEN_FAST | REFP_BUF_EN);
+
+			mdelay(200);
+			snd_soc_component_update_bits(component, BLOCK_EN,
+						      VMID_GEN_FAST,
+						      0);
+		}
+
+		break;
+	case SND_SOC_BIAS_OFF:
+		snd_soc_component_update_bits(component, BLOCK_EN,
+			VMID_GEN_EN | VMID_GEN_FAST | REFP_BUF_EN,
+			0);
+
+		regulator_disable(priv->avdd);
+		break;
+	}
+
+	return 0;
+}
+
+static const struct snd_soc_component_driver t9015_codec_driver = {
+	.set_bias_level		= t9015_set_bias_level,
+	.controls		= t9015_snd_controls,
+	.num_controls		= ARRAY_SIZE(t9015_snd_controls),
+	.dapm_widgets		= t9015_dapm_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(t9015_dapm_widgets),
+	.dapm_routes		= t9015_dapm_routes,
+	.num_dapm_routes	= ARRAY_SIZE(t9015_dapm_routes),
+	.suspend_bias_off	= 1,
+	.endianness		= 1,
+	.non_legacy_dai_naming	= 1,
+};
+
+static const struct regmap_config t9015_regmap_config = {
+	.reg_bits		= 32,
+	.reg_stride		= 4,
+	.val_bits		= 32,
+	.max_register		= POWER_CFG,
+};
+
+static int t9015_probe(struct platform_device *pdev)
+{
+	struct device *dev = &pdev->dev;
+	struct t9015 *priv;
+	void __iomem *regs;
+	struct regmap *regmap;
+	int ret;
+
+	priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
+	if (!priv)
+		return -ENOMEM;
+	platform_set_drvdata(pdev, priv);
+
+	priv->pclk = devm_clk_get(dev, "pclk");
+	if (IS_ERR(priv->pclk)) {
+		if (PTR_ERR(priv->pclk) != -EPROBE_DEFER)
+			dev_err(dev, "failed to get core clock\n");
+		return PTR_ERR(priv->pclk);
+	}
+
+	priv->avdd = devm_regulator_get(dev, "AVDD");
+	if (IS_ERR(priv->avdd)) {
+		if (PTR_ERR(priv->avdd) != -EPROBE_DEFER)
+			dev_err(dev, "failed to AVDD\n");
+		return PTR_ERR(priv->avdd);
+	}
+
+	ret = clk_prepare_enable(priv->pclk);
+	if (ret) {
+		dev_err(dev, "core clock enable failed\n");
+		return ret;
+	}
+
+	ret = devm_add_action_or_reset(dev,
+			(void(*)(void *))clk_disable_unprepare,
+			priv->pclk);
+	if (ret)
+		return ret;
+
+	ret = device_reset(dev);
+	if (ret) {
+		dev_err(dev, "reset failed\n");
+		return ret;
+	}
+
+	regs = devm_platform_ioremap_resource(pdev, 0);
+	if (IS_ERR(regs)) {
+		dev_err(dev, "register map failed\n");
+		return PTR_ERR(regs);
+	}
+
+	regmap = devm_regmap_init_mmio(dev, regs, &t9015_regmap_config);
+	if (IS_ERR(regmap)) {
+		dev_err(dev, "regmap init failed\n");
+		return PTR_ERR(regmap);
+	}
+
+	/*
+	 * Initialize output polarity:
+	 * ATM the output polarity is fixed but in the future it might useful
+	 * to add DT property to set this depending on the platform needs
+	 */
+	regmap_write(regmap, LINEOUT_CFG, 0x1111);
+
+	return devm_snd_soc_register_component(dev, &t9015_codec_driver,
+					       &t9015_dai, 1);
+}
+
+static const struct of_device_id t9015_ids[] = {
+	{ .compatible = "amlogic,t9015", },
+	{ }
+};
+MODULE_DEVICE_TABLE(of, t9015_ids);
+
+static struct platform_driver t9015_driver = {
+	.driver = {
+		.name = "t9015-codec",
+		.of_match_table = of_match_ptr(t9015_ids),
+	},
+	.probe = t9015_probe,
+};
+
+module_platform_driver(t9015_driver);
+
+MODULE_DESCRIPTION("ASoC Amlogic T9015 codec driver");
+MODULE_AUTHOR("Jerome Brunet <[e-mail]>");
+MODULE_LICENSE("GPL");

sound/soc/mxs/mxs-sgtl5000.c

···
 	struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	unsigned int rate = params_rate(params);
 	u32 mclk;
 	int ret;

sound/soc/pxa/Kconfig

···
 	depends on SND_PXA2XX_SOC && MACH_TOSA
 	depends on MFD_TC6393XB
 	depends on AC97_BUS=n
+	select REGMAP
+	select AC97_BUS_NEW
+	select AC97_BUS_COMPAT
 	select SND_PXA2XX_SOC_AC97
 	select SND_SOC_WM9712
 	help
···
 	tristate "SoC AC97 Audio support for e740"
 	depends on SND_PXA2XX_SOC && MACH_E740
 	depends on AC97_BUS=n
+	select REGMAP
+	select AC97_BUS_NEW
+	select AC97_BUS_COMPAT
 	select SND_SOC_WM9705
 	select SND_PXA2XX_SOC_AC97
 	help
···
 	tristate "SoC AC97 Audio support for e750"
 	depends on SND_PXA2XX_SOC && MACH_E750
 	depends on AC97_BUS=n
+	select REGMAP
 	select SND_SOC_WM9705
 	select SND_PXA2XX_SOC_AC97
 	help
···
 	tristate "SoC AC97 Audio support for e800"
 	depends on SND_PXA2XX_SOC && MACH_E800
 	depends on AC97_BUS=n
+	select REGMAP
 	select SND_SOC_WM9712
+	select AC97_BUS_NEW
+	select AC97_BUS_COMPAT
 	select SND_PXA2XX_SOC_AC97
 	help
 	  Say Y if you want to add support for SoC audio on the
···
 	depends on SND_PXA2XX_SOC && (MACH_EM_X270 || MACH_EXEDA || \
 			MACH_CM_X300)
 	depends on AC97_BUS=n
+	select REGMAP
+	select AC97_BUS_NEW
+	select AC97_BUS_COMPAT
 	select SND_PXA2XX_SOC_AC97
 	select SND_SOC_WM9712
 	help
···
 	depends on SND_PXA2XX_SOC && (MACH_PALMLD || MACH_PALMTX || \
 			MACH_PALMT5 || MACH_PALMTE2)
 	depends on AC97_BUS=n
+	select REGMAP
+	select AC97_BUS_NEW
+	select AC97_BUS_COMPAT
 	select SND_PXA2XX_SOC_AC97
 	select SND_SOC_WM9712
 	help
···
 	tristate "SoC Audio support for Marvell Zylonite"
 	depends on SND_PXA2XX_SOC && MACH_ZYLONITE
 	depends on AC97_BUS=n
+	select AC97_BUS_NEW
+	select AC97_BUS_COMPAT
 	select SND_PXA2XX_SOC_AC97
+	select REGMAP
 	select SND_PXA_SOC_SSP
 	select SND_SOC_WM9713
 	help
···
 	tristate "SoC Audio support for MIO A701"
 	depends on SND_PXA2XX_SOC && MACH_MIOA701
 	depends on AC97_BUS=n
+	select REGMAP
+	select AC97_BUS_NEW
+	select AC97_BUS_COMPAT
 	select SND_PXA2XX_SOC_AC97
 	select SND_SOC_WM9713
 	help

sound/soc/pxa/brownstone.c

···
 				       struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	int freq_out, sspa_mclk, sysclk;
 
 	if (params_rate(params) > 11025) {

sound/soc/pxa/corgi.c

···
 	struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	unsigned int clk = 0;
 	int ret = 0;
 

sound/soc/pxa/hx4700.c

···
 			    struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	int ret = 0;
 
 	/* set the I2S system clock as output */

sound/soc/pxa/imote2.c

···
 				 struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	unsigned int clk = 0;
 	int ret;
 

sound/soc/pxa/magician.c

···
 				       struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	unsigned int width;
 	int ret = 0;
 
···
 				      struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	int ret = 0;
 
 	/* set codec DAI configuration */
···
 	adapter = i2c_get_adapter(0);
 	if (!adapter)
 		return -ENODEV;
-	client = i2c_new_device(adapter, i2c_board_info);
+	client = i2c_new_client_device(adapter, i2c_board_info);
 	i2c_put_adapter(adapter);
-	if (!client)
-		return -ENODEV;
+	if (IS_ERR(client))
+		return PTR_ERR(client);
 
 	ret = gpio_request(EGPIO_MAGICIAN_SPK_POWER, "SPK_POWER");
 	if (ret)

sound/soc/pxa/mioa701_wm9713.c

···
 	struct snd_soc_component *component;
 
 	rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[0]);
-	component = rtd->codec_dai->component;
+	component = asoc_rtd_to_codec(rtd, 0)->component;
 	return rear_amp_power(component, SND_SOC_DAPM_EVENT_ON(event));
 }
 
···
 
 static int mioa701_wm9713_init(struct snd_soc_pcm_runtime *rtd)
 {
-	struct snd_soc_component *component = rtd->codec_dai->component;
+	struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
 
 	/* Prepare GPIO8 for rear speaker amplifier */
 	snd_soc_component_update_bits(component, AC97_GPIO_CFG, 0x100, 0x100);

sound/soc/pxa/mmp-pcm.c

···
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct platform_device *pdev = to_platform_device(component->dev);
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	struct mmp_dma_data dma_data;
 	struct resource *r;
 

sound/soc/pxa/mmp-sspa.c

···
 			       struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	struct sspa_priv *sspa_priv = snd_soc_dai_get_drvdata(dai);
 	struct ssp_device *sspa = sspa_priv->sspa;
 	struct snd_dmaengine_dai_dma_data *dma_params;

sound/soc/pxa/poodle.c

···
 	struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	unsigned int clk = 0;
 	int ret = 0;
 

sound/soc/pxa/pxa2xx-i2s.c

···
 			      struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 
 	if (IS_ERR(clk_i2s))
 		return PTR_ERR(clk_i2s);

sound/soc/pxa/spitz.c

···
 	struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	unsigned int clk = 0;
 	int ret = 0;
 

sound/soc/pxa/ttc-dkb.c

···
 
 static int ttc_pm860x_init(struct snd_soc_pcm_runtime *rtd)
 {
-	struct snd_soc_component *component = rtd->codec_dai->component;
+	struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
 
 	/* Headset jack detection */
 	snd_soc_card_jack_new(rtd->card, "Headphone Jack", SND_JACK_HEADPHONE |

sound/soc/pxa/z2.c

···
 	struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	unsigned int clk = 0;
 	int ret = 0;
 

sound/soc/pxa/zylonite.c

···
 static int zylonite_wm9713_init(struct snd_soc_pcm_runtime *rtd)
 {
 	if (clk_pout)
-		snd_soc_dai_set_pll(rtd->codec_dai, 0, 0,
+		snd_soc_dai_set_pll(asoc_rtd_to_codec(rtd, 0), 0, 0,
 				    clk_get_rate(pout), 0);
 
 	return 0;
···
 				    struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	unsigned int wm9713_div = 0;
 	int ret = 0;
 	int rate = params_rate(params);

sound/soc/qcom/Kconfig

···
 
 config SND_SOC_SDM845
 	tristate "SoC Machine driver for SDM845 boards"
-	depends on QCOM_APR && CROS_EC && I2C
+	depends on QCOM_APR && CROS_EC && I2C && SOUNDWIRE
 	select SND_SOC_QDSP6
 	select SND_SOC_QCOM_COMMON
 	select SND_SOC_RT5663

sound/soc/qcom/apq8016_sbc.c

···
 
 static int apq8016_sbc_dai_init(struct snd_soc_pcm_runtime *rtd)
 {
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+	struct snd_soc_dai *codec_dai;
 	struct snd_soc_component *component;
-	struct snd_soc_dai_link *dai_link = rtd->dai_link;
 	struct snd_soc_card *card = rtd->card;
 	struct apq8016_sbc_data *pdata = snd_soc_card_get_drvdata(card);
 	int i, rval;
···
 		pdata->jack_setup = true;
 	}
 
-	for (i = 0 ; i < dai_link->num_codecs; i++) {
-		struct snd_soc_dai *dai = rtd->codec_dais[i];
+	for_each_rtd_codec_dais(rtd, i, codec_dai) {
 
-		component = dai->component;
+		component = codec_dai->component;
 		/* Set default mclk for internal codec */
 		rval = snd_soc_component_set_sysclk(component, 0, 0, DEFAULT_MCLK_RATE,
 				       SND_SOC_CLOCK_IN);

sound/soc/qcom/apq8096.c

···
 			     struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	u32 rx_ch[SLIM_MAX_RX_PORTS], tx_ch[SLIM_MAX_TX_PORTS];
 	u32 rx_ch_cnt = 0, tx_ch_cnt = 0;
 	int ret = 0;
···
 
 static int apq8096_init(struct snd_soc_pcm_runtime *rtd)
 {
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 
 	/*
 	 * Codec SLIMBUS configuration

sound/soc/qcom/lpass-platform.c

···
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
-	struct snd_soc_dai *cpu_dai = soc_runtime->cpu_dai;
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(soc_runtime, 0);
 	struct lpass_data *drvdata = snd_soc_component_get_drvdata(component);
 	struct lpass_variant *v = drvdata->variant;
 	int ret, dma_ch, dir = substream->stream;
···
 	struct snd_pcm_substream *substream;
 	int i;
 
-	for (i = 0; i < ARRAY_SIZE(pcm->streams); i++) {
+	for_each_pcm_streams(i) {
 		substream = pcm->streams[i].substream;
 		if (substream) {
 			snd_dma_free_pages(&substream->dma_buffer);

sound/soc/qcom/qdsp6/q6asm-dai.c

···
 #define Q6ASM_DAI_TX	1
 #define Q6ASM_DAI_RX	2
 
+#define ALAC_CH_LAYOUT_MONO   ((101 << 16) | 1)
+#define ALAC_CH_LAYOUT_STEREO ((101 << 16) | 2)
+
 enum stream_state {
 	Q6ASM_STREAM_IDLE = 0,
 	Q6ASM_STREAM_STOPPED,
···
 };
 
 struct q6asm_dai_data {
+	struct snd_soc_dai_driver *dais;
+	int num_dais;
 	long long int sid;
 };
 
···
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		ret = q6asm_open_write(prtd->audio_client, FORMAT_LINEAR_PCM,
-				       prtd->bits_per_sample);
+				       0, prtd->bits_per_sample);
 	} else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
 		ret = q6asm_open_read(prtd->audio_client, FORMAT_LINEAR_PCM,
 				       prtd->bits_per_sample);
···
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
-	struct snd_soc_dai *cpu_dai = soc_prtd->cpu_dai;
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(soc_prtd, 0);
 	struct q6asm_dai_rtd *prtd;
 	struct q6asm_dai_data *pdata;
 	struct device *dev = component->dev;
···
 	struct snd_soc_pcm_runtime *rtd = stream->private_data;
 	struct snd_soc_component *c = snd_soc_rtdcom_lookup(rtd, DRV_NAME);
 	struct snd_compr_runtime *runtime = stream->runtime;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	struct q6asm_dai_data *pdata;
 	struct device *dev = c->dev;
 	struct q6asm_dai_rtd *prtd;
···
 	int dir = stream->direction;
 	struct q6asm_dai_data *pdata;
 	struct q6asm_flac_cfg flac_cfg;
+	struct q6asm_wma_cfg wma_cfg;
+	struct q6asm_alac_cfg alac_cfg;
+	struct q6asm_ape_cfg ape_cfg;
+	unsigned int wma_v9 = 0;
 	struct device *dev = c->dev;
 	int ret;
 	union snd_codec_options *codec_options;
 	struct snd_dec_flac *flac;
+	struct snd_dec_wma *wma;
+	struct snd_dec_alac *alac;
+	struct snd_dec_ape *ape;
 
 	codec_options = &(prtd->codec_param.codec.options);
 
···
 	prtd->bits_per_sample = 16;
 	if (dir == SND_COMPRESS_PLAYBACK) {
 		ret = q6asm_open_write(prtd->audio_client, params->codec.id,
-					prtd->bits_per_sample);
+				params->codec.profile, prtd->bits_per_sample);
 
 		if (ret < 0) {
 			dev_err(dev, "q6asm_open_write failed\n");
···
 			return -EIO;
 		}
 		break;
+
+	case SND_AUDIOCODEC_WMA:
+		wma = &codec_options->wma_d;
+
+		memset(&wma_cfg, 0x0, sizeof(struct q6asm_wma_cfg));
+
+		wma_cfg.sample_rate =  params->codec.sample_rate;
+		wma_cfg.num_channels = params->codec.ch_in;
+		wma_cfg.bytes_per_sec = params->codec.bit_rate / 8;
+		wma_cfg.block_align = params->codec.align;
+		wma_cfg.bits_per_sample = prtd->bits_per_sample;
+		wma_cfg.enc_options = wma->encoder_option;
+		wma_cfg.adv_enc_options = wma->adv_encoder_option;
+		wma_cfg.adv_enc_options2 = wma->adv_encoder_option2;
+
+		if (wma_cfg.num_channels == 1)
+			wma_cfg.channel_mask = 4; /* Mono Center */
+		else if (wma_cfg.num_channels == 2)
+			wma_cfg.channel_mask = 3; /* Stereo FL/FR */
+		else
+			return -EINVAL;
+
+		/* check the codec profile */
+		switch (params->codec.profile) {
+		case SND_AUDIOPROFILE_WMA9:
+			wma_cfg.fmtag = 0x161;
+			wma_v9 = 1;
+			break;
+
+		case SND_AUDIOPROFILE_WMA10:
+			wma_cfg.fmtag = 0x166;
+			break;
+
+		case SND_AUDIOPROFILE_WMA9_PRO:
+			wma_cfg.fmtag = 0x162;
+			break;
+
+		case SND_AUDIOPROFILE_WMA9_LOSSLESS:
+			wma_cfg.fmtag = 0x163;
+			break;
+
+		case SND_AUDIOPROFILE_WMA10_LOSSLESS:
+			wma_cfg.fmtag = 0x167;
+			break;
+
+		default:
+			dev_err(dev, "Unknown WMA profile:%x\n",
+				params->codec.profile);
+			return -EIO;
+		}
+
+		if (wma_v9)
+			ret = q6asm_stream_media_format_block_wma_v9(
+					prtd->audio_client, &wma_cfg);
+		else
+			ret = q6asm_stream_media_format_block_wma_v10(
+					prtd->audio_client, &wma_cfg);
+		if (ret < 0) {
+			dev_err(dev, "WMA9 CMD failed:%d\n", ret);
+			return -EIO;
+		}
+		break;
+
+	case SND_AUDIOCODEC_ALAC:
+		memset(&alac_cfg, 0x0, sizeof(alac_cfg));
+		alac = &codec_options->alac_d;
+
+		alac_cfg.sample_rate = params->codec.sample_rate;
+		alac_cfg.avg_bit_rate = params->codec.bit_rate;
+		alac_cfg.bit_depth = prtd->bits_per_sample;
+		alac_cfg.num_channels = params->codec.ch_in;
+
+		alac_cfg.frame_length = alac->frame_length;
+		alac_cfg.pb = alac->pb;
+		alac_cfg.mb = alac->mb;
+		alac_cfg.kb = alac->kb;
+		alac_cfg.max_run = alac->max_run;
+		alac_cfg.compatible_version = alac->compatible_version;
+		alac_cfg.max_frame_bytes = alac->max_frame_bytes;
+
+		switch (params->codec.ch_in) {
+		case 1:
+			alac_cfg.channel_layout_tag = ALAC_CH_LAYOUT_MONO;
+			break;
+		case 2:
+			alac_cfg.channel_layout_tag = ALAC_CH_LAYOUT_STEREO;
+			break;
+		}
+		ret = q6asm_stream_media_format_block_alac(prtd->audio_client,
+							   &alac_cfg);
+		if (ret < 0) {
+			dev_err(dev, "ALAC CMD Format block failed:%d\n", ret);
+			return -EIO;
+		}
+		break;
+
+	case SND_AUDIOCODEC_APE:
+		memset(&ape_cfg, 0x0, sizeof(ape_cfg));
+		ape = &codec_options->ape_d;
+
+		ape_cfg.sample_rate = params->codec.sample_rate;
+		ape_cfg.num_channels = params->codec.ch_in;
+		ape_cfg.bits_per_sample = prtd->bits_per_sample;
+
+		ape_cfg.compatible_version = ape->compatible_version;
+		ape_cfg.compression_level = ape->compression_level;
+		ape_cfg.format_flags = ape->format_flags;
+		ape_cfg.blocks_per_frame = ape->blocks_per_frame;
+		ape_cfg.final_frame_blocks = ape->final_frame_blocks;
+		ape_cfg.total_frames = ape->total_frames;
+		ape_cfg.seek_table_present = ape->seek_table_present;
+
+		ret = q6asm_stream_media_format_block_ape(prtd->audio_client,
+							  &ape_cfg);
+		if (ret < 0) {
+			dev_err(dev, "APE CMD Format block failed:%d\n", ret);
+			return -EIO;
+		}
+		break;
+
 	default:
 		break;
 	}
···
 	caps->max_fragment_size = COMPR_PLAYBACK_MAX_FRAGMENT_SIZE;
 	caps->min_fragments = COMPR_PLAYBACK_MIN_NUM_FRAGMENTS;
 	caps->max_fragments = COMPR_PLAYBACK_MAX_NUM_FRAGMENTS;
-	caps->num_codecs = 2;
+	caps->num_codecs = 5;
 	caps->codecs[0] = SND_AUDIOCODEC_MP3;
 	caps->codecs[1] = SND_AUDIOCODEC_FLAC;
+	caps->codecs[2] = SND_AUDIOCODEC_WMA;
+	caps->codecs[3] = SND_AUDIOCODEC_ALAC;
+	caps->codecs[4] = SND_AUDIOCODEC_APE;
 
 	return 0;
 }
···
 	.compr_ops	= &q6asm_dai_compr_ops,
 };
 
-static struct snd_soc_dai_driver q6asm_fe_dais[] = {
+static struct snd_soc_dai_driver q6asm_fe_dais_template[] = {
 	Q6ASM_FEDAI_DRIVER(1),
 	Q6ASM_FEDAI_DRIVER(2),
 	Q6ASM_FEDAI_DRIVER(3),
···
 static int of_q6asm_parse_dai_data(struct device *dev,
 				    struct q6asm_dai_data *pdata)
 {
-	static struct snd_soc_dai_driver *dai_drv;
+	struct snd_soc_dai_driver *dai_drv;
 	struct snd_soc_pcm_stream empty_stream;
 	struct device_node *node;
-	int ret, id, dir;
+	int ret, id, dir, idx = 0;
+
+
+	pdata->num_dais = of_get_child_count(dev->of_node);
+	if (!pdata->num_dais) {
+		dev_err(dev, "No dais found in DT\n");
+		return -EINVAL;
+	}
+
+	pdata->dais = devm_kcalloc(dev, pdata->num_dais, sizeof(*dai_drv),
+				   GFP_KERNEL);
+	if (!pdata->dais)
+		return -ENOMEM;
 
 	memset(&empty_stream, 0, sizeof(empty_stream));
 
···
 			continue;
 		}
 
-		dai_drv = &q6asm_fe_dais[id];
+		dai_drv = &pdata->dais[idx++];
+		*dai_drv = q6asm_fe_dais_template[id];
 
 		ret = of_property_read_u32(node, "direction", &dir);
 		if (ret)
···
 
 	dev_set_drvdata(dev, pdata);
 
-	of_q6asm_parse_dai_data(dev, pdata);
+	rc = of_q6asm_parse_dai_data(dev, pdata);
+	if (rc)
+		return rc;
 
 	return devm_snd_soc_register_component(dev, &q6asm_fe_dai_component,
-					q6asm_fe_dais,
-					ARRAY_SIZE(q6asm_fe_dais));
+					       pdata->dais, pdata->num_dais);
 }
 
 static const struct of_device_id q6asm_dai_device_id[] = {

sound/soc/qcom/qdsp6/q6asm.c

···
 #define ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2	0x00010DA5
 #define ASM_MEDIA_FMT_MP3			0x00010BE9
 #define ASM_MEDIA_FMT_FLAC			0x00010C16
+#define ASM_MEDIA_FMT_WMA_V9			0x00010DA8
+#define ASM_MEDIA_FMT_WMA_V10			0x00010DA7
 #define ASM_DATA_CMD_WRITE_V2			0x00010DAB
 #define ASM_DATA_CMD_READ_V2			0x00010DAC
 #define ASM_SESSION_CMD_SUSPEND			0x00010DEC
···
 #define ASM_STREAM_CMD_OPEN_READ_V3                 0x00010DB4
 #define ASM_DATA_EVENT_READ_DONE_V2 0x00010D9A
 #define ASM_STREAM_CMD_OPEN_READWRITE_V2        0x00010D8D
+#define ASM_MEDIA_FMT_ALAC			0x00012f31
+#define ASM_MEDIA_FMT_APE			0x00012f32
 
 
 #define ASM_LEGACY_STREAM_SESSION	0
···
 	u32 max_frame_size;
 	u16 sample_size;
 	u16 reserved;
+} __packed;
+
+struct asm_wmastdv9_fmt_blk_v2 {
+	struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
+	u16          fmtag;
+	u16          num_channels;
+	u32          sample_rate;
+	u32          bytes_per_sec;
+	u16          blk_align;
+	u16          bits_per_sample;
+	u32          channel_mask;
+	u16          enc_options;
+	u16          reserved;
+} __packed;
+
+struct asm_wmaprov10_fmt_blk_v2 {
+	struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
+	u16          fmtag;
+	u16          num_channels;
+	u32          sample_rate;
+	u32          bytes_per_sec;
+	u16          blk_align;
+	u16          bits_per_sample;
+	u32          channel_mask;
+	u16          enc_options;
+	u16          advanced_enc_options1;
+	u32          advanced_enc_options2;
+} __packed;
+
+struct asm_alac_fmt_blk_v2 {
+	struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
+	u32 frame_length;
+	u8 compatible_version;
+	u8 bit_depth;
+	u8 pb;
+	u8 mb;
+	u8 kb;
+	u8 num_channels;
+	u16 max_run;
+	u32 max_frame_bytes;
+	u32 avg_bit_rate;
+	u32 sample_rate;
+	u32 channel_layout_tag;
+} __packed;
+
+struct asm_ape_fmt_blk_v2 {
+	struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
+	u16 compatible_version;
+	u16 compression_level;
+	u32 format_flags;
+	u32 blocks_per_frame;
+	u32 final_frame_blocks;
+	u32 total_frames;
+	u16 bits_per_sample;
+	u16 num_channels;
+	u32 sample_rate;
+	u32 seek_table_present;
 } __packed;
 
 struct asm_stream_cmd_set_encdec_param {
···
  * Return: Will be an negative value on error or zero on success
  */
 int q6asm_open_write(struct audio_client *ac, uint32_t format,
-		     uint16_t bits_per_sample)
+		     u32 codec_profile, uint16_t bits_per_sample)
 {
 	struct asm_stream_cmd_open_write_v3 *open;
 	struct apr_pkt *pkt;
···
 		break;
 	case SND_AUDIOCODEC_FLAC:
 		open->dec_fmt_id = ASM_MEDIA_FMT_FLAC;
+		break;
+	case SND_AUDIOCODEC_WMA:
+		switch (codec_profile) {
+		case SND_AUDIOPROFILE_WMA9:
+			open->dec_fmt_id = ASM_MEDIA_FMT_WMA_V9;
+			break;
+		case SND_AUDIOPROFILE_WMA10:
+		case SND_AUDIOPROFILE_WMA9_PRO:
+		case SND_AUDIOPROFILE_WMA9_LOSSLESS:
+		case SND_AUDIOPROFILE_WMA10_LOSSLESS:
+			open->dec_fmt_id = ASM_MEDIA_FMT_WMA_V10;
+			break;
+		default:
+			dev_err(ac->dev, "Invalid codec profile 0x%x\n",
+				codec_profile);
+			rc = -EINVAL;
+			goto err;
+		}
+		break;
+	case SND_AUDIOCODEC_ALAC:
+		open->dec_fmt_id = ASM_MEDIA_FMT_ALAC;
+		break;
+	case SND_AUDIOCODEC_APE:
+		open->dec_fmt_id = ASM_MEDIA_FMT_APE;
 		break;
 	default:
 		dev_err(ac->dev, "Invalid format 0x%x\n", format);
···
 	return rc;
 }
 EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_flac);
+
+int q6asm_stream_media_format_block_wma_v9(struct audio_client *ac,
+					   struct q6asm_wma_cfg *cfg)
+{
+	struct asm_wmastdv9_fmt_blk_v2 *fmt;
+	struct apr_pkt *pkt;
+	void *p;
+	int rc, pkt_size;
+
+	pkt_size = APR_HDR_SIZE + sizeof(*fmt);
+	p = kzalloc(pkt_size, GFP_KERNEL);
+	if (!p)
+		return -ENOMEM;
+
+	pkt = p;
+	fmt = p + APR_HDR_SIZE;
+
+	q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
+
+	pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
+	fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk);
+	fmt->fmtag = cfg->fmtag;
+	fmt->num_channels = cfg->num_channels;
+	fmt->sample_rate = cfg->sample_rate;
+	fmt->bytes_per_sec = cfg->bytes_per_sec;
+	fmt->blk_align = cfg->block_align;
+	fmt->bits_per_sample = cfg->bits_per_sample;
+	fmt->channel_mask = cfg->channel_mask;
+	fmt->enc_options = cfg->enc_options;
+	fmt->reserved = 0;
+
+	rc = q6asm_ac_send_cmd_sync(ac, pkt);
+	kfree(pkt);
+
+	return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_wma_v9);
+
+int q6asm_stream_media_format_block_wma_v10(struct audio_client *ac,
+					    struct q6asm_wma_cfg *cfg)
+{
+	struct asm_wmaprov10_fmt_blk_v2 *fmt;
+	struct apr_pkt *pkt;
+	void *p;
+	int rc, pkt_size;
+
+	pkt_size = APR_HDR_SIZE + sizeof(*fmt);
+	p = kzalloc(pkt_size, GFP_KERNEL);
+	if (!p)
+		return -ENOMEM;
+
+	pkt = p;
+	fmt = p + APR_HDR_SIZE;
+
+	q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
+
+	pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
+	fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk);
+	fmt->fmtag = cfg->fmtag;
+	fmt->num_channels = cfg->num_channels;
+	fmt->sample_rate = cfg->sample_rate;
+	fmt->bytes_per_sec = cfg->bytes_per_sec;
+	fmt->blk_align = cfg->block_align;
+	fmt->bits_per_sample = cfg->bits_per_sample;
+	fmt->channel_mask = cfg->channel_mask;
+	fmt->enc_options = cfg->enc_options;
+	fmt->advanced_enc_options1 = cfg->adv_enc_options;
+	fmt->advanced_enc_options2 = cfg->adv_enc_options2;
+
+	rc = q6asm_ac_send_cmd_sync(ac, pkt);
+	kfree(pkt);
+
+	return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_wma_v10);
+
+int q6asm_stream_media_format_block_alac(struct audio_client *ac,
+					 struct q6asm_alac_cfg *cfg)
+{
+	struct asm_alac_fmt_blk_v2 *fmt;
+	struct apr_pkt *pkt;
+	void *p;
+	int rc, pkt_size;
+
+	pkt_size = APR_HDR_SIZE + sizeof(*fmt);
+	p = kzalloc(pkt_size, GFP_KERNEL);
+	if (!p)
+		return -ENOMEM;
+
+	pkt = p;
+	fmt = p + APR_HDR_SIZE;
+
+	q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
+
+	pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
+	fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk);
+
+	fmt->frame_length = cfg->frame_length;
+	fmt->compatible_version = cfg->compatible_version;
+	fmt->bit_depth =  cfg->bit_depth;
+	fmt->num_channels = cfg->num_channels;
+	fmt->max_run = cfg->max_run;
+	fmt->max_frame_bytes = cfg->max_frame_bytes;
+	fmt->avg_bit_rate = cfg->avg_bit_rate;
+	fmt->sample_rate = cfg->sample_rate;
+	fmt->channel_layout_tag = cfg->channel_layout_tag;
+	fmt->pb = cfg->pb;
+	fmt->mb = cfg->mb;
+	fmt->kb = cfg->kb;
+
+	rc = q6asm_ac_send_cmd_sync(ac, pkt);
+	kfree(pkt);
+
+	return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_alac);
+
+int q6asm_stream_media_format_block_ape(struct audio_client *ac,
+					struct q6asm_ape_cfg *cfg)
+{
+	struct asm_ape_fmt_blk_v2 *fmt;
+	struct apr_pkt *pkt;
+	void *p;
+	int rc, pkt_size;
+
+	pkt_size = APR_HDR_SIZE + sizeof(*fmt);
+	p = kzalloc(pkt_size, GFP_KERNEL);
+	if (!p)
+		return -ENOMEM;
+
+	pkt = p;
+	fmt = p + APR_HDR_SIZE;
+
+	q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
+
+	pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
+	fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk);
+
+	fmt->compatible_version = cfg->compatible_version;
+	fmt->compression_level = cfg->compression_level;
+	fmt->format_flags = cfg->format_flags;
+	fmt->blocks_per_frame = cfg->blocks_per_frame;
+	fmt->final_frame_blocks = cfg->final_frame_blocks;
+	fmt->total_frames = cfg->total_frames;
+	fmt->bits_per_sample = cfg->bits_per_sample;
+	fmt->num_channels = cfg->num_channels;
+	fmt->sample_rate = cfg->sample_rate;
+	fmt->seek_table_present = cfg->seek_table_present;
+
+	rc = q6asm_ac_send_cmd_sync(ac, pkt);
+	kfree(pkt);
+
+	return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_ape);
+
 /**
  * q6asm_enc_cfg_blk_pcm_format_support() - setup pcm configuration for capture
  *

sound/soc/qcom/qdsp6/q6asm.h

···
         u16 md5_sum;
 };
 
+struct q6asm_wma_cfg {
+	u32 fmtag;
+	u32 num_channels;
+	u32 sample_rate;
+	u32 bytes_per_sec;
+	u32 block_align;
+	u32 bits_per_sample;
+	u32 channel_mask;
+	u32 enc_options;
+	u32 adv_enc_options;
+	u32 adv_enc_options2;
+};
+
+struct q6asm_alac_cfg {
+	u32 frame_length;
+	u8 compatible_version;
+	u8 bit_depth;
+	u8 pb;
+	u8 mb;
+	u8 kb;
+	u8 num_channels;
+	u16 max_run;
+	u32 max_frame_bytes;
+	u32 avg_bit_rate;
+	u32 sample_rate;
+	u32 channel_layout_tag;
+};
+
+struct q6asm_ape_cfg {
+	u16 compatible_version;
+	u16 compression_level;
+	u32 format_flags;
+	u32 blocks_per_frame;
+	u32 final_frame_blocks;
+	u32 total_frames;
+	u16 bits_per_sample;
+	u16 num_channels;
+	u32 sample_rate;
+	u32 seek_table_present;
+};
+
 typedef void (*q6asm_cb) (uint32_t opcode, uint32_t token,
 			  void *payload, void *priv);
 struct audio_client;
···
 int q6asm_write_async(struct audio_client *ac, uint32_t len, uint32_t msw_ts,
 		       uint32_t lsw_ts, uint32_t flags);
 int q6asm_open_write(struct audio_client *ac, uint32_t format,
-		     uint16_t bits_per_sample);
+		     u32 codec_profile, uint16_t bits_per_sample);
 
 int q6asm_open_read(struct audio_client *ac, uint32_t format,
 		     uint16_t bits_per_sample);
···
 					  uint16_t bits_per_sample);
 int q6asm_stream_media_format_block_flac(struct audio_client *ac,
 					 struct q6asm_flac_cfg *cfg);
+int q6asm_stream_media_format_block_wma_v9(struct audio_client *ac,
+					   struct q6asm_wma_cfg *cfg);
+int q6asm_stream_media_format_block_wma_v10(struct audio_client *ac,
+					    struct q6asm_wma_cfg *cfg);
+int q6asm_stream_media_format_block_alac(struct audio_client *ac,
+					 struct q6asm_alac_cfg *cfg);
+int q6asm_stream_media_format_block_ape(struct audio_client *ac,
+					struct q6asm_ape_cfg *cfg);
 int q6asm_run(struct audio_client *ac, uint32_t flags, uint32_t msw_ts,
 	      uint32_t lsw_ts);
 int q6asm_run_nowait(struct audio_client *ac, uint32_t flags, uint32_t msw_ts,

sound/soc/qcom/qdsp6/q6routing.c

···
 	{"MM_UL6", NULL, "MultiMedia6 Mixer"},
 	{"MM_UL7", NULL, "MultiMedia7 Mixer"},
 	{"MM_UL8", NULL, "MultiMedia8 Mixer"},
-
-	{"MM_DL1",  NULL, "MultiMedia1 Playback" },
-	{"MM_DL2",  NULL, "MultiMedia2 Playback" },
-	{"MM_DL3",  NULL, "MultiMedia3 Playback" },
-	{"MM_DL4",  NULL, "MultiMedia4 Playback" },
-	{"MM_DL5",  NULL, "MultiMedia5 Playback" },
-	{"MM_DL6",  NULL, "MultiMedia6 Playback" },
-	{"MM_DL7",  NULL, "MultiMedia7 Playback" },
-	{"MM_DL8",  NULL, "MultiMedia8 Playback" },
-
-	{"MultiMedia1 Capture", NULL, "MM_UL1"},
-	{"MultiMedia2 Capture", NULL, "MM_UL2"},
-	{"MultiMedia3 Capture", NULL, "MM_UL3"},
-	{"MultiMedia4 Capture", NULL, "MM_UL4"},
-	{"MultiMedia5 Capture", NULL, "MM_UL5"},
-	{"MultiMedia6 Capture", NULL, "MM_UL6"},
-	{"MultiMedia7 Capture", NULL, "MM_UL7"},
-	{"MultiMedia8 Capture", NULL, "MM_UL8"},
-
 };
 
 static int routing_hw_params(struct snd_soc_component *component,
···
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct msm_routing_data *data = dev_get_drvdata(component->dev);
-	unsigned int be_id = rtd->cpu_dai->id;
+	unsigned int be_id = asoc_rtd_to_cpu(rtd, 0)->id;
 	struct session_data *session;
 	int path_type;
 

sound/soc/qcom/sdm845.c

···
 #include <sound/pcm_params.h>
 #include <sound/jack.h>
 #include <sound/soc.h>
+#include <linux/soundwire/sdw.h>
 #include <uapi/linux/input-event-codes.h>
 #include "common.h"
 #include "qdsp6/q6afe.h"
···
 struct sdm845_snd_data {
 	struct snd_soc_jack jack;
 	bool jack_setup;
+	bool stream_prepared[SLIM_MAX_RX_PORTS];
 	struct snd_soc_card *card;
 	uint32_t pri_mi2s_clk_count;
 	uint32_t sec_mi2s_clk_count;
 	uint32_t quat_tdm_clk_count;
+	struct sdw_stream_runtime *sruntime[SLIM_MAX_RX_PORTS];
 };
 
 static unsigned int tdm_slot_offset[8] = {0, 4, 8, 12, 16, 20, 24, 28};
···
 				     struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai_link *dai_link = rtd->dai_link;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+	struct snd_soc_dai *codec_dai;
+	struct sdm845_snd_data *pdata = snd_soc_card_get_drvdata(rtd->card);
 	u32 rx_ch[SLIM_MAX_RX_PORTS], tx_ch[SLIM_MAX_TX_PORTS];
+	struct sdw_stream_runtime *sruntime;
 	u32 rx_ch_cnt = 0, tx_ch_cnt = 0;
 	int ret = 0, i;
 
-	for (i = 0 ; i < dai_link->num_codecs; i++) {
-		ret = snd_soc_dai_get_channel_map(rtd->codec_dais[i],
+	for_each_rtd_codec_dais(rtd, i, codec_dai) {
+		sruntime = snd_soc_dai_get_sdw_stream(codec_dai,
+						      substream->stream);
+		if (sruntime != ERR_PTR(-ENOTSUPP))
+			pdata->sruntime[cpu_dai->id] = sruntime;
+
+		ret = snd_soc_dai_get_channel_map(codec_dai,
 				&tx_ch_cnt, tx_ch, &rx_ch_cnt, rx_ch);
 
 		if (ret != 0 && ret != -ENOTSUPP) {
···
 					struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+	struct snd_soc_dai *codec_dai;
 	int ret = 0, j;
 	int channels, slot_width;
 
···
 		}
 	}
 
-	for (j = 0; j < rtd->num_codecs; j++) {
-		struct snd_soc_dai *codec_dai = rtd->codec_dais[j];
+	for_each_rtd_codec_dais(rtd, j, codec_dai) {
 
 		if (!strcmp(codec_dai->component->name_prefix, "Left")) {
 			ret = snd_soc_dai_set_tdm_slot(
···
 					struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	int ret = 0;
 
 	switch (cpu_dai->id) {
···
 {
 	struct snd_soc_component *component;
 	struct snd_soc_card *card = rtd->card;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	struct sdm845_snd_data *pdata = snd_soc_card_get_drvdata(card);
 	struct snd_jack *jack;
-	struct snd_soc_dai_link *dai_link = rtd->dai_link;
 	/*
 	 * Codec SLIMBUS configuration
 	 * RX1, RX2, RX3, RX4, RX5, RX6, RX7, RX8, RX9, RX10, RX11, RX12, RX13
···
 		}
 		break;
 	case SLIMBUS_0_RX...SLIMBUS_6_TX:
-		for (i = 0 ; i < dai_link->num_codecs; i++) {
-			rval = snd_soc_dai_set_channel_map(rtd->codec_dais[i],
+		for_each_rtd_codec_dais(rtd, i, codec_dai) {
+			rval = snd_soc_dai_set_channel_map(codec_dai,
 							  ARRAY_SIZE(tx_ch),
 							  tx_ch,
 							  ARRAY_SIZE(rx_ch),
···
 			if (rval != 0 && rval != -ENOTSUPP)
 				return rval;
 
-			snd_soc_dai_set_sysclk(rtd->codec_dais[i], 0,
+			snd_soc_dai_set_sysclk(codec_dai, 0,
 					       WCD934X_DEFAULT_MCLK_RATE,
 					       SNDRV_PCM_STREAM_PLAYBACK);
 		}
···
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_card *card = rtd->card;
 	struct sdm845_snd_data *data = snd_soc_card_get_drvdata(card);
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	int j;
 	int ret;
 
···
 
 		codec_dai_fmt |= SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_DSP_B;
 
-		for (j = 0; j < rtd->num_codecs; j++) {
-			codec_dai = rtd->codec_dais[j];
+		for_each_rtd_codec_dais(rtd, j, codec_dai) {
 
 			if (!strcmp(codec_dai->component->name_prefix,
 				    "Left")) {
···
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_card *card = rtd->card;
 	struct sdm845_snd_data *data = snd_soc_card_get_drvdata(card);
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 
 	switch (cpu_dai->id) {
 	case PRIMARY_MI2S_RX:
···
 	}
 }
 
+static int sdm845_snd_prepare(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct sdm845_snd_data *data = snd_soc_card_get_drvdata(rtd->card);
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+	struct sdw_stream_runtime *sruntime = data->sruntime[cpu_dai->id];
+	int ret;
+
+	if (!sruntime)
+		return 0;
+
+	if (data->stream_prepared[cpu_dai->id]) {
+		sdw_disable_stream(sruntime);
+		sdw_deprepare_stream(sruntime);
+		data->stream_prepared[cpu_dai->id] = false;
+	}
+
+	ret = sdw_prepare_stream(sruntime);
+	if (ret)
+		return ret;
+
+	/**
+	 * NOTE: there is a strict hw requirement about the ordering of port
+	 * enables and actual WSA881x PA enable. PA enable should only happen
+	 * after soundwire ports are enabled if not DC on the line is
+	 * accumulated resulting in Click/Pop Noise
+	 * PA enable/mute are handled as part of codec DAPM and digital mute.
+	 */
+
+	ret = sdw_enable_stream(sruntime);
+	if (ret) {
+		sdw_deprepare_stream(sruntime);
+		return ret;
+	}
+	data->stream_prepared[cpu_dai->id] = true;
+
+	return ret;
+}
+
+static int sdm845_snd_hw_free(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct sdm845_snd_data *data = snd_soc_card_get_drvdata(rtd->card);
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+	struct sdw_stream_runtime *sruntime = data->sruntime[cpu_dai->id];
+
+	if (sruntime && data->stream_prepared[cpu_dai->id]) {
+		sdw_disable_stream(sruntime);
+		sdw_deprepare_stream(sruntime);
+		data->stream_prepared[cpu_dai->id] = false;
+	}
+
+	return 0;
+}
+
 static const struct snd_soc_ops sdm845_be_ops = {
 	.hw_params = sdm845_snd_hw_params,
+	.hw_free = sdm845_snd_hw_free,
+	.prepare = sdm845_snd_prepare,
 	.startup = sdm845_snd_startup,
 	.shutdown = sdm845_snd_shutdown,
 };

sound/soc/qcom/storm.c

···
 	 */
 	sysclk_freq = rate * bitwidth * 2 * STORM_SYSCLK_MULT;
 
-	ret = snd_soc_dai_set_sysclk(soc_runtime->cpu_dai, 0, sysclk_freq, 0);
+	ret = snd_soc_dai_set_sysclk(asoc_rtd_to_cpu(soc_runtime, 0), 0, sysclk_freq, 0);
 	if (ret) {
 		dev_err(card->dev, "error setting sysclk to %u: %d\n",
 			sysclk_freq, ret);

sound/soc/rockchip/rk3288_hdmi_analog.c

···
 {
 	int ret = 0;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	int mclk;
 
 	switch (params_rate(params)) {

sound/soc/rockchip/rk3399_gru_sound.c

···
 
 	mclk = params_rate(params) * SOUND_FS;
 
-	ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, 0, mclk, 0);
+	ret = snd_soc_dai_set_sysclk(asoc_rtd_to_cpu(rtd, 0), 0, mclk, 0);
 	if (ret) {
 		dev_err(rtd->card->dev, "%s() error setting sysclk to %u: %d\n",
 				__func__, mclk, ret);
···
 			     struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	unsigned int mclk;
 	int ret;
 
···
 			     struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	int mclk, ret;
 
 	/* in bypass mode, the mclk has to be one of the frequencies below */
···
 
 static int rockchip_sound_da7219_init(struct snd_soc_pcm_runtime *rtd)
 {
-	struct snd_soc_component *component = rtd->codec_dais[0]->component;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	int ret;
 
 	/* We need default MCLK and PLL settings for the accessory detection */
···
 
 	mclk = params_rate(params) * SOUND_FS;
 
-	ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, 0, mclk, 0);
+	ret = snd_soc_dai_set_sysclk(asoc_rtd_to_cpu(rtd, 0), 0, mclk, 0);
 	if (ret) {
 		dev_err(rtd->card->dev, "%s() error setting sysclk to %u: %d\n",
 				__func__, mclk, ret);

sound/soc/rockchip/rockchip_max98090.c

···
 {
 	int ret = 0;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	int mclk;
 
 	switch (params_rate(params)) {
···
 static int rk_hdmi_init(struct snd_soc_pcm_runtime *runtime)
 {
 	struct snd_soc_card *card = runtime->card;
-	struct snd_soc_component *component = runtime->codec_dai->component;
+	struct snd_soc_component *component = asoc_rtd_to_codec(runtime, 0)->component;
 	int ret;
 
 	/* enable jack detection */

sound/soc/rockchip/rockchip_rt5645.c

···
 {
 	int ret = 0;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	int mclk;
 
 	switch (params_rate(params)) {
···
 		return ret;
 	}
 
-	return rt5645_set_jack_detect(runtime->codec_dai->component,
+	return rt5645_set_jack_detect(asoc_rtd_to_codec(runtime, 0)->component,
 				     &headset_jack,
 				     &headset_jack,
 				     &headset_jack);

sound/soc/samsung/Kconfig

···
 
 config SND_SOC_BELLS
 	tristate "Audio support for Wolfson Bells"
-	depends on MFD_ARIZONA && I2C && SPI_MASTER
+	depends on MFD_ARIZONA && MFD_WM5102 && MFD_WM5110 && I2C && SPI_MASTER
 	depends on MACH_WLF_CRAGG_6410 || COMPILE_TEST
 	select SND_SAMSUNG_I2S
 	select SND_SOC_WM5102
···
 
 config SND_SOC_SAMSUNG_TM2_WM5110
 	tristate "SoC I2S Audio support for WM5110 on TM2 board"
-	depends on SND_SOC_SAMSUNG && MFD_ARIZONA && I2C && SPI_MASTER
+	depends on SND_SOC_SAMSUNG && MFD_ARIZONA && MFD_WM5110 && I2C && SPI_MASTER
 	depends on GPIOLIB || COMPILE_TEST
 	select SND_SOC_MAX98504
 	select SND_SOC_WM5110

sound/soc/samsung/arndale.c

···
 				    struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	int rfs, ret;
 	unsigned long rclk;
 
···
 				    struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	unsigned int rfs, rclk;
 
 	/* Ensure AIF1CLK is >= 3 MHz for optimal performance */
···
 
 	ret = devm_snd_soc_register_card(card->dev, card);
 	if (ret) {
-		dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret);
+		if (ret != -EPROBE_DEFER)
+			dev_err(&pdev->dev,
+				"snd_soc_register_card() failed: %d\n", ret);
 		goto err_put_of_nodes;
 	}
 	return 0;

sound/soc/samsung/bells.c

···
 	int ret;
 
 	rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[DAI_DSP_CODEC]);
-	codec_dai = rtd->codec_dai;
+	codec_dai = asoc_rtd_to_codec(rtd, 0);
 	component = codec_dai->component;
 
 	if (dapm->dev != codec_dai->dev)
···
 	int ret;
 
 	rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[DAI_DSP_CODEC]);
-	codec_dai = rtd->codec_dai;
+	codec_dai = asoc_rtd_to_codec(rtd, 0);
 	component = codec_dai->component;
 
 	if (dapm->dev != codec_dai->dev)
···
 	int ret;
 
 	rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[DAI_AP_DSP]);
-	wm0010 = rtd->codec_dai->component;
+	wm0010 = asoc_rtd_to_codec(rtd, 0)->component;
 
 	rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[DAI_DSP_CODEC]);
-	component = rtd->codec_dai->component;
-	aif1_dai = rtd->codec_dai;
+	component = asoc_rtd_to_codec(rtd, 0)->component;
+	aif1_dai = asoc_rtd_to_codec(rtd, 0);
 
 	ret = snd_soc_component_set_sysclk(component, ARIZONA_CLK_SYSCLK,
 				       ARIZONA_CLK_SRC_FLL1,
···
 	}
 
 	rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[DAI_CODEC_CP]);
-	aif2_dai = rtd->cpu_dai;
+	aif2_dai = asoc_rtd_to_cpu(rtd, 0);
 
 	ret = snd_soc_dai_set_sysclk(aif2_dai, ARIZONA_CLK_ASYNCCLK, 0, 0);
 	if (ret != 0) {
···
 		return 0;
 
 	rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[DAI_CODEC_SUB]);
-	aif3_dai = rtd->cpu_dai;
-	wm9081_dai = rtd->codec_dai;
+	aif3_dai = asoc_rtd_to_cpu(rtd, 0);
+	wm9081_dai = asoc_rtd_to_codec(rtd, 0);
 
 	ret = snd_soc_dai_set_sysclk(aif3_dai, ARIZONA_CLK_SYSCLK, 0, 0);
 	if (ret != 0) {

sound/soc/samsung/h1940_uda1380.c

···
 				struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	int div;
 	int ret;
 	unsigned int rate = params_rate(params);

sound/soc/samsung/i2s.c

···
 	struct samsung_i2s_priv *priv = snd_soc_dai_get_drvdata(dai);
 	int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE);
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct i2s_dai *i2s = to_info(rtd->cpu_dai);
+	struct i2s_dai *i2s = to_info(asoc_rtd_to_cpu(rtd, 0));
 	unsigned long flags;
 
 	switch (cmd) {

sound/soc/samsung/jive_wm8750.c

···
 			  struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	struct s3c_i2sv2_rate_calc div;
 	unsigned int clk = 0;
 	int ret = 0;

sound/soc/samsung/littlemill.c

···
 	int ret;
 
 	rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[0]);
-	aif1_dai = rtd->codec_dai;
+	aif1_dai = asoc_rtd_to_codec(rtd, 0);
 
 	if (dapm->dev != aif1_dai->dev)
 		return 0;
···
 	int ret;
 
 	rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[0]);
-	aif1_dai = rtd->codec_dai;
+	aif1_dai = asoc_rtd_to_codec(rtd, 0);
 
 	if (dapm->dev != aif1_dai->dev)
 		return 0;
···
 				struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	int ret;
 
 	sample_rate = params_rate(params);
···
 	int ret;
 
 	rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[1]);
-	aif2_dai = rtd->cpu_dai;
+	aif2_dai = asoc_rtd_to_cpu(rtd, 0);
 
 	switch (event) {
 	case SND_SOC_DAPM_PRE_PMU:
···
 	int ret;
 
 	rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[0]);
-	component = rtd->codec_dai->component;
-	aif1_dai = rtd->codec_dai;
+	component = asoc_rtd_to_codec(rtd, 0)->component;
+	aif1_dai = asoc_rtd_to_codec(rtd, 0);
 
 	rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[1]);
-	aif2_dai = rtd->cpu_dai;
+	aif2_dai = asoc_rtd_to_cpu(rtd, 0);
 
 	ret = snd_soc_dai_set_sysclk(aif1_dai, WM8994_SYSCLK_MCLK2,
 				     32768, SND_SOC_CLOCK_IN);
···
 	card->dev = &pdev->dev;
 
 	ret = devm_snd_soc_register_card(&pdev->dev, card);
-	if (ret)
+	if (ret && ret != -EPROBE_DEFER)
 		dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
 			ret);
 

sound/soc/samsung/lowland.c

···
 
 static int lowland_wm5100_init(struct snd_soc_pcm_runtime *rtd)
 {
-	struct snd_soc_component *component = rtd->codec_dai->component;
+	struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
 	int ret;
 
 	ret = snd_soc_component_set_sysclk(component, WM5100_CLK_SYSCLK,
···
 
 static int lowland_wm9081_init(struct snd_soc_pcm_runtime *rtd)
 {
-	struct snd_soc_component *component = rtd->codec_dai->component;
+	struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
 
 	snd_soc_dapm_nc_pin(&rtd->card->dapm, "LINEOUT");
 
···
 	card->dev = &pdev->dev;
 
 	ret = devm_snd_soc_register_card(&pdev->dev, card);
-	if (ret)
+	if (ret && ret != -EPROBE_DEFER)
 		dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
 			ret);
 

sound/soc/samsung/neo1973_wm8753.c

···
 	struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	unsigned int pll_out = 0, bclk = 0;
 	int ret = 0;
 	unsigned long iis_clkrate;
···
 static int neo1973_hifi_hw_free(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 
 	/* disable the PLL */
 	return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0, 0);
···
 	struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	unsigned int pcmdiv = 0;
 	int ret = 0;
 	unsigned long iis_clkrate;
···
 static int neo1973_voice_hw_free(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 
 	/* disable the PLL */
 	return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0, 0);

sound/soc/samsung/odroid.c

···
 		return ret;
 
 	if (rtd->num_codecs > 1) {
-		struct snd_soc_dai *codec_dai = rtd->codec_dais[1];
+		struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 1);
 
 		ret = snd_soc_dai_set_sysclk(codec_dai, 0, rclk_freq,
 					     SND_SOC_CLOCK_IN);
···
 
 	ret = devm_snd_soc_register_card(dev, card);
 	if (ret < 0) {
-		dev_err(dev, "snd_soc_register_card() failed: %d\n", ret);
+		if (ret != -EPROBE_DEFER)
+			dev_err(dev, "snd_soc_register_card() failed: %d\n",
+				ret);
 		goto err_put_clk_i2s;
 	}
 

sound/soc/samsung/pcm.c

···
 			       struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct s3c_pcm_info *pcm = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+	struct s3c_pcm_info *pcm = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
 	unsigned long flags;
 
 	dev_dbg(pcm->dev, "Entered %s\n", __func__);
···
 				 struct snd_soc_dai *socdai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct s3c_pcm_info *pcm = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+	struct s3c_pcm_info *pcm = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
 	void __iomem *regs = pcm->regs;
 	struct clk *clk;
 	int sclk_div, sync_div;

sound/soc/samsung/rx1950_uda1380.c

···
 				struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	int div;
 	int ret;
 	unsigned int rate = params_rate(params);

sound/soc/samsung/s3c-i2s-v2.c

···
 			       struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct s3c_i2sv2_info *i2s = to_info(rtd->cpu_dai);
+	struct s3c_i2sv2_info *i2s = to_info(asoc_rtd_to_cpu(rtd, 0));
 	int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE);
 	unsigned long irqs;
 	int ret = 0;

sound/soc/samsung/s3c24xx_simtec.c

···
 			    struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	int ret;
 
 	ret = snd_soc_dai_set_sysclk(codec_dai, 0,

sound/soc/samsung/s3c24xx_uda134x.c

···
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct s3c24xx_uda134x *priv = snd_soc_card_get_drvdata(rtd->card);
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	int ret = 0;
 
 	mutex_lock(&priv->clk_lock);
···
 					struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	unsigned int clk = 0;
 	int ret = 0;
 	int clk_source, fs_mode;

sound/soc/samsung/smartq_wm8987.c

···
 	struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	unsigned int clk = 0;
 	int ret;
 

sound/soc/samsung/smdk_spdif.c

···
 		struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	unsigned long pll_out, rclk_rate;
 	int ret, ratio;
 

sound/soc/samsung/smdk_wm8580.c

···
 	struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	unsigned int pll_out;
 	int rfs, ret;
 

sound/soc/samsung/smdk_wm8994.c

···
 	struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	unsigned int pll_out;
 	int ret;
 
···
 
 	ret = devm_snd_soc_register_card(&pdev->dev, card);
 
-	if (ret)
+	if (ret && ret != -EPROBE_DEFER)
 		dev_err(&pdev->dev, "snd_soc_register_card() failed:%d\n", ret);
 
 	return ret;

sound/soc/samsung/smdk_wm8994pcm.c

···
 			      struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	unsigned long mclk_freq;
 	int rfs, ret;
 
···
 
 	smdk_pcm.dev = &pdev->dev;
 	ret = devm_snd_soc_register_card(&pdev->dev, &smdk_pcm);
-	if (ret)
+	if (ret && ret != -EPROBE_DEFER)
 		dev_err(&pdev->dev, "snd_soc_register_card failed %d\n", ret);
 
 	return ret;

sound/soc/samsung/snow.c

···
 
 	/* In the multi-codec case codec_dais 0 is MAX98095 and 1 is HDMI. */
 	if (rtd->num_codecs > 1)
-		codec_dai = rtd->codec_dais[0];
+		codec_dai = asoc_rtd_to_codec(rtd, 0);
 	else
-		codec_dai = rtd->codec_dai;
+		codec_dai = asoc_rtd_to_codec(rtd, 0);
 
 	/* Set the MCLK rate for the codec */
 	return snd_soc_dai_set_sysclk(codec_dai, 0,
···
 
 	ret = devm_snd_soc_register_card(dev, card);
 	if (ret) {
-		dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
+		if (ret != -EPROBE_DEFER)
+			dev_err(&pdev->dev,
+				"snd_soc_register_card failed (%d)\n", ret);
 		return ret;
 	}
 

sound/soc/samsung/spdif.c

···
 				struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct samsung_spdif_info *spdif = to_info(rtd->cpu_dai);
+	struct samsung_spdif_info *spdif = to_info(asoc_rtd_to_cpu(rtd, 0));
 	unsigned long flags;
 
 	dev_dbg(spdif->dev, "Entered %s\n", __func__);
···
 				struct snd_soc_dai *socdai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct samsung_spdif_info *spdif = to_info(rtd->cpu_dai);
+	struct samsung_spdif_info *spdif = to_info(asoc_rtd_to_cpu(rtd, 0));
 	void __iomem *regs = spdif->regs;
 	struct snd_dmaengine_dai_dma_data *dma_data;
 	u32 con, clkcon, cstas;
···
 		return -EINVAL;
 	}
 
-	snd_soc_dai_set_dma_data(rtd->cpu_dai, substream, dma_data);
+	snd_soc_dai_set_dma_data(asoc_rtd_to_cpu(rtd, 0), substream, dma_data);
 
 	spin_lock_irqsave(&spdif->lock, flags);
 
···
 				struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct samsung_spdif_info *spdif = to_info(rtd->cpu_dai);
+	struct samsung_spdif_info *spdif = to_info(asoc_rtd_to_cpu(rtd, 0));
 	void __iomem *regs = spdif->regs;
 	u32 con, clkcon;
 

sound/soc/samsung/speyside.c

···
 	int ret;
 
 	rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[1]);
-	codec_dai = rtd->codec_dai;
+	codec_dai = asoc_rtd_to_codec(rtd, 0);
 
 	if (dapm->dev != codec_dai->dev)
 		return 0;
···
 	int ret;
 
 	rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[1]);
-	codec_dai = rtd->codec_dai;
+	codec_dai = asoc_rtd_to_codec(rtd, 0);
 
 	if (dapm->dev != codec_dai->dev)
 		return 0;
···
 
 static int speyside_wm0010_init(struct snd_soc_pcm_runtime *rtd)
 {
-	struct snd_soc_dai *dai = rtd->codec_dai;
+	struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0);
 	int ret;
 
 	ret = snd_soc_dai_set_sysclk(dai, 0, MCLK_AUDIO_RATE, 0);
···
 
 static int speyside_wm8996_init(struct snd_soc_pcm_runtime *rtd)
 {
-	struct snd_soc_dai *dai = rtd->codec_dai;
+	struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0);
 	struct snd_soc_component *component = dai->component;
 	int ret;
 
···
 	card->dev = &pdev->dev;
 
 	ret = devm_snd_soc_register_card(&pdev->dev, card);
-	if (ret)
+	if (ret && ret != -EPROBE_DEFER)
 		dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
 			ret);
 

sound/soc/samsung/tm2_wm5110.c

···
 				struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_component *component = rtd->codec_dai->component;
+	struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
 	struct tm2_machine_priv *priv = snd_soc_card_get_drvdata(rtd->card);
 
 	switch (params_rate(params)) {
···
 				struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_component *component = rtd->codec_dai->component;
+	struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
 	unsigned int asyncclk_rate;
 	int ret;
 
···
 static int tm2_aif2_hw_free(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_component *component = rtd->codec_dai->component;
+	struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
 	int ret;
 
 	/* disable FLL2 */
···
 			      struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	unsigned int bfs;
 	int bitwidth, ret;
 
···
 
 	rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[0]);
 
-	if (dapm->dev != rtd->codec_dai->dev)
+	if (dapm->dev != asoc_rtd_to_codec(rtd, 0)->dev)
 		return 0;
 
 	switch (level) {
···
 	int ret;
 
 	rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[TM2_DAI_AIF1]);
-	aif1_dai = rtd->codec_dai;
-	priv->component = rtd->codec_dai->component;
+	aif1_dai = asoc_rtd_to_codec(rtd, 0);
+	priv->component = asoc_rtd_to_codec(rtd, 0)->component;
 
 	ret = snd_soc_dai_set_sysclk(aif1_dai, ARIZONA_CLK_SYSCLK, 0, 0);
 	if (ret < 0) {
···
 	}
 
 	rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[TM2_DAI_AIF2]);
-	aif2_dai = rtd->codec_dai;
+	aif2_dai = asoc_rtd_to_codec(rtd, 0);
 
 	ret = snd_soc_dai_set_sysclk(aif2_dai, ARIZONA_CLK_ASYNCCLK, 0, 0);
 	if (ret < 0) {
···
 
 	ret = devm_snd_soc_register_card(dev, card);
 	if (ret < 0) {
-		dev_err(dev, "Failed to register card: %d\n", ret);
+		if (ret != -EPROBE_DEFER)
+			dev_err(dev, "Failed to register card: %d\n", ret);
 		goto dai_node_put;
 	}
 

sound/soc/samsung/tobermory.c

···
 	int ret;
 
 	rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[0]);
-	codec_dai = rtd->codec_dai;
+	codec_dai = asoc_rtd_to_codec(rtd, 0);
 
 	if (dapm->dev != codec_dai->dev)
 		return 0;
···
 	int ret;
 
 	rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[0]);
-	codec_dai = rtd->codec_dai;
+	codec_dai = asoc_rtd_to_codec(rtd, 0);
 
 	if (dapm->dev != codec_dai->dev)
 		return 0;
···
 	int ret;
 
 	rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[0]);
-	component = rtd->codec_dai->component;
-	codec_dai = rtd->codec_dai;
+	component = asoc_rtd_to_codec(rtd, 0)->component;
+	codec_dai = asoc_rtd_to_codec(rtd, 0);
 
 	ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK,
 				     32768, SND_SOC_CLOCK_IN);
···
 	card->dev = &pdev->dev;
 
 	ret = devm_snd_soc_register_card(&pdev->dev, card);
-	if (ret)
+	if (ret && ret != -EPROBE_DEFER)
 		dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
 			ret);
 

sound/soc/sh/dma-sh7760.c

···
 			    struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct camelot_pcm *cam = &cam_pcm_data[rtd->cpu_dai->id];
+	struct camelot_pcm *cam = &cam_pcm_data[asoc_rtd_to_cpu(rtd, 0)->id];
 	int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1;
 	int ret, dmairq;
 
···
 		ret = dmabrg_request_irq(dmairq, camelot_rxdma, cam);
 		if (unlikely(ret)) {
 			pr_debug("audio unit %d irqs already taken!\n",
-			     rtd->cpu_dai->id);
+			     asoc_rtd_to_cpu(rtd, 0)->id);
 			return -EBUSY;
 		}
 		(void)dmabrg_request_irq(dmairq + 1,camelot_rxdma, cam);
···
 		ret = dmabrg_request_irq(dmairq, camelot_txdma, cam);
 		if (unlikely(ret)) {
 			pr_debug("audio unit %d irqs already taken!\n",
-			     rtd->cpu_dai->id);
+			     asoc_rtd_to_cpu(rtd, 0)->id);
 			return -EBUSY;
 		}
 		(void)dmabrg_request_irq(dmairq + 1, camelot_txdma, cam);
···
 			     struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct camelot_pcm *cam = &cam_pcm_data[rtd->cpu_dai->id];
+	struct camelot_pcm *cam = &cam_pcm_data[asoc_rtd_to_cpu(rtd, 0)->id];
 	int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1;
 	int dmairq;
 
···
 			     struct snd_pcm_hw_params *hw_params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct camelot_pcm *cam = &cam_pcm_data[rtd->cpu_dai->id];
+	struct camelot_pcm *cam = &cam_pcm_data[asoc_rtd_to_cpu(rtd, 0)->id];
 	int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1;
 	int ret;
 
···
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct camelot_pcm *cam = &cam_pcm_data[rtd->cpu_dai->id];
+	struct camelot_pcm *cam = &cam_pcm_data[asoc_rtd_to_cpu(rtd, 0)->id];
 
 	pr_debug("PCM data: addr 0x%08lx len %d\n",
 		 (u32)runtime->dma_addr, runtime->dma_bytes);
···
 			   struct snd_pcm_substream *substream, int cmd)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct camelot_pcm *cam = &cam_pcm_data[rtd->cpu_dai->id];
+	struct camelot_pcm *cam = &cam_pcm_data[asoc_rtd_to_cpu(rtd, 0)->id];
 	int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1;
 
 	switch (cmd) {
···
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct camelot_pcm *cam = &cam_pcm_data[rtd->cpu_dai->id];
+	struct camelot_pcm *cam = &cam_pcm_data[asoc_rtd_to_cpu(rtd, 0)->id];
 	int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1;
 	unsigned long pos;
 

sound/soc/sh/fsi.c

···
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 
-	return  rtd->cpu_dai;
+	return  asoc_rtd_to_cpu(rtd, 0);
 }
 
 static struct fsi_priv *fsi_get_priv_frm_dai(struct snd_soc_dai *dai)
···
 	if (!master)
 		return -ENOMEM;
 
-	master->base = devm_ioremap(&pdev->dev,
-					    res->start, resource_size(res));
+	master->base = devm_ioremap(&pdev->dev, res->start, resource_size(res));
 	if (!master->base) {
 		dev_err(&pdev->dev, "Unable to ioremap FSI registers.\n");
 		return -ENXIO;

sound/soc/sh/migor.c

···
 			   struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	int ret;
 	unsigned int rate = params_rate(params);
 
···
 	clk_set_rate(&siumckb_clk, codec_freq);
 	dev_dbg(codec_dai->dev, "%s: configure %luHz\n", __func__, codec_freq);
 
-	ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, SIU_CLKB_EXT,
+	ret = snd_soc_dai_set_sysclk(asoc_rtd_to_cpu(rtd, 0), SIU_CLKB_EXT,
 				     codec_freq / 2, SND_SOC_CLOCK_IN);
 
 	if (!ret)
···
 static int migor_hw_free(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 
 	if (use_count) {
 		use_count--;

sound/soc/sh/rcar/core.c

···
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 
-	return  rtd->cpu_dai;
+	return  asoc_rtd_to_cpu(rtd, 0);
 }
 
 static

sound/soc/soc-compress.c

···
 	int playback = 0, capture = 0;
 	int i;
 
-	if (rtd->num_codecs > 1) {
+	if (rtd->num_cpus > 1 ||
+	    rtd->num_codecs > 1) {
 		dev_err(rtd->card->dev,
-			"Compress ASoC: Multicodec not supported\n");
+			"Compress ASoC: Multi CPU/Codec not supported\n");
 		return -EINVAL;
 	}
 

sound/soc/soc-core.c

···
 void snd_soc_close_delayed_work(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	int playback = SNDRV_PCM_STREAM_PLAYBACK;
 
 	mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass);
 
 	dev_dbg(rtd->dev,
 		"ASoC: pop wq checking: %s status: %s waiting: %s\n",
 		codec_dai->driver->playback.stream_name,
-		codec_dai->playback_active ? "active" : "inactive",
+		codec_dai->stream_active[playback] ? "active" : "inactive",
 		rtd->pop_wait ? "yes" : "no");
 
 	/* are we waiting on this codec DAI stream */
 	if (rtd->pop_wait == 1) {
 		rtd->pop_wait = 0;
-		snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK,
+		snd_soc_dapm_stream_event(rtd, playback,
 					  SND_SOC_DAPM_STREAM_STOP);
 	}
 
···
 	struct snd_soc_component *component;
 	struct device *dev;
 	int ret;
+	int stream;
 
 	/*
 	 * for rtd->dev
···
 
 	rtd->dev = dev;
 	INIT_LIST_HEAD(&rtd->list);
-	INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_PLAYBACK].be_clients);
-	INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_CAPTURE].be_clients);
-	INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_PLAYBACK].fe_clients);
-	INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_CAPTURE].fe_clients);
+	for_each_pcm_streams(stream) {
+		INIT_LIST_HEAD(&rtd->dpcm[stream].be_clients);
+		INIT_LIST_HEAD(&rtd->dpcm[stream].fe_clients);
+	}
 	dev_set_drvdata(dev, rtd);
 	INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work);
 
 	/*
-	 * for rtd->codec_dais
+	 * for rtd->dais
 	 */
-	rtd->codec_dais = devm_kcalloc(dev, dai_link->num_codecs,
+	rtd->dais = devm_kcalloc(dev, dai_link->num_cpus + dai_link->num_codecs,
 					sizeof(struct snd_soc_dai *),
 					GFP_KERNEL);
-	if (!rtd->codec_dais)
+	if (!rtd->dais)
 		goto free_rtd;
+
+	/*
+	 * dais = [][][][][][][][][][][][][][][][][][]
+	 *	  ^cpu_dais         ^codec_dais
+	 *	  |--- num_cpus ---|--- num_codecs --|
+	 */
+	rtd->cpu_dais	= &rtd->dais[0];
+	rtd->codec_dais	= &rtd->dais[dai_link->num_cpus];
 
 	/*
 	 * rtd remaining settings
···
 	struct snd_soc_card *card = dev_get_drvdata(dev);
 	struct snd_soc_component *component;
 	struct snd_soc_pcm_runtime *rtd;
+	int playback = SNDRV_PCM_STREAM_PLAYBACK;
 	int i;
 
 	/* If the card is not initialized yet there is nothing to do */
···
 		if (rtd->dai_link->ignore_suspend)
 			continue;
 
-		for_each_rtd_codec_dai(rtd, i, dai) {
-			if (dai->playback_active)
-				snd_soc_dai_digital_mute(dai, 1,
-						SNDRV_PCM_STREAM_PLAYBACK);
+		for_each_rtd_codec_dais(rtd, i, dai) {
+			if (dai->stream_active[playback])
+				snd_soc_dai_digital_mute(dai, 1, playback);
 		}
 	}
 
···
 	snd_soc_flush_all_delayed_work(card);
 
 	for_each_card_rtds(card, rtd) {
+		int stream;
 
 		if (rtd->dai_link->ignore_suspend)
 			continue;
 
-		snd_soc_dapm_stream_event(rtd,
-					  SNDRV_PCM_STREAM_PLAYBACK,
-					  SND_SOC_DAPM_STREAM_SUSPEND);
-
-		snd_soc_dapm_stream_event(rtd,
-					  SNDRV_PCM_STREAM_CAPTURE,
-					  SND_SOC_DAPM_STREAM_SUSPEND);
+		for_each_pcm_streams(stream)
+			snd_soc_dapm_stream_event(rtd, stream,
+						  SND_SOC_DAPM_STREAM_SUSPEND);
 	}
 
 	/* Recheck all endpoints too, their state is affected by suspend */
···
 	}
 
 	for_each_card_rtds(card, rtd) {
+		int stream;
 
 		if (rtd->dai_link->ignore_suspend)
 			continue;
 
-		snd_soc_dapm_stream_event(rtd,
-					  SNDRV_PCM_STREAM_PLAYBACK,
-					  SND_SOC_DAPM_STREAM_RESUME);
-
-		snd_soc_dapm_stream_event(rtd,
-					  SNDRV_PCM_STREAM_CAPTURE,
-					  SND_SOC_DAPM_STREAM_RESUME);
+		for_each_pcm_streams(stream)
+			snd_soc_dapm_stream_event(rtd, stream,
+						  SND_SOC_DAPM_STREAM_RESUME);
 	}
 
 	/* unmute any active DACs */
 	for_each_card_rtds(card, rtd) {
 		struct snd_soc_dai *dai;
+		int playback = SNDRV_PCM_STREAM_PLAYBACK;
 
 		if (rtd->dai_link->ignore_suspend)
 			continue;
 
-		for_each_rtd_codec_dai(rtd, i, dai) {
-			if (dai->playback_active)
-				snd_soc_dai_digital_mute(dai, 0,
-						SNDRV_PCM_STREAM_PLAYBACK);
+		for_each_rtd_codec_dais(rtd, i, dai) {
+			if (dai->stream_active[playback])
+				snd_soc_dai_digital_mute(dai, 0, playback);
 		}
 	}
 
···
 				     struct snd_soc_dai_link *link)
 {
 	int i;
-	struct snd_soc_dai_link_component *codec, *platform;
+	struct snd_soc_dai_link_component *cpu, *codec, *platform;
 
 	for_each_link_codecs(link, i, codec) {
 		/*
···
 			return -EPROBE_DEFER;
 	}
 
-	/* FIXME */
-	if (link->num_cpus > 1) {
-		dev_err(card->dev,
-			"ASoC: multi cpu is not yet supported %s\n",
-			link->name);
-		return -EINVAL;
-	}
+	for_each_link_cpus(link, i, cpu) {
+		/*
+		 * CPU device may be specified by either name or OF node, but
+		 * can be left unspecified, and will be matched based on DAI
+		 * name alone..
+		 */
+		if (cpu->name && cpu->of_node) {
+			dev_err(card->dev,
+				"ASoC: Neither/both cpu name/of_node are set for %s\n",
+				link->name);
+			return -EINVAL;
+		}
 
-	/*
-	 * CPU device may be specified by either name or OF node, but
-	 * can be left unspecified, and will be matched based on DAI
-	 * name alone..
-	 */
-	if (link->cpus->name && link->cpus->of_node) {
-		dev_err(card->dev,
-			"ASoC: Neither/both cpu name/of_node are set for %s\n",
-			link->name);
-		return -EINVAL;
-	}
+		/*
+		 * Defer card registration if cpu dai component is not added to
+		 * component list.
+		 */
+		if ((cpu->of_node || cpu->name) &&
+		    !soc_find_component(cpu))
+			return -EPROBE_DEFER;
 
-	/*
-	 * Defer card registration if cpu dai component is not added to
-	 * component list.
-	 */
-	if ((link->cpus->of_node || link->cpus->name) &&
-	    !soc_find_component(link->cpus))
-		return -EPROBE_DEFER;
-
-	/*
-	 * At least one of CPU DAI name or CPU device name/node must be
-	 * specified
-	 */
-	if (!link->cpus->dai_name &&
-	    !(link->cpus->name || link->cpus->of_node)) {
-		dev_err(card->dev,
-			"ASoC: Neither cpu_dai_name nor cpu_name/of_node are set for %s\n",
-			link->name);
-		return -EINVAL;
+		/*
+		 * At least one of CPU DAI name or CPU device name/node must be
+		 * specified
+		 */
+		if (!cpu->dai_name &&
+		    !(cpu->name || cpu->of_node)) {
+			dev_err(card->dev,
+				"ASoC: Neither cpu_dai_name nor cpu_name/of_node are set for %s\n",
+				link->name);
+			return -EINVAL;
+		}
 	}
 
 	return 0;
···
 			    struct snd_soc_dai_link *dai_link)
 {
 	struct snd_soc_pcm_runtime *rtd;
-	struct snd_soc_dai_link_component *codec, *platform;
+	struct snd_soc_dai_link_component *codec, *platform, *cpu;
 	struct snd_soc_component *component;
 	int i, ret;
 
···
 	if (!rtd)
 		return -ENOMEM;
 
-	/* FIXME: we need multi CPU support in the future */
-	rtd->cpu_dai = snd_soc_find_dai(dai_link->cpus);
-	if (!rtd->cpu_dai) {
-		dev_info(card->dev, "ASoC: CPU DAI %s not registered\n",
-			 dai_link->cpus->dai_name);
-		goto _err_defer;
+	rtd->num_cpus = dai_link->num_cpus;
+	for_each_link_cpus(dai_link, i, cpu) {
+		rtd->cpu_dais[i] = snd_soc_find_dai(cpu);
+		if (!rtd->cpu_dais[i]) {
+			dev_info(card->dev, "ASoC: CPU DAI %s not registered\n",
+				 cpu->dai_name);
+			goto _err_defer;
+		}
+		snd_soc_rtd_add_component(rtd, rtd->cpu_dais[i]->component);
 	}
-	snd_soc_rtd_add_component(rtd, rtd->cpu_dai->component);
+
+	/* Single cpu links expect cpu and cpu_dai in runtime data */
+	rtd->cpu_dai = rtd->cpu_dais[0];
 
 	/* Find CODEC from registered CODECs */
 	rtd->num_codecs = dai_link->num_codecs;
···
 }
 EXPORT_SYMBOL_GPL(snd_soc_add_pcm_runtime);
 
-static int soc_dai_pcm_new(struct snd_soc_dai **dais, int num_dais,
-			   struct snd_soc_pcm_runtime *rtd)
+static int soc_dai_pcm_new(struct snd_soc_pcm_runtime *rtd)
 {
+	struct snd_soc_dai *dai;
 	int i, ret = 0;
 
-	for (i = 0; i < num_dais; ++i) {
-		struct snd_soc_dai_driver *drv = dais[i]->driver;
+	for_each_rtd_dais(rtd, i, dai) {
+		struct snd_soc_dai_driver *drv = dai->driver;
 
 		if (drv->pcm_new)
-			ret = drv->pcm_new(rtd, dais[i]);
+			ret = drv->pcm_new(rtd, dai);
 		if (ret < 0) {
-			dev_err(dais[i]->dev,
+			dev_err(dai->dev,
 				"ASoC: Failed to bind %s with pcm device\n",
-				dais[i]->name);
+				dai->name);
 			return ret;
 		}
 	}
···
 			dai_link->stream_name, ret);
 		return ret;
 	}
-	ret = soc_dai_pcm_new(&cpu_dai, 1, rtd);
-	if (ret < 0)
-		return ret;
-	ret = soc_dai_pcm_new(rtd->codec_dais,
-			      rtd->num_codecs, rtd);
-	return ret;
+
+	return soc_dai_pcm_new(rtd);
 }
 
 static void soc_set_name_prefix(struct snd_soc_card *card,
···
 	ret = snd_soc_dapm_add_routes(dapm,
 				      component->driver->dapm_routes,
 				      component->driver->num_dapm_routes);
-	if (ret < 0)
-		goto err_probe;
+	if (ret < 0) {
+		if (card->disable_route_checks) {
+			dev_info(card->dev,
+				 "%s: disable_route_checks set, ignoring errors on add_routes\n",
+				 __func__);
+		} else {
+			dev_err(card->dev,
+				"%s: snd_soc_dapm_add_routes failed: %d\n",
+				__func__, ret);
+			goto err_probe;
+		}
+	}
 
 	/* see for_each_card_components */
 	list_add(&component->card_list, &card->component_dev_list);
···
 static void soc_remove_link_dais(struct snd_soc_card *card)
 {
 	int i;
-	struct snd_soc_dai *codec_dai;
+	struct snd_soc_dai *dai;
 	struct snd_soc_pcm_runtime *rtd;
 	int order;
 
 	for_each_comp_order(order) {
 		for_each_card_rtds(card, rtd) {
-			/* remove the CODEC DAI */
-			for_each_rtd_codec_dai(rtd, i, codec_dai)
-				soc_remove_dai(codec_dai, order);
-
-			soc_remove_dai(rtd->cpu_dai, order);
+			/* remove DAIs */
+			for_each_rtd_dais(rtd, i, dai)
+				soc_remove_dai(dai, order);
 		}
 	}
 }
 
 static int soc_probe_link_dais(struct snd_soc_card *card)
 {
-	struct snd_soc_dai *codec_dai;
+	struct snd_soc_dai *dai;
 	struct snd_soc_pcm_runtime *rtd;
 	int i, order, ret;
 
···
 				"ASoC: probe %s dai link %d late %d\n",
 				card->name, rtd->num, order);
 
-			ret = soc_probe_dai(rtd->cpu_dai, order);
-			if (ret)
-				return ret;
-
-			/* probe the CODEC DAI */
-			for_each_rtd_codec_dai(rtd, i, codec_dai) {
-				ret = soc_probe_dai(codec_dai, order);
+			/* probe the CPU DAI */
+			for_each_rtd_dais(rtd, i, dai) {
+				ret = soc_probe_dai(dai, order);
 				if (ret)
 					return ret;
 			}
···
 int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd,
 	unsigned int dai_fmt)
 {
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *cpu_dai;
 	struct snd_soc_dai *codec_dai;
+	unsigned int inv_dai_fmt;
 	unsigned int i;
 	int ret;
 
-	for_each_rtd_codec_dai(rtd, i, codec_dai) {
+	for_each_rtd_codec_dais(rtd, i, codec_dai) {
 		ret = snd_soc_dai_set_fmt(codec_dai, dai_fmt);
 		if (ret != 0 && ret != -ENOTSUPP) {
 			dev_warn(codec_dai->dev,
···
 	 * Flip the polarity for the "CPU" end of a CODEC<->CODEC link
 	 * the component which has non_legacy_dai_naming is Codec
 	 */
-	if (cpu_dai->component->driver->non_legacy_dai_naming) {
-		unsigned int inv_dai_fmt;
-
-		inv_dai_fmt = dai_fmt & ~SND_SOC_DAIFMT_MASTER_MASK;
-		switch (dai_fmt & SND_SOC_DAIFMT_MASTER_MASK) {
-		case SND_SOC_DAIFMT_CBM_CFM:
-			inv_dai_fmt |= SND_SOC_DAIFMT_CBS_CFS;
-			break;
-		case SND_SOC_DAIFMT_CBM_CFS:
-			inv_dai_fmt |= SND_SOC_DAIFMT_CBS_CFM;
-			break;
-		case SND_SOC_DAIFMT_CBS_CFM:
-			inv_dai_fmt |= SND_SOC_DAIFMT_CBM_CFS;
-			break;
-		case SND_SOC_DAIFMT_CBS_CFS:
-			inv_dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
-			break;
-		}
-
-		dai_fmt = inv_dai_fmt;
+	inv_dai_fmt = dai_fmt & ~SND_SOC_DAIFMT_MASTER_MASK;
+	switch (dai_fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		inv_dai_fmt |= SND_SOC_DAIFMT_CBS_CFS;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFS:
+		inv_dai_fmt |= SND_SOC_DAIFMT_CBS_CFM;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFM:
+		inv_dai_fmt |= SND_SOC_DAIFMT_CBM_CFS;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		inv_dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
+		break;
 	}
+	for_each_rtd_cpu_dais(rtd, i, cpu_dai) {
+		unsigned int fmt = dai_fmt;
 
-	ret = snd_soc_dai_set_fmt(cpu_dai, dai_fmt);
-	if (ret != 0 && ret != -ENOTSUPP) {
-		dev_warn(cpu_dai->dev,
-			 "ASoC: Failed to set DAI format: %d\n", ret);
-		return ret;
+		if (cpu_dai->component->driver->non_legacy_dai_naming)
+			fmt = inv_dai_fmt;
+
+		ret = snd_soc_dai_set_fmt(cpu_dai, fmt);
+		if (ret != 0 && ret != -ENOTSUPP) {
+			dev_warn(cpu_dai->dev,
+				 "ASoC: Failed to set DAI format: %d\n", ret);
+			return ret;
+		}
 	}
 
 	return 0;
···
 
 	ret = snd_soc_dapm_add_routes(&card->dapm, card->dapm_routes,
 				      card->num_dapm_routes);
-	if (ret < 0)
-		goto probe_end;
+	if (ret < 0) {
+		if (card->disable_route_checks) {
+			dev_info(card->dev,
+				 "%s: disable_route_checks set, ignoring errors on add_routes\n",
+				 __func__);
+		} else {
+			dev_err(card->dev,
+				 "%s: snd_soc_dapm_add_routes failed: %d\n",
+				 __func__, ret);
+			goto probe_end;
+		}
+	}
 
 	ret = snd_soc_dapm_add_routes(&card->dapm, card->of_dapm_routes,
 				      card->num_of_dapm_routes);
···
 			*dai_name = dai->driver->name;
 			if (!*dai_name)
 				*dai_name = pos->name;
+		} else if (ret) {
+			/*
+			 * if another error than ENOTSUPP is returned go on and
+			 * check if another component is provided with the same
+			 * node. This may happen if a device provides several
+			 * components
+			 */
+			continue;
 		}
 
 		break;

sound/soc/soc-dai.c

···
 {
 	int ret = 0;
 
-	if (dai->driver->ops->startup)
+	if (!dai->started &&
+	    dai->driver->ops->startup)
 		ret = dai->driver->ops->startup(substream, dai);
+
+	if (ret == 0)
+		dai->started = 1;
 
 	return ret;
 }
···
 void snd_soc_dai_shutdown(struct snd_soc_dai *dai,
 			 struct snd_pcm_substream *substream)
 {
-	if (dai->driver->ops->shutdown)
+	if (dai->started &&
+	    dai->driver->ops->shutdown)
 		dai->driver->ops->shutdown(substream, dai);
+
+	dai->started = 0;
 }
 
 int snd_soc_dai_prepare(struct snd_soc_dai *dai,
···
  */
 bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int dir)
 {
-	struct snd_soc_pcm_stream *stream;
-
-	if (dir == SNDRV_PCM_STREAM_PLAYBACK)
-		stream = &dai->driver->playback;
-	else
-		stream = &dai->driver->capture;
+	struct snd_soc_pcm_stream *stream = snd_soc_dai_get_pcm_stream(dai, dir);
 
 	/* If the codec specifies any channels at all, it supports the stream */
 	return stream->channels_min;

sound/soc/soc-dapm.c

···
 
 	mutex_lock(&card->dapm_mutex);
 
-	list_for_each_entry(w, &card->widgets, list) {
+	for_each_card_widgets(card, w) {
 		if (w->is_ep) {
 			dapm_mark_dirty(w, "Rechecking endpoints");
 			if (w->is_ep & SND_SOC_DAPM_EP_SINK)
···
 
 	memset(&card->dapm_stats, 0, sizeof(card->dapm_stats));
 
-	list_for_each_entry(w, &card->widgets, list) {
+	for_each_card_widgets(card, w) {
 		w->new_power = w->power;
 		w->power_checked = false;
 	}
···
 
 	*kcontrol = NULL;
 
-	list_for_each_entry(w, &dapm->card->widgets, list) {
+	for_each_card_widgets(dapm->card, w) {
 		if (w == kcontrolw || w->dapm != kcontrolw->dapm)
 			continue;
 		for (i = 0; i < w->num_kcontrols; i++) {
···
 	}
 }
 
+static void dapm_widget_list_free(struct snd_soc_dapm_widget_list **list)
+{
+	kfree(*list);
+}
+
 static int dapm_widget_list_create(struct snd_soc_dapm_widget_list **list,
 	struct list_head *widgets)
 {
···
 	mutex_unlock(&card->dapm_mutex);
 
 	return paths;
+}
+
+void snd_soc_dapm_dai_free_widgets(struct snd_soc_dapm_widget_list **list)
+{
+	dapm_widget_list_free(list);
 }
 
 /*
···
 					i, cur_subseq);
 	}
 
-	list_for_each_entry(d, &card->dapm_list, list) {
+	for_each_card_dapms(card, d)
 		soc_dapm_async_complete(d);
-	}
 }
 
 static void dapm_widget_update(struct snd_soc_card *card)
···
 
 	wlist = dapm_kcontrol_get_wlist(update->kcontrol);
 
-	for (wi = 0; wi < wlist->num_widgets; wi++) {
-		w = wlist->widgets[wi];
-
+	for_each_dapm_widgets(wlist, wi, w) {
 		if (w->event && (w->event_flags & SND_SOC_DAPM_PRE_REG)) {
 			ret = w->event(w, update->kcontrol, SND_SOC_DAPM_PRE_REG);
 			if (ret != 0)
···
 				w->name, ret);
 	}
 
-	for (wi = 0; wi < wlist->num_widgets; wi++) {
-		w = wlist->widgets[wi];
-
+	for_each_dapm_widgets(wlist, wi, w) {
 		if (w->event && (w->event_flags & SND_SOC_DAPM_POST_REG)) {
 			ret = w->event(w, update->kcontrol, SND_SOC_DAPM_POST_REG);
 			if (ret != 0)
···
 
 	trace_snd_soc_dapm_start(card);
 
-	list_for_each_entry(d, &card->dapm_list, list) {
+	for_each_card_dapms(card, d) {
 		if (dapm_idle_bias_off(d))
 			d->target_bias_level = SND_SOC_BIAS_OFF;
 		else
···
 		dapm_power_one_widget(w, &up_list, &down_list);
 	}
 
-	list_for_each_entry(w, &card->widgets, list) {
+	for_each_card_widgets(card, w) {
 		switch (w->id) {
 		case snd_soc_dapm_pre:
 		case snd_soc_dapm_post:
···
 	 * they're not ground referenced.
 	 */
 	bias = SND_SOC_BIAS_OFF;
-	list_for_each_entry(d, &card->dapm_list, list)
+	for_each_card_dapms(card, d)
 		if (d->target_bias_level > bias)
 			bias = d->target_bias_level;
-	list_for_each_entry(d, &card->dapm_list, list)
+	for_each_card_dapms(card, d)
 		if (!dapm_idle_bias_off(d))
 			d->target_bias_level = bias;
 
···
 	/* Run card bias changes at first */
 	dapm_pre_sequence_async(&card->dapm, 0);
 	/* Run other bias changes in parallel */
-	list_for_each_entry(d, &card->dapm_list, list) {
+	for_each_card_dapms(card, d) {
 		if (d != &card->dapm && d->bias_level != d->target_bias_level)
 			async_schedule_domain(dapm_pre_sequence_async, d,
 						&async_domain);
···
 	dapm_seq_run(card, &up_list, event, true);
 
 	/* Run all the bias changes in parallel */
-	list_for_each_entry(d, &card->dapm_list, list) {
+	for_each_card_dapms(card, d) {
 		if (d != &card->dapm && d->bias_level != d->target_bias_level)
 			async_schedule_domain(dapm_post_sequence_async, d,
 						&async_domain);
···
 	dapm_post_sequence_async(&card->dapm, 0);
 
 	/* do we need to notify any clients that DAPM event is complete */
-	list_for_each_entry(d, &card->dapm_list, list) {
+	for_each_card_dapms(card, d) {
 		if (!d->component)
 			continue;
 
···
 	card->update = NULL;
 	mutex_unlock(&card->dapm_mutex);
 	if (ret > 0)
-		soc_dpcm_runtime_update(card);
+		snd_soc_dpcm_runtime_update(card);
 	return ret;
 }
 EXPORT_SYMBOL_GPL(snd_soc_dapm_mux_update_power);
···
 	card->update = NULL;
 	mutex_unlock(&card->dapm_mutex);
 	if (ret > 0)
-		soc_dpcm_runtime_update(card);
+		snd_soc_dpcm_runtime_update(card);
 	return ret;
 }
 EXPORT_SYMBOL_GPL(snd_soc_dapm_mixer_update_power);
···
 	if (!cmpnt->card)
 		return 0;
 
-	list_for_each_entry(w, &cmpnt->card->widgets, list) {
+	for_each_card_widgets(cmpnt->card, w) {
 		if (w->dapm != dapm)
 			continue;
 
···
 
 	mutex_lock(&rtd->card->dapm_mutex);
 
-	for_each_rtd_codec_dai(rtd, i, codec_dai) {
+	for_each_rtd_codec_dais(rtd, i, codec_dai) {
 		struct snd_soc_component *cmpnt = codec_dai->component;
 
 		count += dapm_widget_show_component(cmpnt, buf + count);
···
 {
 	struct snd_soc_dapm_widget *w, *next_w;
 
-	list_for_each_entry_safe(w, next_w, &dapm->card->widgets, list) {
+	for_each_card_widgets_safe(dapm->card, w, next_w) {
 		if (w->dapm != dapm)
 			continue;
 		snd_soc_dapm_free_widget(w);
···
 	struct snd_soc_dapm_widget *w;
 	struct snd_soc_dapm_widget *fallback = NULL;
 
-	list_for_each_entry(w, &dapm->card->widgets, list) {
+	for_each_card_widgets(dapm->card, w) {
 		if (!strcmp(w->name, pin)) {
 			if (w->dapm == dapm)
 				return w;
···
 	struct snd_soc_dapm_widget *w;
 	int ret;
 
-	if (dir == SNDRV_PCM_STREAM_PLAYBACK)
-		w = dai->playback_widget;
-	else
-		w = dai->capture_widget;
+	w = snd_soc_dai_get_widget(dai, dir);
 
 	if (!w)
 		return 0;
···
 	 * find src and dest widgets over all widgets but favor a widget from
 	 * current DAPM context
 	 */
-	list_for_each_entry(w, &dapm->card->widgets, list) {
+	for_each_card_widgets(dapm->card, w) {
 		if (!wsink && !(strcmp(w->name, sink))) {
 			wtsink = w;
 			if (w->dapm == dapm) {
···
 
 	mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT);
 
-	list_for_each_entry(w, &card->widgets, list)
+	for_each_card_widgets(card, w)
 	{
 		if (w->new)
 			continue;
···
 	mutex_unlock(&card->dapm_mutex);
 
 	if (ret > 0)
-		soc_dpcm_runtime_update(card);
+		snd_soc_dpcm_runtime_update(card);
 
 	return change;
 }
···
 	mutex_unlock(&card->dapm_mutex);
 
 	if (ret > 0)
-		soc_dpcm_runtime_update(card);
+		snd_soc_dpcm_runtime_update(card);
 
 	return change;
 }
···
 			ret = PTR_ERR(w->pinctrl);
 			goto request_failed;
 		}
+
+		/* set to sleep_state when initializing */
+		dapm_pinctrl_event(w, NULL, SND_SOC_DAPM_POST_PMD);
 		break;
 	case snd_soc_dapm_clock_supply:
 		w->clk = devm_clk_get(dapm->dev, w->name);
···
 	w->dapm = dapm;
 	INIT_LIST_HEAD(&w->list);
 	INIT_LIST_HEAD(&w->dirty);
+	/* see for_each_card_widgets */
 	list_add_tail(&w->list, &dapm->card->widgets);
 
 	snd_soc_dapm_for_each_direction(dir) {
···
 	struct snd_soc_dai *dai;
 
 	/* For each DAI widget... */
-	list_for_each_entry(dai_w, &card->widgets, list) {
+	for_each_card_widgets(card, dai_w) {
 		switch (dai_w->id) {
 		case snd_soc_dapm_dai_in:
 		case snd_soc_dapm_dai_out:
···
 		dai = dai_w->priv;
 
 		/* ...find all widgets with the same stream and link them */
-		list_for_each_entry(w, &card->widgets, list) {
+		for_each_card_widgets(card, w) {
 			if (w->dapm != dai_w->dapm)
 				continue;
 
···
 	return 0;
 }
 
-static void dapm_connect_dai_link_widgets(struct snd_soc_card *card,
-					  struct snd_soc_pcm_runtime *rtd)
+static void dapm_add_valid_dai_widget(struct snd_soc_card *card,
+				      struct snd_soc_pcm_runtime *rtd,
+				      struct snd_soc_dai *codec_dai,
+				      struct snd_soc_dai *cpu_dai)
 {
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	struct snd_soc_dai *codec_dai;
 	struct snd_soc_dapm_widget *playback = NULL, *capture = NULL;
 	struct snd_soc_dapm_widget *codec, *playback_cpu, *capture_cpu;
 	struct snd_pcm_substream *substream;
 	struct snd_pcm_str *streams = rtd->pcm->streams;
-	int i;
 
 	if (rtd->dai_link->params) {
 		playback_cpu = cpu_dai->capture_widget;
···
 		capture_cpu = capture;
 	}
 
-	for_each_rtd_codec_dai(rtd, i, codec_dai) {
-		/* connect BE DAI playback if widgets are valid */
-		codec = codec_dai->playback_widget;
+	/* connect BE DAI playback if widgets are valid */
+	codec = codec_dai->playback_widget;
 
-		if (playback_cpu && codec) {
-			if (!playback) {
-				substream = streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
-				playback = snd_soc_dapm_new_dai(card, substream,
-								"playback");
-				if (IS_ERR(playback)) {
-					dev_err(rtd->dev,
-						"ASoC: Failed to create DAI %s: %ld\n",
-						codec_dai->name,
-						PTR_ERR(playback));
-					continue;
-				}
-
-				snd_soc_dapm_add_path(&card->dapm, playback_cpu,
-						      playback, NULL, NULL);
+	if (playback_cpu && codec) {
+		if (!playback) {
+			substream = streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
+			playback = snd_soc_dapm_new_dai(card, substream,
+							"playback");
+			if (IS_ERR(playback)) {
+				dev_err(rtd->dev,
+					"ASoC: Failed to create DAI %s: %ld\n",
+					codec_dai->name,
+					PTR_ERR(playback));
+				goto capture;
 			}
 
-			dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n",
-				cpu_dai->component->name, playback_cpu->name,
-				codec_dai->component->name, codec->name);
-
-			snd_soc_dapm_add_path(&card->dapm, playback, codec,
-					      NULL, NULL);
+			snd_soc_dapm_add_path(&card->dapm, playback_cpu,
+					      playback, NULL, NULL);
 		}
+
+		dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n",
+			cpu_dai->component->name, playback_cpu->name,
+			codec_dai->component->name, codec->name);
+
+		snd_soc_dapm_add_path(&card->dapm, playback, codec,
+				      NULL, NULL);
 	}
 
-	for_each_rtd_codec_dai(rtd, i, codec_dai) {
-		/* connect BE DAI capture if widgets are valid */
-		codec = codec_dai->capture_widget;
+capture:
+	/* connect BE DAI capture if widgets are valid */
+	codec = codec_dai->capture_widget;
 
-		if (codec && capture_cpu) {
-			if (!capture) {
-				substream = streams[SNDRV_PCM_STREAM_CAPTURE].substream;
-				capture = snd_soc_dapm_new_dai(card, substream,
-							       "capture");
-				if (IS_ERR(capture)) {
-					dev_err(rtd->dev,
-						"ASoC: Failed to create DAI %s: %ld\n",
-						codec_dai->name,
-						PTR_ERR(capture));
-					continue;
-				}
-
-				snd_soc_dapm_add_path(&card->dapm, capture,
-						      capture_cpu, NULL, NULL);
+	if (codec && capture_cpu) {
+		if (!capture) {
+			substream = streams[SNDRV_PCM_STREAM_CAPTURE].substream;
+			capture = snd_soc_dapm_new_dai(card, substream,
+						       "capture");
+			if (IS_ERR(capture)) {
+				dev_err(rtd->dev,
+					"ASoC: Failed to create DAI %s: %ld\n",
+					codec_dai->name,
+					PTR_ERR(capture));
+				return;
 			}
 
-			dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n",
-				codec_dai->component->name, codec->name,
-				cpu_dai->component->name, capture_cpu->name);
-
-			snd_soc_dapm_add_path(&card->dapm, codec, capture,
-					      NULL, NULL);
+			snd_soc_dapm_add_path(&card->dapm, capture,
+					      capture_cpu, NULL, NULL);
 		}
+
+		dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n",
+			codec_dai->component->name, codec->name,
+			cpu_dai->component->name, capture_cpu->name);
+
+		snd_soc_dapm_add_path(&card->dapm, codec, capture,
+				      NULL, NULL);
 	}
+}
+
+static void dapm_connect_dai_link_widgets(struct snd_soc_card *card,
+					  struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_dai *codec_dai;
+	int i;
+
+	if (rtd->num_cpus == 1) {
+		for_each_rtd_codec_dais(rtd, i, codec_dai)
+			dapm_add_valid_dai_widget(card, rtd, codec_dai,
+						  rtd->cpu_dais[0]);
+	} else if (rtd->num_codecs == rtd->num_cpus) {
+		for_each_rtd_codec_dais(rtd, i, codec_dai)
+			dapm_add_valid_dai_widget(card, rtd, codec_dai,
+						  rtd->cpu_dais[i]);
+	} else {
+		dev_err(card->dev,
+			"N cpus to M codecs link is not supported yet\n");
+	}
+
 }
 
 static void soc_dapm_dai_stream_event(struct snd_soc_dai *dai, int stream,
···
 	struct snd_soc_dapm_widget *w;
 	unsigned int ep;
 
-	if (stream == SNDRV_PCM_STREAM_PLAYBACK)
-		w = dai->playback_widget;
-	else
-		w = dai->capture_widget;
+	w = snd_soc_dai_get_widget(dai, stream);
 
 	if (w) {
 		dapm_mark_dirty(w, "stream event");
···
 static void soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream,
 	int event)
 {
-	struct snd_soc_dai *codec_dai;
+	struct snd_soc_dai *dai;
 	int i;
 
-	soc_dapm_dai_stream_event(rtd->cpu_dai, stream, event);
-	for_each_rtd_codec_dai(rtd, i, codec_dai)
-		soc_dapm_dai_stream_event(codec_dai, stream, event);
+	for_each_rtd_dais(rtd, i, dai)
+		soc_dapm_dai_stream_event(dai, stream, event);
 
 	dapm_power_widgets(rtd->card, event);
 }
···
 	}
 
 	INIT_LIST_HEAD(&dapm->list);
+	/* see for_each_card_dapms */
 	list_add(&dapm->list, &card->dapm_list);
 }
 EXPORT_SYMBOL_GPL(snd_soc_dapm_init);
···
 
 	mutex_lock(&card->dapm_mutex);
 
-	list_for_each_entry(w, &dapm->card->widgets, list) {
+	for_each_card_widgets(dapm->card, w) {
 		if (w->dapm != dapm)
 			continue;
 		if (w->power) {
···
 {
 	struct snd_soc_dapm_context *dapm;
 
-	list_for_each_entry(dapm, &card->dapm_list, list) {
+	for_each_card_dapms(card, dapm) {
 		if (dapm != &card->dapm) {
 			soc_dapm_shutdown_dapm(dapm);
 			if (dapm->bias_level == SND_SOC_BIAS_STANDBY)

sound/soc/soc-generic-dmaengine-pcm.c

···
 	struct snd_dmaengine_dai_dma_data *dma_data;
 	int ret;
 
+	if (rtd->num_cpus > 1) {
+		dev_err(rtd->dev,
+			"%s doesn't support Multi CPU yet\n", __func__);
+		return -EINVAL;
+	}
+
 	dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
 
 	ret = snd_hwparams_to_dma_slave_config(substream, params, slave_config);
···
 	struct dma_chan *chan = pcm->chan[substream->stream];
 	struct snd_dmaengine_dai_dma_data *dma_data;
 	struct snd_pcm_hardware hw;
+
+	if (rtd->num_cpus > 1) {
+		dev_err(rtd->dev,
+			"%s doesn't support Multi CPU yet\n", __func__);
+		return -EINVAL;
+	}
 
 	if (pcm->config && pcm->config->pcm_hardware)
 		return snd_soc_set_runtime_hwparams(substream,
···
 	struct snd_dmaengine_dai_dma_data *dma_data;
 	dma_filter_fn fn = NULL;
 
+	if (rtd->num_cpus > 1) {
+		dev_err(rtd->dev,
+			"%s doesn't support Multi CPU yet\n", __func__);
+		return NULL;
+	}
+
 	dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
 
 	if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) && pcm->chan[0])
···
 		max_buffer_size = SIZE_MAX;
 	}
 
-	for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; i++) {
+	for_each_pcm_streams(i) {
 		substream = rtd->pcm->streams[i].substream;
 		if (!substream)
 			continue;
···
 		dev = config->dma_dev;
 	}
 
-	for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE;
-	     i++) {
+	for_each_pcm_streams(i) {
 		if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX)
 			name = "rx-tx";
 		else
···
 {
 	unsigned int i;
 
-	for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE;
-	     i++) {
+	for_each_pcm_streams(i) {
 		if (!pcm->chan[i])
 			continue;
 		dma_release_channel(pcm->chan[i]);

sound/soc/soc-pcm.c

···
 
 #define DPCM_MAX_BE_USERS	8
 
+#ifdef CONFIG_DEBUG_FS
+static const char *dpcm_state_string(enum snd_soc_dpcm_state state)
+{
+	switch (state) {
+	case SND_SOC_DPCM_STATE_NEW:
+		return "new";
+	case SND_SOC_DPCM_STATE_OPEN:
+		return "open";
+	case SND_SOC_DPCM_STATE_HW_PARAMS:
+		return "hw_params";
+	case SND_SOC_DPCM_STATE_PREPARE:
+		return "prepare";
+	case SND_SOC_DPCM_STATE_START:
+		return "start";
+	case SND_SOC_DPCM_STATE_STOP:
+		return "stop";
+	case SND_SOC_DPCM_STATE_SUSPEND:
+		return "suspend";
+	case SND_SOC_DPCM_STATE_PAUSED:
+		return "paused";
+	case SND_SOC_DPCM_STATE_HW_FREE:
+		return "hw_free";
+	case SND_SOC_DPCM_STATE_CLOSE:
+		return "close";
+	}
+
+	return "unknown";
+}
+
+static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe,
+			       int stream, char *buf, size_t size)
+{
+	struct snd_pcm_hw_params *params = &fe->dpcm[stream].hw_params;
+	struct snd_soc_dpcm *dpcm;
+	ssize_t offset = 0;
+	unsigned long flags;
+
+	/* FE state */
+	offset += scnprintf(buf + offset, size - offset,
+			   "[%s - %s]\n", fe->dai_link->name,
+			   stream ? "Capture" : "Playback");
+
+	offset += scnprintf(buf + offset, size - offset, "State: %s\n",
+			   dpcm_state_string(fe->dpcm[stream].state));
+
+	if ((fe->dpcm[stream].state >= SND_SOC_DPCM_STATE_HW_PARAMS) &&
+	    (fe->dpcm[stream].state <= SND_SOC_DPCM_STATE_STOP))
+		offset += scnprintf(buf + offset, size - offset,
+				   "Hardware Params: "
+				   "Format = %s, Channels = %d, Rate = %d\n",
+				   snd_pcm_format_name(params_format(params)),
+				   params_channels(params),
+				   params_rate(params));
+
+	/* BEs state */
+	offset += scnprintf(buf + offset, size - offset, "Backends:\n");
+
+	if (list_empty(&fe->dpcm[stream].be_clients)) {
+		offset += scnprintf(buf + offset, size - offset,
+				   " No active DSP links\n");
+		goto out;
+	}
+
+	spin_lock_irqsave(&fe->card->dpcm_lock, flags);
+	for_each_dpcm_be(fe, stream, dpcm) {
+		struct snd_soc_pcm_runtime *be = dpcm->be;
+		params = &dpcm->hw_params;
+
+		offset += scnprintf(buf + offset, size - offset,
+				   "- %s\n", be->dai_link->name);
+
+		offset += scnprintf(buf + offset, size - offset,
+				   "   State: %s\n",
+				   dpcm_state_string(be->dpcm[stream].state));
+
+		if ((be->dpcm[stream].state >= SND_SOC_DPCM_STATE_HW_PARAMS) &&
+		    (be->dpcm[stream].state <= SND_SOC_DPCM_STATE_STOP))
+			offset += scnprintf(buf + offset, size - offset,
+					   "   Hardware Params: "
+					   "Format = %s, Channels = %d, Rate = %d\n",
+					   snd_pcm_format_name(params_format(params)),
+					   params_channels(params),
+					   params_rate(params));
+	}
+	spin_unlock_irqrestore(&fe->card->dpcm_lock, flags);
+out:
+	return offset;
+}
+
+static ssize_t dpcm_state_read_file(struct file *file, char __user *user_buf,
+				    size_t count, loff_t *ppos)
+{
+	struct snd_soc_pcm_runtime *fe = file->private_data;
+	ssize_t out_count = PAGE_SIZE, offset = 0, ret = 0;
+	int stream;
+	char *buf;
+
+	if (fe->num_cpus > 1) {
+		dev_err(fe->dev,
+			"%s doesn't support Multi CPU yet\n", __func__);
+		return -EINVAL;
+	}
+
+	buf = kmalloc(out_count, GFP_KERNEL);
+	if (!buf)
+		return -ENOMEM;
+
+	for_each_pcm_streams(stream)
+		if (snd_soc_dai_stream_valid(fe->cpu_dai, stream))
+			offset += dpcm_show_state(fe, stream,
+						  buf + offset,
+						  out_count - offset);
+
+	ret = simple_read_from_buffer(user_buf, count, ppos, buf, offset);
+
+	kfree(buf);
+	return ret;
+}
+
+static const struct file_operations dpcm_state_fops = {
+	.open = simple_open,
+	.read = dpcm_state_read_file,
+	.llseek = default_llseek,
+};
+
+void soc_dpcm_debugfs_add(struct snd_soc_pcm_runtime *rtd)
+{
+	if (!rtd->dai_link)
+		return;
+
+	if (!rtd->dai_link->dynamic)
+		return;
+
+	if (!rtd->card->debugfs_card_root)
+		return;
+
+	rtd->debugfs_dpcm_root = debugfs_create_dir(rtd->dai_link->name,
+						    rtd->card->debugfs_card_root);
+
+	debugfs_create_file("state", 0444, rtd->debugfs_dpcm_root,
+			    rtd, &dpcm_state_fops);
+}
+
+static void dpcm_create_debugfs_state(struct snd_soc_dpcm *dpcm, int stream)
+{
+	char *name;
+
+	name = kasprintf(GFP_KERNEL, "%s:%s", dpcm->be->dai_link->name,
+			 stream ? "capture" : "playback");
+	if (name) {
+		dpcm->debugfs_state = debugfs_create_dir(
+			name, dpcm->fe->debugfs_dpcm_root);
+		debugfs_create_u32("state", 0644, dpcm->debugfs_state,
+				   &dpcm->state);
+		kfree(name);
+	}
+}
+
+static void dpcm_remove_debugfs_state(struct snd_soc_dpcm *dpcm)
+{
+	debugfs_remove_recursive(dpcm->debugfs_state);
+}
+
+#else
+static inline void dpcm_create_debugfs_state(struct snd_soc_dpcm *dpcm,
+					     int stream)
+{
+}
+
+static inline void dpcm_remove_debugfs_state(struct snd_soc_dpcm *dpcm)
+{
+}
+#endif
+
 static int soc_rtd_startup(struct snd_soc_pcm_runtime *rtd,
 			   struct snd_pcm_substream *substream)
 {
···
 	return 0;
 }
 
+static void snd_soc_runtime_action(struct snd_soc_pcm_runtime *rtd,
+				   int stream, int action)
+{
+	struct snd_soc_dai *dai;
+	int i;
+
+	lockdep_assert_held(&rtd->card->pcm_mutex);
+
+	for_each_rtd_dais(rtd, i, dai) {
+		dai->stream_active[stream] += action;
+		dai->active += action;
+		dai->component->active += action;
+	}
+}
+
 /**
  * snd_soc_runtime_activate() - Increment active count for PCM runtime components
  * @rtd: ASoC PCM runtime that is activated
···
  */
 void snd_soc_runtime_activate(struct snd_soc_pcm_runtime *rtd, int stream)
 {
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	struct snd_soc_dai *codec_dai;
-	int i;
-
-	lockdep_assert_held(&rtd->card->pcm_mutex);
-
-	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		cpu_dai->playback_active++;
-		for_each_rtd_codec_dai(rtd, i, codec_dai)
-			codec_dai->playback_active++;
-	} else {
-		cpu_dai->capture_active++;
-		for_each_rtd_codec_dai(rtd, i, codec_dai)
-			codec_dai->capture_active++;
-	}
-
-	cpu_dai->active++;
-	cpu_dai->component->active++;
-	for_each_rtd_codec_dai(rtd, i, codec_dai) {
-		codec_dai->active++;
-		codec_dai->component->active++;
-	}
+	snd_soc_runtime_action(rtd, stream, 1);
 }
+EXPORT_SYMBOL_GPL(snd_soc_runtime_activate);
 
 /**
  * snd_soc_runtime_deactivate() - Decrement active count for PCM runtime components
···
  */
 void snd_soc_runtime_deactivate(struct snd_soc_pcm_runtime *rtd, int stream)
 {
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	struct snd_soc_dai *codec_dai;
-	int i;
-
-	lockdep_assert_held(&rtd->card->pcm_mutex);
-
-	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		cpu_dai->playback_active--;
-		for_each_rtd_codec_dai(rtd, i, codec_dai)
-			codec_dai->playback_active--;
-	} else {
-		cpu_dai->capture_active--;
-		for_each_rtd_codec_dai(rtd, i, codec_dai)
-			codec_dai->capture_active--;
-	}
-
-	cpu_dai->active--;
-	cpu_dai->component->active--;
-	for_each_rtd_codec_dai(rtd, i, codec_dai) {
-		codec_dai->component->active--;
-		codec_dai->active--;
-	}
+	snd_soc_runtime_action(rtd, stream, -1);
 }
+EXPORT_SYMBOL_GPL(snd_soc_runtime_deactivate);
 
 /**
  * snd_soc_runtime_ignore_pmdown_time() - Check whether to ignore the power down delay
···
 				struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	struct snd_soc_dai *codec_dai;
+	struct snd_soc_dai *dai;
+	struct snd_soc_dai *cpu_dai;
 	unsigned int rate, channels, sample_bits, symmetry, i;
 
 	rate = params_rate(params);
···
 	sample_bits = snd_pcm_format_physical_width(params_format(params));
 
 	/* reject unmatched parameters when applying symmetry */
-	symmetry = cpu_dai->driver->symmetric_rates ||
-		rtd->dai_link->symmetric_rates;
+	symmetry = rtd->dai_link->symmetric_rates;
 
-	for_each_rtd_codec_dai(rtd, i, codec_dai)
-		symmetry |= codec_dai->driver->symmetric_rates;
+	for_each_rtd_cpu_dais(rtd, i, dai)
+		symmetry |= dai->driver->symmetric_rates;
 
-	if (symmetry && cpu_dai->rate && cpu_dai->rate != rate) {
-		dev_err(rtd->dev, "ASoC: unmatched rate symmetry: %d - %d\n",
-				cpu_dai->rate, rate);
-		return -EINVAL;
+	if (symmetry) {
+		for_each_rtd_cpu_dais(rtd, i, cpu_dai) {
+			if (cpu_dai->rate && cpu_dai->rate != rate) {
+				dev_err(rtd->dev, "ASoC: unmatched rate symmetry: %d - %d\n",
+					cpu_dai->rate, rate);
+				return -EINVAL;
+			}
+		}
 	}
 
-	symmetry = cpu_dai->driver->symmetric_channels ||
-		rtd->dai_link->symmetric_channels;
+	symmetry = rtd->dai_link->symmetric_channels;
 
-	for_each_rtd_codec_dai(rtd, i, codec_dai)
-		symmetry |= codec_dai->driver->symmetric_channels;
+	for_each_rtd_dais(rtd, i, dai)
+		symmetry |= dai->driver->symmetric_channels;
 
-	if (symmetry && cpu_dai->channels && cpu_dai->channels != channels) {
-		dev_err(rtd->dev, "ASoC: unmatched channel symmetry: %d - %d\n",
-				cpu_dai->channels, channels);
-		return -EINVAL;
+	if (symmetry) {
+		for_each_rtd_cpu_dais(rtd, i, cpu_dai) {
+			if (cpu_dai->channels &&
+			    cpu_dai->channels != channels) {
+				dev_err(rtd->dev, "ASoC: unmatched channel symmetry: %d - %d\n",
+					cpu_dai->channels, channels);
+				return -EINVAL;
+			}
+		}
 	}
 
-	symmetry = cpu_dai->driver->symmetric_samplebits ||
-		rtd->dai_link->symmetric_samplebits;
+	symmetry = rtd->dai_link->symmetric_samplebits;
 
-	for_each_rtd_codec_dai(rtd, i, codec_dai)
-		symmetry |= codec_dai->driver->symmetric_samplebits;
+	for_each_rtd_dais(rtd, i, dai)
+		symmetry |= dai->driver->symmetric_samplebits;
 
-	if (symmetry && cpu_dai->sample_bits && cpu_dai->sample_bits != sample_bits) {
-		dev_err(rtd->dev, "ASoC: unmatched sample bits symmetry: %d - %d\n",
-				cpu_dai->sample_bits, sample_bits);
-		return -EINVAL;
+	if (symmetry) {
+		for_each_rtd_cpu_dais(rtd, i, cpu_dai) {
+			if (cpu_dai->sample_bits &&
+			    cpu_dai->sample_bits != sample_bits) {
+				dev_err(rtd->dev, "ASoC: unmatched sample bits symmetry: %d - %d\n",
+					cpu_dai->sample_bits, sample_bits);
+				return -EINVAL;
+			}
+		}
 	}
 
 	return 0;
···
 static bool soc_pcm_has_symmetry(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai_driver *cpu_driver = rtd->cpu_dai->driver;
 	struct snd_soc_dai_link *link = rtd->dai_link;
-	struct snd_soc_dai *codec_dai;
+	struct snd_soc_dai *dai;
 	unsigned int symmetry, i;
 
-	symmetry = cpu_driver->symmetric_rates || link->symmetric_rates ||
-		cpu_driver->symmetric_channels || link->symmetric_channels ||
-		cpu_driver->symmetric_samplebits || link->symmetric_samplebits;
+	symmetry = link->symmetric_rates ||
+		link->symmetric_channels ||
+		link->symmetric_samplebits;
 
-	for_each_rtd_codec_dai(rtd, i, codec_dai)
+	for_each_rtd_dais(rtd, i, dai)
 		symmetry = symmetry ||
-			codec_dai->driver->symmetric_rates ||
-			codec_dai->driver->symmetric_channels ||
-			codec_dai->driver->symmetric_samplebits;
+			dai->driver->symmetric_rates ||
+			dai->driver->symmetric_channels ||
+			dai->driver->symmetric_samplebits;
 
 	return symmetry;
 }
···
 static void soc_pcm_apply_msb(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *cpu_dai;
 	struct snd_soc_dai *codec_dai;
+	struct snd_soc_pcm_stream *pcm_codec, *pcm_cpu;
+	int stream = substream->stream;
 	int i;
-	unsigned int bits = 0, cpu_bits;
+	unsigned int bits = 0, cpu_bits = 0;
 
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		for_each_rtd_codec_dai(rtd, i, codec_dai) {
-			if (codec_dai->driver->playback.sig_bits == 0) {
-				bits = 0;
-				break;
-			}
-			bits = max(codec_dai->driver->playback.sig_bits, bits);
+	for_each_rtd_codec_dais(rtd, i, codec_dai) {
+		pcm_codec = snd_soc_dai_get_pcm_stream(codec_dai, stream);
+
+		if (pcm_codec->sig_bits == 0) {
+			bits = 0;
+			break;
 		}
-		cpu_bits = cpu_dai->driver->playback.sig_bits;
-	} else {
-		for_each_rtd_codec_dai(rtd, i, codec_dai) {
-			if (codec_dai->driver->capture.sig_bits == 0) {
-				bits = 0;
-				break;
-			}
-			bits = max(codec_dai->driver->capture.sig_bits, bits);
+		bits = max(pcm_codec->sig_bits, bits);
+	}
+
+	for_each_rtd_cpu_dais(rtd, i, cpu_dai) {
+		pcm_cpu = snd_soc_dai_get_pcm_stream(cpu_dai, stream);
+
+		if (pcm_cpu->sig_bits == 0) {
+			cpu_bits = 0;
+			break;
 		}
-		cpu_bits = cpu_dai->driver->capture.sig_bits;
+		cpu_bits = max(pcm_cpu->sig_bits, cpu_bits);
 	}
 
 	soc_pcm_set_msb(substream, bits);
 	soc_pcm_set_msb(substream, cpu_bits);
 }
 
-static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream)
+/**
+ * snd_soc_runtime_calc_hw() - Calculate hw limits for a PCM stream
+ * @rtd: ASoC PCM runtime
+ * @hw: PCM hardware parameters (output)
+ * @stream: Direction of the PCM stream
+ *
+ * Calculates the subset of stream parameters supported by all DAIs
+ * associated with the PCM stream.
+ */
+int snd_soc_runtime_calc_hw(struct snd_soc_pcm_runtime *rtd,
+			    struct snd_pcm_hardware *hw, int stream)
 {
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct snd_pcm_hardware *hw = &runtime->hw;
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *codec_dai;
-	struct snd_soc_dai_driver *cpu_dai_drv = rtd->cpu_dai->driver;
-	struct snd_soc_dai_driver *codec_dai_drv;
+	struct snd_soc_dai *cpu_dai;
 	struct snd_soc_pcm_stream *codec_stream;
 	struct snd_soc_pcm_stream *cpu_stream;
 	unsigned int chan_min = 0, chan_max = UINT_MAX;
+	unsigned int cpu_chan_min = 0, cpu_chan_max = UINT_MAX;
 	unsigned int rate_min = 0, rate_max = UINT_MAX;
-	unsigned int rates = UINT_MAX;
+	unsigned int cpu_rate_min = 0, cpu_rate_max = UINT_MAX;
+	unsigned int rates = UINT_MAX, cpu_rates = UINT_MAX;
 	u64 formats = ULLONG_MAX;
 	int i;
 
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		cpu_stream = &cpu_dai_drv->playback;
-	else
-		cpu_stream = &cpu_dai_drv->capture;
+	/* first calculate min/max only for CPUs in the DAI link */
+	for_each_rtd_cpu_dais(rtd, i, cpu_dai) {
 
-	/* first calculate min/max only for CODECs in the DAI link */
-	for_each_rtd_codec_dai(rtd, i, codec_dai) {
+		/*
+		 * Skip CPUs which don't support the current stream type.
+		 * Otherwise, since the rate, channel, and format values will
+		 * zero in that case, we would have no usable settings left,
+		 * causing the resulting setup to fail.
+		 */
+		if (!snd_soc_dai_stream_valid(cpu_dai, stream))
+			continue;
+
+		cpu_stream = snd_soc_dai_get_pcm_stream(cpu_dai, stream);
+
+		cpu_chan_min = max(cpu_chan_min, cpu_stream->channels_min);
+		cpu_chan_max = min(cpu_chan_max, cpu_stream->channels_max);
+		cpu_rate_min = max(cpu_rate_min, cpu_stream->rate_min);
+		cpu_rate_max = min_not_zero(cpu_rate_max, cpu_stream->rate_max);
+		formats &= cpu_stream->formats;
+		cpu_rates = snd_pcm_rate_mask_intersect(cpu_stream->rates,
+							cpu_rates);
+	}
+
+	/* second calculate min/max only for CODECs in the DAI link */
+	for_each_rtd_codec_dais(rtd, i, codec_dai) {
 
 		/*
 		 * Skip CODECs which don't support the current stream type.
 		 * Otherwise, since the rate, channel, and format values will
 		 * zero in that case, we would have no usable settings left,
 		 * causing the resulting setup to fail.
-		 * At least one CODEC should match, otherwise we should have
-		 * bailed out on a higher level, since there would be no
-		 * CODEC to support the transfer direction in that case.
 		 */
-		if (!snd_soc_dai_stream_valid(codec_dai,
-					      substream->stream))
+		if (!snd_soc_dai_stream_valid(codec_dai, stream))
 			continue;
 
-		codec_dai_drv = codec_dai->driver;
-		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-			codec_stream = &codec_dai_drv->playback;
-		else
-			codec_stream = &codec_dai_drv->capture;
+		codec_stream = snd_soc_dai_get_pcm_stream(codec_dai, stream);
+
 		chan_min = max(chan_min, codec_stream->channels_min);
 		chan_max = min(chan_max, codec_stream->channels_max);
 		rate_min = max(rate_min, codec_stream->rate_min);
···
 		rates = snd_pcm_rate_mask_intersect(codec_stream->rates, rates);
 	}
 
+	/* Verify both a valid CPU DAI and a valid CODEC DAI were found */
+	if (!chan_min || !cpu_chan_min)
+		return -EINVAL;
+
 	/*
 	 * chan min/max cannot be enforced if there are multiple CODEC DAIs
-	 * connected to a single CPU DAI, use CPU DAI's directly and let
+	 * connected to CPU DAI(s), use CPU DAI's directly and let
 	 * channel allocation be fixed up later
 	 */
 	if (rtd->num_codecs > 1) {
-		chan_min = cpu_stream->channels_min;
-		chan_max = cpu_stream->channels_max;
+		chan_min = cpu_chan_min;
+		chan_max = cpu_chan_max;
 	}
 
-	hw->channels_min = max(chan_min, cpu_stream->channels_min);
-	hw->channels_max = min(chan_max, cpu_stream->channels_max);
-	if (hw->formats)
-		hw->formats &= formats & cpu_stream->formats;
-	else
-		hw->formats = formats & cpu_stream->formats;
-	hw->rates = snd_pcm_rate_mask_intersect(rates, cpu_stream->rates);
+	/* finally find a intersection between CODECs and CPUs */
+	hw->channels_min = max(chan_min, cpu_chan_min);
+	hw->channels_max = min(chan_max, cpu_chan_max);
+	hw->formats = formats;
+	hw->rates = snd_pcm_rate_mask_intersect(rates, cpu_rates);
 
-	snd_pcm_limit_hw_rates(runtime);
+	snd_pcm_hw_limit_rates(hw);
 
-	hw->rate_min = max(hw->rate_min, cpu_stream->rate_min);
+	hw->rate_min = max(hw->rate_min, cpu_rate_min);
 	hw->rate_min = max(hw->rate_min, rate_min);
-	hw->rate_max = min_not_zero(hw->rate_max, cpu_stream->rate_max);
+	hw->rate_max = min_not_zero(hw->rate_max, cpu_rate_max);
 	hw->rate_max = min_not_zero(hw->rate_max, rate_max);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_runtime_calc_hw);
+
+static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_hardware *hw = &substream->runtime->hw;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	u64 formats = hw->formats;
+
+	/*
+	 * At least one CPU and one CODEC should match. Otherwise, we should
+	 * have bailed out on a higher level, since there would be no CPU or
+	 * CODEC to support the transfer direction in that case.
+	 */
+	snd_soc_runtime_calc_hw(rtd, hw, substream->stream);
+
+	if (formats)
+		hw->formats &= formats;
 }
 
-static int soc_pcm_components_open(struct snd_pcm_substream *substream,
-				   struct snd_soc_component **last)
+static int soc_pcm_components_open(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_component *last = NULL;
 	struct snd_soc_component *component;
 	int i, ret = 0;
 
 	for_each_rtd_components(rtd, i, component) {
-		*last = component;
+		last = component;
 
 		ret = snd_soc_component_module_get_when_open(component);
 		if (ret < 0) {
 			dev_err(component->dev,
 				"ASoC: can't get module %s\n",
 				component->name);
-			return ret;
+			break;
 		}
 
 		ret = snd_soc_component_open(component, substream);
 		if (ret < 0) {
+			snd_soc_component_module_put_when_close(component);
 			dev_err(component->dev,
 				"ASoC: can't open component %s: %d\n",
 				component->name, ret);
-			return ret;
+			break;
 		}
 	}
-	*last = NULL;
-	return 0;
+
+	if (ret < 0) {
+		/* rollback on error */
+		for_each_rtd_components(rtd, i, component) {
+			if (component == last)
+				break;
+
+			snd_soc_component_close(component, substream);
+			snd_soc_component_module_put_when_close(component);
+		}
+	}
+
+	return ret;
 }
 
-static int soc_pcm_components_close(struct snd_pcm_substream *substream,
-				    struct snd_soc_component *last)
+static int soc_pcm_components_close(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_component *component;
-	int i, ret = 0;
+	int i, r, ret = 0;
 
 	for_each_rtd_components(rtd, i, component) {
-		if (component == last)
-			break;
+		r = snd_soc_component_close(component, substream);
+		if (r < 0)
+			ret = r; /* use last ret */
 
-		ret |= snd_soc_component_close(component, substream);
 		snd_soc_component_module_put_when_close(component);
 	}
 
 	return ret;
+}
+
+/*
+ * Called by ALSA when a PCM substream is closed. Private data can be
+ * freed here. The cpu DAI, codec DAI, machine and components are also
+ * shutdown.
+ */
+static int soc_pcm_close(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_component *component;
+	struct snd_soc_dai *dai;
+	int i;
+
+	mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass);
+
+	snd_soc_runtime_deactivate(rtd, substream->stream);
+
+	for_each_rtd_dais(rtd, i, dai)
+		snd_soc_dai_shutdown(dai, substream);
+
+	soc_rtd_shutdown(rtd, substream);
+
+	soc_pcm_components_close(substream);
+
+	snd_soc_dapm_stream_stop(rtd, substream->stream);
+
+	mutex_unlock(&rtd->card->pcm_mutex);
+
+	for_each_rtd_components(rtd, i, component) {
+		pm_runtime_mark_last_busy(component->dev);
+		pm_runtime_put_autosuspend(component->dev);
+	}
+
+	for_each_rtd_components(rtd, i, component)
+		if (!component->active)
+			pinctrl_pm_select_sleep_state(component->dev);
+
+	return 0;
 }
 
 /*
···
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_component *component;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	struct snd_soc_dai *codec_dai;
+	struct snd_soc_dai *dai;
 	const char *codec_dai_name = "multicodec";
+	const char *cpu_dai_name = "multicpu";
 	int i, ret = 0;
 
 	for_each_rtd_components(rtd, i, component)
···
 
 	mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass);
 
-	/* startup the audio subsystem */
-	ret = snd_soc_dai_startup(cpu_dai, substream);
-	if (ret < 0) {
-		dev_err(cpu_dai->dev, "ASoC: can't open interface %s: %d\n",
-			cpu_dai->name, ret);
-		goto out;
-	}
-
-	ret = soc_pcm_components_open(substream, &component);
+	ret = soc_pcm_components_open(substream);
 	if (ret < 0)
 		goto component_err;
-
-	for_each_rtd_codec_dai(rtd, i, codec_dai) {
-		ret = snd_soc_dai_startup(codec_dai, substream);
-		if (ret < 0) {
-			dev_err(codec_dai->dev,
-				"ASoC: can't open codec %s: %d\n",
-				codec_dai->name, ret);
-			goto codec_dai_err;
-		}
-
-		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-			codec_dai->tx_mask = 0;
-		else
-			codec_dai->rx_mask = 0;
-	}
 
 	ret = soc_rtd_startup(rtd, substream);
 	if (ret < 0) {
 		pr_err("ASoC: %s startup failed: %d\n",
 		       rtd->dai_link->name, ret);
-		goto machine_err;
+		goto rtd_startup_err;
+	}
+
+	/* startup the audio subsystem */
+	for_each_rtd_dais(rtd, i, dai) {
+		ret = snd_soc_dai_startup(dai, substream);
+		if (ret < 0) {
+			dev_err(dai->dev,
+				"ASoC: can't open DAI %s: %d\n",
+				dai->name, ret);
+			goto config_err;
+		}
+
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+			dai->tx_mask = 0;
+		else
+			dai->rx_mask = 0;
 	}
 
 	/* Dynamic PCM DAI links compat checks use dynamic capabilities */
···
 	if (rtd->num_codecs == 1)
 		codec_dai_name = rtd->codec_dai->name;
 
+	if (rtd->num_cpus == 1)
+		cpu_dai_name = rtd->cpu_dai->name;
+
 	if (soc_pcm_has_symmetry(substream))
 		runtime->hw.info |= SNDRV_PCM_INFO_JOINT_DUPLEX;
 
 	ret = -EINVAL;
 	if (!runtime->hw.rates) {
 		printk(KERN_ERR "ASoC: %s <-> %s No matching rates\n",
-			codec_dai_name, cpu_dai->name);
+			codec_dai_name, cpu_dai_name);
 		goto config_err;
 	}
 	if (!runtime->hw.formats) {
 		printk(KERN_ERR "ASoC: %s <-> %s No matching formats\n",
-			codec_dai_name, cpu_dai->name);
+			codec_dai_name, cpu_dai_name);
 		goto config_err;
 	}
 	if (!runtime->hw.channels_min || !runtime->hw.channels_max ||
 	    runtime->hw.channels_min > runtime->hw.channels_max) {
 		printk(KERN_ERR "ASoC: %s <-> %s No matching channels\n",
-				codec_dai_name, cpu_dai->name);
+				codec_dai_name, cpu_dai_name);
 		goto config_err;
 	}
 
 	soc_pcm_apply_msb(substream);
 
 	/* Symmetry only applies if we've already got an active stream. */
-	if (cpu_dai->active) {
-		ret = soc_pcm_apply_symmetry(substream, cpu_dai);
-		if (ret != 0)
-			goto config_err;
-	}
-
-	for_each_rtd_codec_dai(rtd, i, codec_dai) {
-		if (codec_dai->active) {
-			ret = soc_pcm_apply_symmetry(substream, codec_dai);
+	for_each_rtd_dais(rtd, i, dai) {
+		if (dai->active) {
+			ret = soc_pcm_apply_symmetry(substream, dai);
 			if (ret != 0)
 				goto config_err;
 		}
 	}
 
 	pr_debug("ASoC: %s <-> %s info:\n",
-			codec_dai_name, cpu_dai->name);
+		 codec_dai_name, cpu_dai_name);
 	pr_debug("ASoC: rate mask 0x%x\n", runtime->hw.rates);
 	pr_debug("ASoC: min ch %d max ch %d\n", runtime->hw.channels_min,
 		 runtime->hw.channels_max);
···
 	return 0;
 
 config_err:
+	for_each_rtd_dais(rtd, i, dai)
+		snd_soc_dai_shutdown(dai, substream);
+
 	soc_rtd_shutdown(rtd, substream);
-
-machine_err:
-	i = rtd->num_codecs;
-
-codec_dai_err:
-	for_each_rtd_codec_dai_rollback(rtd, i, codec_dai)
-		snd_soc_dai_shutdown(codec_dai, substream);
-
+rtd_startup_err:
+	soc_pcm_components_close(substream);
 component_err:
-	soc_pcm_components_close(substream, component);
-
-	snd_soc_dai_shutdown(cpu_dai, substream);
-out:
 	mutex_unlock(&rtd->card->pcm_mutex);
 
 	for_each_rtd_components(rtd, i, component) {
···
 }
 
 /*
- * Called by ALSA when a PCM substream is closed. Private data can be
- * freed here. The cpu DAI, codec DAI, machine and components are also
- * shutdown.
- */
-static int soc_pcm_close(struct snd_pcm_substream *substream)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_component *component;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	struct snd_soc_dai *codec_dai;
-	int i;
-
-	mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass);
-
-	snd_soc_runtime_deactivate(rtd, substream->stream);
-
-	/* clear the corresponding DAIs rate when inactive */
-	if (!cpu_dai->active)
-		cpu_dai->rate = 0;
-
-	for_each_rtd_codec_dai(rtd, i, codec_dai) {
-		if (!codec_dai->active)
-			codec_dai->rate = 0;
-	}
-
-	snd_soc_dai_digital_mute(cpu_dai, 1, substream->stream);
-
-	snd_soc_dai_shutdown(cpu_dai, substream);
-
-	for_each_rtd_codec_dai(rtd, i, codec_dai)
-		snd_soc_dai_shutdown(codec_dai, substream);
-
-	soc_rtd_shutdown(rtd, substream);
-
-	soc_pcm_components_close(substream, NULL);
-
-	snd_soc_dapm_stream_stop(rtd, substream->stream);
-
-	mutex_unlock(&rtd->card->pcm_mutex);
-
-	for_each_rtd_components(rtd, i, component) {
-		pm_runtime_mark_last_busy(component->dev);
-		pm_runtime_put_autosuspend(component->dev);
-	}
-
-	for_each_rtd_components(rtd, i, component)
-		if (!component->active)
-			pinctrl_pm_select_sleep_state(component->dev);
-
-	return 0;
-}
-
-/*
  * Called by ALSA when the PCM substream is prepared, can set format, sample
  * rate, etc.  This function is non atomic and can be called multiple times,
  * it can refer to the runtime info.
···
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_component *component;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	struct snd_soc_dai *codec_dai;
+	struct snd_soc_dai *dai;
 	int i, ret = 0;
 
 	mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass);
···
 		}
 	}
 
-	for_each_rtd_codec_dai(rtd, i, codec_dai) {
-		ret = snd_soc_dai_prepare(codec_dai, substream);
+	for_each_rtd_dais(rtd, i, dai) {
+		ret = snd_soc_dai_prepare(dai, substream);
 		if (ret < 0) {
-			dev_err(codec_dai->dev,
-				"ASoC: codec DAI prepare error: %d\n",
-				ret);
+			dev_err(dai->dev,
+				"ASoC: DAI prepare error: %d\n", ret);
 			goto out;
 		}
-	}
-
-	ret = snd_soc_dai_prepare(cpu_dai, substream);
-	if (ret < 0) {
-		dev_err(cpu_dai->dev,
-			"ASoC: cpu DAI prepare error: %d\n", ret);
-		goto out;
 	}
 
 	/* cancel any delayed stream shutdown that is pending */
···
 	snd_soc_dapm_stream_event(rtd, substream->stream,
 			SND_SOC_DAPM_STREAM_START);
 
-	for_each_rtd_codec_dai(rtd, i, codec_dai)
-		snd_soc_dai_digital_mute(codec_dai, 0,
-					 substream->stream);
-	snd_soc_dai_digital_mute(cpu_dai, 0, substream->stream);
+	for_each_rtd_dais(rtd, i, dai)
+		snd_soc_dai_digital_mute(dai, 0, substream->stream);
 
 out:
 	mutex_unlock(&rtd->card->pcm_mutex);
···
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_component *component;
-	int i, ret = 0;
+	int i, r, ret = 0;
 
 	for_each_rtd_components(rtd, i, component) {
 		if (component == last)
 			break;
 
-		ret |= snd_soc_component_hw_free(component, substream);
+		r = snd_soc_component_hw_free(component, substream);
+		if (r < 0)
+			ret = r; /* use last ret */
 	}
 
 	return ret;
···
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_component *component;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *cpu_dai;
 	struct snd_soc_dai *codec_dai;
 	int i, ret = 0;
 
···
 		goto out;
 	}
 
-	for_each_rtd_codec_dai(rtd, i, codec_dai) {
+	for_each_rtd_codec_dais(rtd, i, codec_dai) {
 		struct snd_pcm_hw_params codec_params;
 
 		/*
···
 		snd_soc_dapm_update_dai(substream, &codec_params, codec_dai);
 	}
 
-	ret = snd_soc_dai_hw_params(cpu_dai, substream, params);
-	if (ret < 0)
-		goto interface_err;
+	for_each_rtd_cpu_dais(rtd, i, cpu_dai) {
+		/*
+		 * Skip CPUs which don't support the current stream
+		 * type. See soc_pcm_init_runtime_hw() for more details
+		 */
+		if (!snd_soc_dai_stream_valid(cpu_dai, substream->stream))
+			continue;
 
-	/* store the parameters for each DAIs */
-	cpu_dai->rate = params_rate(params);
-	cpu_dai->channels = params_channels(params);
-	cpu_dai->sample_bits =
-		snd_pcm_format_physical_width(params_format(params));
+		ret = snd_soc_dai_hw_params(cpu_dai, substream, params);
+		if (ret < 0)
+			goto interface_err;
 
-	snd_soc_dapm_update_dai(substream, params, cpu_dai);
+		/* store the parameters for each DAI */
+		cpu_dai->rate = params_rate(params);
+		cpu_dai->channels = params_channels(params);
+		cpu_dai->sample_bits =
+			snd_pcm_format_physical_width(params_format(params));
+
+		snd_soc_dapm_update_dai(substream, params, cpu_dai);
+	}
 
 	for_each_rtd_components(rtd, i, component) {
 		ret = snd_soc_component_hw_params(component, substream, params);
···
 component_err:
 	soc_pcm_components_hw_free(substream, component);
 
-	snd_soc_dai_hw_free(cpu_dai, substream);
-	cpu_dai->rate = 0;
+	i = rtd->num_cpus;
 
 interface_err:
+	for_each_rtd_cpu_dais_rollback(rtd, i, cpu_dai) {
+		if (!snd_soc_dai_stream_valid(cpu_dai, substream->stream))
+			continue;
+
+		snd_soc_dai_hw_free(cpu_dai, substream);
+		cpu_dai->rate = 0;
+	}
+
 	i = rtd->num_codecs;
 
 codec_err:
-	for_each_rtd_codec_dai_rollback(rtd, i, codec_dai) {
+	for_each_rtd_codec_dais_rollback(rtd, i, codec_dai) {
 		if (!snd_soc_dai_stream_valid(codec_dai, substream->stream))
 			continue;
 
···
 static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	struct snd_soc_dai *codec_dai;
-	bool playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+	struct snd_soc_dai *dai;
 	int i;
 
 	mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass);
 
 	/* clear the corresponding DAIs parameters when going to be inactive */
-	if (cpu_dai->active == 1) {
-		cpu_dai->rate = 0;
-		cpu_dai->channels = 0;
-		cpu_dai->sample_bits = 0;
-	}
+	for_each_rtd_dais(rtd, i, dai) {
+		int active = dai->stream_active[substream->stream];
 
-	for_each_rtd_codec_dai(rtd, i, codec_dai) {
-		if (codec_dai->active == 1) {
-			codec_dai->rate = 0;
-			codec_dai->channels = 0;
-			codec_dai->sample_bits = 0;
+		if (dai->active == 1) {
+			dai->rate = 0;
+			dai->channels = 0;
+			dai->sample_bits = 0;
 		}
-	}
 
-	/* apply codec digital mute */
-	for_each_rtd_codec_dai(rtd, i, codec_dai) {
-		if ((playback && codec_dai->playback_active == 1) ||
-		    (!playback && codec_dai->capture_active == 1))
-			snd_soc_dai_digital_mute(codec_dai, 1,
-						 substream->stream);
+		if (active == 1)
+			snd_soc_dai_digital_mute(dai, 1, substream->stream);
 	}
 
 	/* free any machine hw params */
···
 	soc_pcm_components_hw_free(substream, NULL);
 
 	/* now free hw params for the DAIs  */
-	for_each_rtd_codec_dai(rtd, i, codec_dai) {
-		if (!snd_soc_dai_stream_valid(codec_dai, substream->stream))
+	for_each_rtd_dais(rtd, i, dai) {
+		if (!snd_soc_dai_stream_valid(dai, substream->stream))
 			continue;
 
-		snd_soc_dai_hw_free(codec_dai, substream);
+		snd_soc_dai_hw_free(dai, substream);
 	}
-
-	snd_soc_dai_hw_free(cpu_dai, substream);
 
 	mutex_unlock(&rtd->card->pcm_mutex);
 	return 0;
···
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_component *component;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	struct snd_soc_dai *codec_dai;
+	struct snd_soc_dai *dai;
 	int i, ret;
 
 	ret = soc_rtd_trigger(rtd, substream, cmd);
···
 			return ret;
 	}
 
-	ret = snd_soc_dai_trigger(cpu_dai, substream, cmd);
-	if (ret < 0)
-		return ret;
-
-	for_each_rtd_codec_dai(rtd, i, codec_dai) {
-		ret = snd_soc_dai_trigger(codec_dai, substream, cmd);
+	for_each_rtd_dais(rtd, i, dai) {
+		ret = snd_soc_dai_trigger(dai, substream, cmd);
 		if (ret < 0)
 			return ret;
 	}
···
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_component *component;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	struct snd_soc_dai *codec_dai;
+	struct snd_soc_dai *dai;
 	int i, ret;
 
-	for_each_rtd_codec_dai(rtd, i, codec_dai) {
-		ret = snd_soc_dai_trigger(codec_dai, substream, cmd);
+	for_each_rtd_dais(rtd, i, dai) {
+		ret = snd_soc_dai_trigger(dai, substream, cmd);
 		if (ret < 0)
 			return ret;
 	}
-
-	ret = snd_soc_dai_trigger(cpu_dai, substream, cmd);
-	if (ret < 0)
-		return ret;
 
 	for_each_rtd_components(rtd, i, component) {
 		ret = snd_soc_component_trigger(component, substream, cmd);
···
 				   int cmd)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	struct snd_soc_dai *codec_dai;
+	struct snd_soc_dai *dai;
 	int i, ret;
 
-	for_each_rtd_codec_dai(rtd, i, codec_dai) {
-		ret = snd_soc_dai_bespoke_trigger(codec_dai, substream, cmd);
+	for_each_rtd_dais(rtd, i, dai) {
+		ret = snd_soc_dai_bespoke_trigger(dai, substream, cmd);
 		if (ret < 0)
 			return ret;
 	}
-
-	ret = snd_soc_dai_bespoke_trigger(cpu_dai, substream, cmd);
-	if (ret < 0)
-		return ret;
 
 	return 0;
 }
···
 static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *cpu_dai;
 	struct snd_soc_dai *codec_dai;
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	snd_pcm_uframes_t offset = 0;
 	snd_pcm_sframes_t delay = 0;
 	snd_pcm_sframes_t codec_delay = 0;
+	snd_pcm_sframes_t cpu_delay = 0;
 	int i;
 
 	/* clearing the previous total delay */
···
 	/* base delay if assigned in pointer callback */
 	delay = runtime->delay;
 
-	delay += snd_soc_dai_delay(cpu_dai, substream);
+	for_each_rtd_cpu_dais(rtd, i, cpu_dai) {
+		cpu_delay = max(cpu_delay,
+				snd_soc_dai_delay(cpu_dai, substream));
+	}
+	delay += cpu_delay;
 
-	for_each_rtd_codec_dai(rtd, i, codec_dai) {
+	for_each_rtd_codec_dais(rtd, i, codec_dai) {
 		codec_delay = max(codec_delay,
 				  snd_soc_dai_delay(codec_dai, substream));
 	}
···
 {
 	struct snd_soc_dpcm *dpcm;
 	unsigned long flags;
-#ifdef CONFIG_DEBUG_FS
-	char *name;
-#endif
 
 	/* only add new dpcms */
 	for_each_dpcm_be(fe, stream, dpcm) {
···
 			stream ? "capture" : "playback",  fe->dai_link->name,
 			stream ? "<-" : "->", be->dai_link->name);
 
-#ifdef CONFIG_DEBUG_FS
-	name = kasprintf(GFP_KERNEL, "%s:%s", be->dai_link->name,
-			 stream ? "capture" : "playback");
-	if (name) {
-		dpcm->debugfs_state = debugfs_create_dir(name,
-							 fe->debugfs_dpcm_root);
-		debugfs_create_u32("state", 0644, dpcm->debugfs_state,
-				   &dpcm->state);
-		kfree(name);
-	}
-#endif
+	dpcm_create_debugfs_state(dpcm, stream);
+
 	return 1;
 }
 
···
 		/* BEs still alive need new FE */
 		dpcm_be_reparent(fe, dpcm->be, stream);
 
-#ifdef CONFIG_DEBUG_FS
-		debugfs_remove_recursive(dpcm->debugfs_state);
-#endif
+		dpcm_remove_debugfs_state(dpcm);
+
 		spin_lock_irqsave(&fe->card->dpcm_lock, flags);
 		list_del(&dpcm->list_be);
 		list_del(&dpcm->list_fe);
···
 		struct snd_soc_dapm_widget *widget, int stream)
 {
 	struct snd_soc_pcm_runtime *be;
+	struct snd_soc_dapm_widget *w;
 	struct snd_soc_dai *dai;
 	int i;
 
 	dev_dbg(card->dev, "ASoC: find BE for widget %s\n", widget->name);
 
-	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		for_each_card_rtds(card, be) {
+	for_each_card_rtds(card, be) {
 
-			if (!be->dai_link->no_pcm)
-				continue;
+		if (!be->dai_link->no_pcm)
+			continue;
+
+		for_each_rtd_dais(be, i, dai) {
+			w = snd_soc_dai_get_widget(dai, stream);
 
 			dev_dbg(card->dev, "ASoC: try BE : %s\n",
-				be->cpu_dai->playback_widget ?
-				be->cpu_dai->playback_widget->name : "(not set)");
+				w ? w->name : "(not set)");
 
-			if (be->cpu_dai->playback_widget == widget)
+			if (w == widget)
 				return be;
-
-			for_each_rtd_codec_dai(be, i, dai) {
-				if (dai->playback_widget == widget)
-					return be;
-			}
-		}
-	} else {
-
-		for_each_card_rtds(card, be) {
-
-			if (!be->dai_link->no_pcm)
-				continue;
-
-			dev_dbg(card->dev, "ASoC: try BE %s\n",
-				be->cpu_dai->capture_widget ?
-				be->cpu_dai->capture_widget->name : "(not set)");
-
-			if (be->cpu_dai->capture_widget == widget)
-				return be;
-
-			for_each_rtd_codec_dai(be, i, dai) {
-				if (dai->capture_widget == widget)
-					return be;
-			}
 		}
 	}
 
-	/* dai link name and stream name set correctly ? */
-	dev_err(card->dev, "ASoC: can't get %s BE for %s\n",
-		stream ? "capture" : "playback", widget->name);
+	/* Widget provided is not a BE */
 	return NULL;
-}
-
-static inline struct snd_soc_dapm_widget *
-	dai_get_widget(struct snd_soc_dai *dai, int stream)
-{
-	if (stream == SNDRV_PCM_STREAM_PLAYBACK)
-		return dai->playback_widget;
-	else
-		return dai->capture_widget;
 }
 
 static int widget_in_list(struct snd_soc_dapm_widget_list *list,
 		struct snd_soc_dapm_widget *widget)
 {
+	struct snd_soc_dapm_widget *w;
 	int i;
 
-	for (i = 0; i < list->num_widgets; i++) {
-		if (widget == list->widgets[i])
+	for_each_dapm_widgets(list, i, w)
+		if (widget == w)
 			return 1;
-	}
 
 	return 0;
 }
···
 {
 	struct snd_soc_card *card = widget->dapm->card;
 	struct snd_soc_pcm_runtime *rtd;
-	struct snd_soc_dai *dai;
-	int i;
+	int stream;
 
-	if (dir == SND_SOC_DAPM_DIR_OUT) {
-		for_each_card_rtds(card, rtd) {
-			if (!rtd->dai_link->no_pcm)
-				continue;
+	/* adjust dir to stream */
+	if (dir == SND_SOC_DAPM_DIR_OUT)
+		stream = SNDRV_PCM_STREAM_PLAYBACK;
+	else
+		stream = SNDRV_PCM_STREAM_CAPTURE;
 
-			if (rtd->cpu_dai->playback_widget == widget)
-				return true;
-
-			for_each_rtd_codec_dai(rtd, i, dai) {
-				if (dai->playback_widget == widget)
-					return true;
-			}
-		}
-	} else { /* SND_SOC_DAPM_DIR_IN */
-		for_each_card_rtds(card, rtd) {
-			if (!rtd->dai_link->no_pcm)
-				continue;
-
-			if (rtd->cpu_dai->capture_widget == widget)
-				return true;
-
-			for_each_rtd_codec_dai(rtd, i, dai) {
-				if (dai->capture_widget == widget)
-					return true;
-			}
-		}
-	}
+	rtd = dpcm_get_be(card, widget, stream);
+	if (rtd)
+		return true;
 
 	return false;
 }
···
 {
 	struct snd_soc_dai *cpu_dai = fe->cpu_dai;
 	int paths;
+
+	if (fe->num_cpus > 1) {
+		dev_err(fe->dev,
+			"%s doesn't support Multi CPU yet\n", __func__);
+		return -EINVAL;
+	}
 
 	/* get number of valid DAI paths and their widgets */
 	paths = snd_soc_dapm_dai_get_connected_widgets(cpu_dai, stream, list,
···
 	return paths;
 }
 
-static int dpcm_prune_paths(struct snd_soc_pcm_runtime *fe, int stream,
-	struct snd_soc_dapm_widget_list **list_)
+void dpcm_path_put(struct snd_soc_dapm_widget_list **list)
 {
-	struct snd_soc_dpcm *dpcm;
-	struct snd_soc_dapm_widget_list *list = *list_;
+	snd_soc_dapm_dai_free_widgets(list);
+}
+
+static bool dpcm_be_is_active(struct snd_soc_dpcm *dpcm, int stream,
+			      struct snd_soc_dapm_widget_list *list)
+{
 	struct snd_soc_dapm_widget *widget;
 	struct snd_soc_dai *dai;
+	unsigned int i;
+
+	/* is there a valid DAI widget for this BE */
+	for_each_rtd_dais(dpcm->be, i, dai) {
+		widget = snd_soc_dai_get_widget(dai, stream);
+
+		/*
+		 * The BE is pruned only if none of the dai
+		 * widgets are in the active list.
+		 */
+		if (widget && widget_in_list(list, widget))
+			return true;
+	}
+
+	return false;
+}
+
+static int dpcm_prune_paths(struct snd_soc_pcm_runtime *fe, int stream,
+			    struct snd_soc_dapm_widget_list **list_)
+{
+	struct snd_soc_dpcm *dpcm;
 	int prune = 0;
-	int do_prune;
 
 	/* Destroy any old FE <--> BE connections */
 	for_each_dpcm_be(fe, stream, dpcm) {
-		unsigned int i;
-
-		/* is there a valid CPU DAI widget for this BE */
-		widget = dai_get_widget(dpcm->be->cpu_dai, stream);
-
-		/* prune the BE if it's no longer in our active list */
-		if (widget && widget_in_list(list, widget))
-			continue;
-
-		/* is there a valid CODEC DAI widget for this BE */
-		do_prune = 1;
-		for_each_rtd_codec_dai(dpcm->be, i, dai) {
-			widget = dai_get_widget(dai, stream);
-
-			/* prune the BE if it's no longer in our active list */
-			if (widget && widget_in_list(list, widget))
-				do_prune = 0;
-		}
-		if (!do_prune)
+		if (dpcm_be_is_active(dpcm, stream, *list_))
 			continue;
 
 		dev_dbg(fe->dev, "ASoC: pruning %s BE %s for %s\n",
···
 	struct snd_soc_card *card = fe->card;
 	struct snd_soc_dapm_widget_list *list = *list_;
 	struct snd_soc_pcm_runtime *be;
+	struct snd_soc_dapm_widget *widget;
 	int i, new = 0, err;
 
 	/* Create any new FE <--> BE connections */
-	for (i = 0; i < list->num_widgets; i++) {
+	for_each_dapm_widgets(list, i, widget) {
 
-		switch (list->widgets[i]->id) {
+		switch (widget->id) {
 		case snd_soc_dapm_dai_in:
 			if (stream != SNDRV_PCM_STREAM_PLAYBACK)
 				continue;
···
 		}
 
 		/* is there a valid BE rtd for this widget */
-		be = dpcm_get_be(card, list->widgets[i], stream);
+		be = dpcm_get_be(card, widget, stream);
 		if (!be) {
 			dev_err(fe->dev, "ASoC: no BE found for %s\n",
-					list->widgets[i]->name);
+					widget->name);
 			continue;
 		}
-
-		/* make sure BE is a real BE */
-		if (!be->dai_link->no_pcm)
-			continue;
 
 		/* don't connect if FE is not running */
 		if (!fe->dpcm[stream].runtime && !fe->fe_compr)
···
 		err = dpcm_be_connect(fe, be, stream);
 		if (err < 0) {
 			dev_err(fe->dev, "ASoC: can't connect %s\n",
-				list->widgets[i]->name);
+				widget->name);
 			break;
 		} else if (err == 0) /* already connected */
 			continue;
···
 
 	for_each_dpcm_be(fe, stream, dpcm) {
 		struct snd_soc_pcm_runtime *be = dpcm->be;
-		struct snd_soc_dai_driver *codec_dai_drv;
 		struct snd_soc_pcm_stream *codec_stream;
 		int i;
 
-		for_each_rtd_codec_dai(be, i, dai) {
+		for_each_rtd_codec_dais(be, i, dai) {
 			/*
 			 * Skip CODECs which don't support the current stream
 			 * type. See soc_pcm_init_runtime_hw() for more details
···
 			if (!snd_soc_dai_stream_valid(dai, stream))
 				continue;
 
-			codec_dai_drv = dai->driver;
-			if (stream == SNDRV_PCM_STREAM_PLAYBACK)
-				codec_stream = &codec_dai_drv->playback;
-			else
-				codec_stream = &codec_dai_drv->capture;
+			codec_stream = snd_soc_dai_get_pcm_stream(dai, stream);
 
 			*formats &= codec_stream->formats;
 		}
···
 
 	for_each_dpcm_be(fe, stream, dpcm) {
 		struct snd_soc_pcm_runtime *be = dpcm->be;
-		struct snd_soc_dai_driver *cpu_dai_drv =  be->cpu_dai->driver;
-		struct snd_soc_dai_driver *codec_dai_drv;
 		struct snd_soc_pcm_stream *codec_stream;
 		struct snd_soc_pcm_stream *cpu_stream;
+		struct snd_soc_dai *dai;
+		int i;
 
-		if (stream == SNDRV_PCM_STREAM_PLAYBACK)
-			cpu_stream = &cpu_dai_drv->playback;
-		else
-			cpu_stream = &cpu_dai_drv->capture;
+		for_each_rtd_cpu_dais(be, i, dai) {
+			/*
+			 * Skip CPUs which don't support the current stream
+			 * type. See soc_pcm_init_runtime_hw() for more details
+			 */
+			if (!snd_soc_dai_stream_valid(dai, stream))
+				continue;
 
-		*channels_min = max(*channels_min, cpu_stream->channels_min);
-		*channels_max = min(*channels_max, cpu_stream->channels_max);
+			cpu_stream = snd_soc_dai_get_pcm_stream(dai, stream);
+
+			*channels_min = max(*channels_min,
+					    cpu_stream->channels_min);
+			*channels_max = min(*channels_max,
+					    cpu_stream->channels_max);
+		}
 
 		/*
 		 * chan min/max cannot be enforced if there are multiple CODEC
 		 * DAIs connected to a single CPU DAI, use CPU DAI's directly
 		 */
 		if (be->num_codecs == 1) {
-			codec_dai_drv = be->codec_dais[0]->driver;
-
-			if (stream == SNDRV_PCM_STREAM_PLAYBACK)
-				codec_stream = &codec_dai_drv->playback;
-			else
-				codec_stream = &codec_dai_drv->capture;
+			codec_stream = snd_soc_dai_get_pcm_stream(be->codec_dais[0], stream);
 
 			*channels_min = max(*channels_min,
 					    codec_stream->channels_min);
···
 
 	for_each_dpcm_be(fe, stream, dpcm) {
 		struct snd_soc_pcm_runtime *be = dpcm->be;
-		struct snd_soc_dai_driver *cpu_dai_drv =  be->cpu_dai->driver;
-		struct snd_soc_dai_driver *codec_dai_drv;
-		struct snd_soc_pcm_stream *codec_stream;
-		struct snd_soc_pcm_stream *cpu_stream;
+		struct snd_soc_pcm_stream *pcm;
 		struct snd_soc_dai *dai;
 		int i;
 
-		if (stream == SNDRV_PCM_STREAM_PLAYBACK)
-			cpu_stream = &cpu_dai_drv->playback;
-		else
-			cpu_stream = &cpu_dai_drv->capture;
-
-		*rate_min = max(*rate_min, cpu_stream->rate_min);
-		*rate_max = min_not_zero(*rate_max, cpu_stream->rate_max);
-		*rates = snd_pcm_rate_mask_intersect(*rates, cpu_stream->rates);
-
-		for_each_rtd_codec_dai(be, i, dai) {
+		for_each_rtd_dais(be, i, dai) {
 			/*
-			 * Skip CODECs which don't support the current stream
+			 * Skip DAIs which don't support the current stream
 			 * type. See soc_pcm_init_runtime_hw() for more details
 			 */
 			if (!snd_soc_dai_stream_valid(dai, stream))
 				continue;
 
-			codec_dai_drv = dai->driver;
-			if (stream == SNDRV_PCM_STREAM_PLAYBACK)
-				codec_stream = &codec_dai_drv->playback;
-			else
-				codec_stream = &codec_dai_drv->capture;
+			pcm = snd_soc_dai_get_pcm_stream(dai, stream);
 
-			*rate_min = max(*rate_min, codec_stream->rate_min);
-			*rate_max = min_not_zero(*rate_max,
-						 codec_stream->rate_max);
-			*rates = snd_pcm_rate_mask_intersect(*rates,
-						codec_stream->rates);
+			*rate_min = max(*rate_min, pcm->rate_min);
+			*rate_max = min_not_zero(*rate_max, pcm->rate_max);
+			*rates = snd_pcm_rate_mask_intersect(*rates, pcm->rates);
 		}
 	}
 }
···
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	struct snd_soc_dai_driver *cpu_dai_drv = cpu_dai->driver;
+	struct snd_soc_dai *cpu_dai;
+	int i;
 
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		dpcm_init_runtime_hw(runtime, &cpu_dai_drv->playback);
-	else
-		dpcm_init_runtime_hw(runtime, &cpu_dai_drv->capture);
+	for_each_rtd_cpu_dais(rtd, i, cpu_dai) {
+		/*
+		 * Skip CPUs which don't support the current stream
+		 * type. See soc_pcm_init_runtime_hw() for more details
+		 */
+		if (!snd_soc_dai_stream_valid(cpu_dai, substream->stream))
+			continue;
+
+		dpcm_init_runtime_hw(runtime,
+			snd_soc_dai_get_pcm_stream(cpu_dai,
+						   substream->stream));
+	}
 
 	dpcm_runtime_merge_format(substream, &runtime->hw.formats);
 	dpcm_runtime_merge_chan(substream, &runtime->hw.channels_min,
···
 {
 	struct snd_soc_dpcm *dpcm;
 	struct snd_soc_pcm_runtime *fe = fe_substream->private_data;
-	struct snd_soc_dai *fe_cpu_dai = fe->cpu_dai;
+	struct snd_soc_dai *fe_cpu_dai;
 	int err;
+	int i;
 
 	/* apply symmetry for FE */
 	if (soc_pcm_has_symmetry(fe_substream))
 		fe_substream->runtime->hw.info |= SNDRV_PCM_INFO_JOINT_DUPLEX;
 
-	/* Symmetry only applies if we've got an active stream. */
-	if (fe_cpu_dai->active) {
-		err = soc_pcm_apply_symmetry(fe_substream, fe_cpu_dai);
-		if (err < 0)
-			return err;
+	for_each_rtd_cpu_dais (fe, i, fe_cpu_dai) {
+		/* Symmetry only applies if we've got an active stream. */
+		if (fe_cpu_dai->active) {
+			err = soc_pcm_apply_symmetry(fe_substream, fe_cpu_dai);
+			if (err < 0)
+				return err;
+		}
 	}
 
 	/* apply symmetry for BE */
···
 		struct snd_pcm_substream *be_substream =
 			snd_soc_dpcm_get_substream(be, stream);
 		struct snd_soc_pcm_runtime *rtd;
-		struct snd_soc_dai *codec_dai;
+		struct snd_soc_dai *dai;
 		int i;
 
 		/* A backend may not have the requested substream */
···
 			be_substream->runtime->hw.info |= SNDRV_PCM_INFO_JOINT_DUPLEX;
 
 		/* Symmetry only applies if we've got an active stream. */
-		if (rtd->cpu_dai->active) {
-			err = soc_pcm_apply_symmetry(fe_substream,
-						     rtd->cpu_dai);
-			if (err < 0)
-				return err;
-		}
-
-		for_each_rtd_codec_dai(rtd, i, codec_dai) {
-			if (codec_dai->active) {
-				err = soc_pcm_apply_symmetry(fe_substream,
-							     codec_dai);
+		for_each_rtd_dais(rtd, i, dai) {
+			if (dai->active) {
+				err = soc_pcm_apply_symmetry(fe_substream, dai);
 				if (err < 0)
 					return err;
 			}
···
 
 	dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_FE);
 
-	ret = dpcm_be_dai_startup(fe, fe_substream->stream);
+	ret = dpcm_be_dai_startup(fe, stream);
 	if (ret < 0) {
 		dev_err(fe->dev,"ASoC: failed to start some BEs %d\n", ret);
 		goto be_err;
···
 	snd_pcm_limit_hw_rates(runtime);
 
 	ret = dpcm_apply_symmetry(fe_substream, stream);
-	if (ret < 0) {
+	if (ret < 0)
 		dev_err(fe->dev, "ASoC: failed to apply dpcm symmetry %d\n",
 			ret);
-		goto unwind;
-	}
-
-	dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO);
-	return 0;
 
 unwind:
-	dpcm_be_dai_startup_unwind(fe, fe_substream->stream);
+	if (ret < 0)
+		dpcm_be_dai_startup_unwind(fe, stream);
 be_err:
 	dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO);
 	return ret;
···
 	dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_FE);
 
 	/* shutdown the BEs */
-	dpcm_be_dai_shutdown(fe, substream->stream);
+	dpcm_be_dai_shutdown(fe, stream);
 
 	dev_dbg(fe->dev, "ASoC: close FE %s\n", fe->dai_link->name);
 
···
 	mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME);
 	dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_FE);
 
-	memcpy(&fe->dpcm[substream->stream].hw_params, params,
+	memcpy(&fe->dpcm[stream].hw_params, params,
 			sizeof(struct snd_pcm_hw_params));
-	ret = dpcm_be_dai_hw_params(fe, substream->stream);
+	ret = dpcm_be_dai_hw_params(fe, stream);
 	if (ret < 0) {
 		dev_err(fe->dev,"ASoC: hw_params BE failed %d\n", ret);
 		goto out;
···
 		goto out;
 	}
 
-	ret = dpcm_be_dai_prepare(fe, substream->stream);
+	ret = dpcm_be_dai_prepare(fe, stream);
 	if (ret < 0)
 		goto out;
 
···
 	return ret;
 }
 
-static int dpcm_run_new_update(struct snd_soc_pcm_runtime *fe, int stream)
-{
-	int ret;
-
-	dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_BE);
-	ret = dpcm_run_update_startup(fe, stream);
-	if (ret < 0)
-		dev_err(fe->dev, "ASoC: failed to startup some BEs\n");
-	dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO);
-
-	return ret;
-}
-
-static int dpcm_run_old_update(struct snd_soc_pcm_runtime *fe, int stream)
-{
-	int ret;
-
-	dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_BE);
-	ret = dpcm_run_update_shutdown(fe, stream);
-	if (ret < 0)
-		dev_err(fe->dev, "ASoC: failed to shutdown some BEs\n");
-	dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO);
-
-	return ret;
-}
-
 static int soc_dpcm_fe_runtime_update(struct snd_soc_pcm_runtime *fe, int new)
 {
 	struct snd_soc_dapm_widget_list *list;
+	int stream;
 	int count, paths;
+	int ret;
+
+	if (fe->num_cpus > 1) {
+		dev_err(fe->dev,
+			"%s doesn't support Multi CPU yet\n", __func__);
+		return -EINVAL;
+	}
 
 	if (!fe->dai_link->dynamic)
 		return 0;
···
 	dev_dbg(fe->dev, "ASoC: DPCM %s runtime update for FE %s\n",
 		new ? "new" : "old", fe->dai_link->name);
 
-	/* skip if FE doesn't have playback capability */
-	if (!snd_soc_dai_stream_valid(fe->cpu_dai,   SNDRV_PCM_STREAM_PLAYBACK) ||
-	    !snd_soc_dai_stream_valid(fe->codec_dai, SNDRV_PCM_STREAM_PLAYBACK))
-		goto capture;
+	for_each_pcm_streams(stream) {
 
-	/* skip if FE isn't currently playing */
-	if (!fe->cpu_dai->playback_active || !fe->codec_dai->playback_active)
-		goto capture;
+		/* skip if FE doesn't have playback/capture capability */
+		if (!snd_soc_dai_stream_valid(fe->cpu_dai,   stream) ||
+		    !snd_soc_dai_stream_valid(fe->codec_dai, stream))
+			continue;
 
-	paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_PLAYBACK, &list);
-	if (paths < 0) {
-		dev_warn(fe->dev, "ASoC: %s no valid %s path\n",
-			 fe->dai_link->name,  "playback");
-		return paths;
+		/* skip if FE isn't currently playing/capturing */
+		if (!fe->cpu_dai->stream_active[stream] ||
+		    !fe->codec_dai->stream_active[stream])
+			continue;
+
+		paths = dpcm_path_get(fe, stream, &list);
+		if (paths < 0) {
+			dev_warn(fe->dev, "ASoC: %s no valid %s path\n",
+				 fe->dai_link->name,
+				 stream == SNDRV_PCM_STREAM_PLAYBACK ?
+				 "playback" : "capture");
+			return paths;
+		}
+
+		/* update any playback/capture paths */
+		count = dpcm_process_paths(fe, stream, &list, new);
+		if (count) {
+			dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_BE);
+			if (new)
+				ret = dpcm_run_update_startup(fe, stream);
+			else
+				ret = dpcm_run_update_shutdown(fe, stream);
+			if (ret < 0)
+				dev_err(fe->dev, "ASoC: failed to shutdown some BEs\n");
+			dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO);
+
+			dpcm_clear_pending_state(fe, stream);
+			dpcm_be_disconnect(fe, stream);
+		}
+
+		dpcm_path_put(&list);
 	}
-
-	/* update any playback paths */
-	count = dpcm_process_paths(fe, SNDRV_PCM_STREAM_PLAYBACK, &list, new);
-	if (count) {
-		if (new)
-			dpcm_run_new_update(fe, SNDRV_PCM_STREAM_PLAYBACK);
-		else
-			dpcm_run_old_update(fe, SNDRV_PCM_STREAM_PLAYBACK);
-
-		dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_PLAYBACK);
-		dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_PLAYBACK);
-	}
-
-	dpcm_path_put(&list);
-
-capture:
-	/* skip if FE doesn't have capture capability */
-	if (!snd_soc_dai_stream_valid(fe->cpu_dai,   SNDRV_PCM_STREAM_CAPTURE) ||
-	    !snd_soc_dai_stream_valid(fe->codec_dai, SNDRV_PCM_STREAM_CAPTURE))
-		return 0;
-
-	/* skip if FE isn't currently capturing */
-	if (!fe->cpu_dai->capture_active || !fe->codec_dai->capture_active)
-		return 0;
-
-	paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_CAPTURE, &list);
-	if (paths < 0) {
-		dev_warn(fe->dev, "ASoC: %s no valid %s path\n",
-			 fe->dai_link->name,  "capture");
-		return paths;
-	}
-
-	/* update any old capture paths */
-	count = dpcm_process_paths(fe, SNDRV_PCM_STREAM_CAPTURE, &list, new);
-	if (count) {
-		if (new)
-			dpcm_run_new_update(fe, SNDRV_PCM_STREAM_CAPTURE);
-		else
-			dpcm_run_old_update(fe, SNDRV_PCM_STREAM_CAPTURE);
-
-		dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_CAPTURE);
-		dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_CAPTURE);
-	}
-
-	dpcm_path_put(&list);
 
 	return 0;
 }
···
 /* Called by DAPM mixer/mux changes to update audio routing between PCMs and
  * any DAI links.
  */
-int soc_dpcm_runtime_update(struct snd_soc_card *card)
+int snd_soc_dpcm_runtime_update(struct snd_soc_card *card)
 {
 	struct snd_soc_pcm_runtime *fe;
 	int ret = 0;
···
 	mutex_unlock(&card->mutex);
 	return ret;
 }
-int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute)
+EXPORT_SYMBOL_GPL(snd_soc_dpcm_runtime_update);
+
+static void dpcm_fe_dai_cleanup(struct snd_pcm_substream *fe_substream)
 {
+	struct snd_soc_pcm_runtime *fe = fe_substream->private_data;
 	struct snd_soc_dpcm *dpcm;
-	struct snd_soc_dai *dai;
+	int stream = fe_substream->stream;
 
-	for_each_dpcm_be(fe, SNDRV_PCM_STREAM_PLAYBACK, dpcm) {
+	/* mark FE's links ready to prune */
+	for_each_dpcm_be(fe, stream, dpcm)
+		dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE;
 
-		struct snd_soc_pcm_runtime *be = dpcm->be;
-		int i;
+	dpcm_be_disconnect(fe, stream);
 
-		if (be->dai_link->ignore_suspend)
-			continue;
+	fe->dpcm[stream].runtime = NULL;
+}
 
-		for_each_rtd_codec_dai(be, i, dai) {
-			struct snd_soc_dai_driver *drv = dai->driver;
+static int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream)
+{
+	struct snd_soc_pcm_runtime *fe = fe_substream->private_data;
+	int ret;
 
-			dev_dbg(be->dev, "ASoC: BE digital mute %s\n",
-					 be->dai_link->name);
+	mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME);
+	ret = dpcm_fe_dai_shutdown(fe_substream);
 
-			if (drv->ops && drv->ops->digital_mute &&
-							dai->playback_active)
-				drv->ops->digital_mute(dai, mute);
-		}
-	}
+	dpcm_fe_dai_cleanup(fe_substream);
 
-	return 0;
+	mutex_unlock(&fe->card->mutex);
+	return ret;
 }
 
 static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream)
 {
 	struct snd_soc_pcm_runtime *fe = fe_substream->private_data;
-	struct snd_soc_dpcm *dpcm;
 	struct snd_soc_dapm_widget_list *list;
 	int ret;
 	int stream = fe_substream->stream;
···
 
 	ret = dpcm_path_get(fe, stream, &list);
 	if (ret < 0) {
-		mutex_unlock(&fe->card->mutex);
-		return ret;
+		goto open_end;
 	} else if (ret == 0) {
 		dev_dbg(fe->dev, "ASoC: %s no valid %s route\n",
 			fe->dai_link->name, stream ? "capture" : "playback");
···
 	dpcm_process_paths(fe, stream, &list, 1);
 
 	ret = dpcm_fe_dai_startup(fe_substream);
-	if (ret < 0) {
-		/* clean up all links */
-		for_each_dpcm_be(fe, stream, dpcm)
-			dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE;
-
-		dpcm_be_disconnect(fe, stream);
-		fe->dpcm[stream].runtime = NULL;
-	}
+	if (ret < 0)
+		dpcm_fe_dai_cleanup(fe_substream);
 
 	dpcm_clear_pending_state(fe, stream);
 	dpcm_path_put(&list);
-	mutex_unlock(&fe->card->mutex);
-	return ret;
-}
-
-static int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream)
-{
-	struct snd_soc_pcm_runtime *fe = fe_substream->private_data;
-	struct snd_soc_dpcm *dpcm;
-	int stream = fe_substream->stream, ret;
-
-	mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME);
-	ret = dpcm_fe_dai_shutdown(fe_substream);
-
-	/* mark FE's links ready to prune */
-	for_each_dpcm_be(fe, stream, dpcm)
-		dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE;
-
-	dpcm_be_disconnect(fe, stream);
-
-	fe->dpcm[stream].runtime = NULL;
+open_end:
 	mutex_unlock(&fe->card->mutex);
 	return ret;
 }
···
 int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
 {
 	struct snd_soc_dai *codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *cpu_dai;
 	struct snd_soc_component *component;
 	struct snd_pcm *pcm;
 	char new_name[64];
···
 		capture = rtd->dai_link->dpcm_capture;
 	} else {
 		/* Adapt stream for codec2codec links */
-		struct snd_soc_pcm_stream *cpu_capture = rtd->dai_link->params ?
-			&cpu_dai->driver->playback : &cpu_dai->driver->capture;
-		struct snd_soc_pcm_stream *cpu_playback = rtd->dai_link->params ?
-			&cpu_dai->driver->capture : &cpu_dai->driver->playback;
+		int cpu_capture = rtd->dai_link->params ?
+			SNDRV_PCM_STREAM_PLAYBACK : SNDRV_PCM_STREAM_CAPTURE;
+		int cpu_playback = rtd->dai_link->params ?
+			SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK;
 
-		for_each_rtd_codec_dai(rtd, i, codec_dai) {
+		for_each_rtd_codec_dais(rtd, i, codec_dai) {
+			if (rtd->num_cpus == 1) {
+				cpu_dai = rtd->cpu_dais[0];
+			} else if (rtd->num_cpus == rtd->num_codecs) {
+				cpu_dai = rtd->cpu_dais[i];
+			} else {
+				dev_err(rtd->card->dev,
+					"N cpus to M codecs link is not supported yet\n");
+				return -EINVAL;
+			}
+
 			if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_PLAYBACK) &&
-			    snd_soc_dai_stream_valid(cpu_dai,   SNDRV_PCM_STREAM_CAPTURE))
+			    snd_soc_dai_stream_valid(cpu_dai,   cpu_playback))
 				playback = 1;
 			if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_CAPTURE) &&
-			    snd_soc_dai_stream_valid(cpu_dai,   SNDRV_PCM_STREAM_PLAYBACK))
+			    snd_soc_dai_stream_valid(cpu_dai,   cpu_capture))
 				capture = 1;
 		}
-
-		capture = capture && cpu_capture->channels_min;
-		playback = playback && cpu_playback->channels_min;
 	}
 
 	if (rtd->dai_link->playback_only) {
···
 out:
 	dev_info(rtd->card->dev, "%s <-> %s mapping ok\n",
 		 (rtd->num_codecs > 1) ? "multicodec" : rtd->codec_dai->name,
-		 cpu_dai->name);
+		 (rtd->num_cpus > 1) ? "multicpu" : rtd->cpu_dai->name);
 	return ret;
 }
 
···
 }
 EXPORT_SYMBOL_GPL(snd_soc_dpcm_get_substream);
 
-/* get the BE runtime state */
-enum snd_soc_dpcm_state
-	snd_soc_dpcm_be_get_state(struct snd_soc_pcm_runtime *be, int stream)
-{
-	return be->dpcm[stream].state;
-}
-EXPORT_SYMBOL_GPL(snd_soc_dpcm_be_get_state);
-
-/* set the BE runtime state */
-void snd_soc_dpcm_be_set_state(struct snd_soc_pcm_runtime *be,
-		int stream, enum snd_soc_dpcm_state state)
-{
-	be->dpcm[stream].state = state;
-}
-EXPORT_SYMBOL_GPL(snd_soc_dpcm_be_set_state);
-
-/*
- * We can only hw_free, stop, pause or suspend a BE DAI if any of it's FE
- * are not running, paused or suspended for the specified stream direction.
- */
-int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe,
-		struct snd_soc_pcm_runtime *be, int stream)
+static int snd_soc_dpcm_check_state(struct snd_soc_pcm_runtime *fe,
+				    struct snd_soc_pcm_runtime *be,
+				    int stream,
+				    const enum snd_soc_dpcm_state *states,
+				    int num_states)
 {
 	struct snd_soc_dpcm *dpcm;
 	int state;
 	int ret = 1;
 	unsigned long flags;
+	int i;
 
 	spin_lock_irqsave(&fe->card->dpcm_lock, flags);
 	for_each_dpcm_fe(be, stream, dpcm) {
···
 			continue;
 
 		state = dpcm->fe->dpcm[stream].state;
-		if (state == SND_SOC_DPCM_STATE_START ||
-			state == SND_SOC_DPCM_STATE_PAUSED ||
-			state == SND_SOC_DPCM_STATE_SUSPEND) {
-			ret = 0;
-			break;
+		for (i = 0; i < num_states; i++) {
+			if (state == states[i]) {
+				ret = 0;
+				break;
+			}
 		}
 	}
 	spin_unlock_irqrestore(&fe->card->dpcm_lock, flags);
 
-	/* it's safe to free/stop this BE DAI */
+	/* it's safe to do this BE DAI */
 	return ret;
+}
+
+/*
+ * We can only hw_free, stop, pause or suspend a BE DAI if any of it's FE
+ * are not running, paused or suspended for the specified stream direction.
+ */
+int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe,
+		struct snd_soc_pcm_runtime *be, int stream)
+{
+	const enum snd_soc_dpcm_state state[] = {
+		SND_SOC_DPCM_STATE_START,
+		SND_SOC_DPCM_STATE_PAUSED,
+		SND_SOC_DPCM_STATE_SUSPEND,
+	};
+
+	return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state));
 }
 EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_free_stop);
 
···
 int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe,
 		struct snd_soc_pcm_runtime *be, int stream)
 {
-	struct snd_soc_dpcm *dpcm;
-	int state;
-	int ret = 1;
-	unsigned long flags;
+	const enum snd_soc_dpcm_state state[] = {
+		SND_SOC_DPCM_STATE_START,
+		SND_SOC_DPCM_STATE_PAUSED,
+		SND_SOC_DPCM_STATE_SUSPEND,
+		SND_SOC_DPCM_STATE_PREPARE,
+	};
 
-	spin_lock_irqsave(&fe->card->dpcm_lock, flags);
-	for_each_dpcm_fe(be, stream, dpcm) {
-
-		if (dpcm->fe == fe)
-			continue;
-
-		state = dpcm->fe->dpcm[stream].state;
-		if (state == SND_SOC_DPCM_STATE_START ||
-			state == SND_SOC_DPCM_STATE_PAUSED ||
-			state == SND_SOC_DPCM_STATE_SUSPEND ||
-			state == SND_SOC_DPCM_STATE_PREPARE) {
-			ret = 0;
-			break;
-		}
-	}
-	spin_unlock_irqrestore(&fe->card->dpcm_lock, flags);
-
-	/* it's safe to change hw_params */
-	return ret;
+	return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state));
 }
 EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_params);
-
-#ifdef CONFIG_DEBUG_FS
-static const char *dpcm_state_string(enum snd_soc_dpcm_state state)
-{
-	switch (state) {
-	case SND_SOC_DPCM_STATE_NEW:
-		return "new";
-	case SND_SOC_DPCM_STATE_OPEN:
-		return "open";
-	case SND_SOC_DPCM_STATE_HW_PARAMS:
-		return "hw_params";
-	case SND_SOC_DPCM_STATE_PREPARE:
-		return "prepare";
-	case SND_SOC_DPCM_STATE_START:
-		return "start";
-	case SND_SOC_DPCM_STATE_STOP:
-		return "stop";
-	case SND_SOC_DPCM_STATE_SUSPEND:
-		return "suspend";
-	case SND_SOC_DPCM_STATE_PAUSED:
-		return "paused";
-	case SND_SOC_DPCM_STATE_HW_FREE:
-		return "hw_free";
-	case SND_SOC_DPCM_STATE_CLOSE:
-		return "close";
-	}
-
-	return "unknown";
-}
-
-static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe,
-				int stream, char *buf, size_t size)
-{
-	struct snd_pcm_hw_params *params = &fe->dpcm[stream].hw_params;
-	struct snd_soc_dpcm *dpcm;
-	ssize_t offset = 0;
-	unsigned long flags;
-
-	/* FE state */
-	offset += scnprintf(buf + offset, size - offset,
-			"[%s - %s]\n", fe->dai_link->name,
-			stream ? "Capture" : "Playback");
-
-	offset += scnprintf(buf + offset, size - offset, "State: %s\n",
-	                dpcm_state_string(fe->dpcm[stream].state));
-
-	if ((fe->dpcm[stream].state >= SND_SOC_DPCM_STATE_HW_PARAMS) &&
-	    (fe->dpcm[stream].state <= SND_SOC_DPCM_STATE_STOP))
-		offset += scnprintf(buf + offset, size - offset,
-				"Hardware Params: "
-				"Format = %s, Channels = %d, Rate = %d\n",
-				snd_pcm_format_name(params_format(params)),
-				params_channels(params),
-				params_rate(params));
-
-	/* BEs state */
-	offset += scnprintf(buf + offset, size - offset, "Backends:\n");
-
-	if (list_empty(&fe->dpcm[stream].be_clients)) {
-		offset += scnprintf(buf + offset, size - offset,
-				" No active DSP links\n");
-		goto out;
-	}
-
-	spin_lock_irqsave(&fe->card->dpcm_lock, flags);
-	for_each_dpcm_be(fe, stream, dpcm) {
-		struct snd_soc_pcm_runtime *be = dpcm->be;
-		params = &dpcm->hw_params;
-
-		offset += scnprintf(buf + offset, size - offset,
-				"- %s\n", be->dai_link->name);
-
-		offset += scnprintf(buf + offset, size - offset,
-				"   State: %s\n",
-				dpcm_state_string(be->dpcm[stream].state));
-
-		if ((be->dpcm[stream].state >= SND_SOC_DPCM_STATE_HW_PARAMS) &&
-		    (be->dpcm[stream].state <= SND_SOC_DPCM_STATE_STOP))
-			offset += scnprintf(buf + offset, size - offset,
-				"   Hardware Params: "
-				"Format = %s, Channels = %d, Rate = %d\n",
-				snd_pcm_format_name(params_format(params)),
-				params_channels(params),
-				params_rate(params));
-	}
-	spin_unlock_irqrestore(&fe->card->dpcm_lock, flags);
-out:
-	return offset;
-}
-
-static ssize_t dpcm_state_read_file(struct file *file, char __user *user_buf,
-				size_t count, loff_t *ppos)
-{
-	struct snd_soc_pcm_runtime *fe = file->private_data;
-	ssize_t out_count = PAGE_SIZE, offset = 0, ret = 0;
-	char *buf;
-
-	buf = kmalloc(out_count, GFP_KERNEL);
-	if (!buf)
-		return -ENOMEM;
-
-	if (snd_soc_dai_stream_valid(fe->cpu_dai, SNDRV_PCM_STREAM_PLAYBACK))
-		offset += dpcm_show_state(fe, SNDRV_PCM_STREAM_PLAYBACK,
-					buf + offset, out_count - offset);
-
-	if (snd_soc_dai_stream_valid(fe->cpu_dai, SNDRV_PCM_STREAM_CAPTURE))
-		offset += dpcm_show_state(fe, SNDRV_PCM_STREAM_CAPTURE,
-					buf + offset, out_count - offset);
-
-	ret = simple_read_from_buffer(user_buf, count, ppos, buf, offset);
-
-	kfree(buf);
-	return ret;
-}
-
-static const struct file_operations dpcm_state_fops = {
-	.open = simple_open,
-	.read = dpcm_state_read_file,
-	.llseek = default_llseek,
-};
-
-void soc_dpcm_debugfs_add(struct snd_soc_pcm_runtime *rtd)
-{
-	if (!rtd->dai_link)
-		return;
-
-	if (!rtd->dai_link->dynamic)
-		return;
-
-	if (!rtd->card->debugfs_card_root)
-		return;
-
-	rtd->debugfs_dpcm_root = debugfs_create_dir(rtd->dai_link->name,
-			rtd->card->debugfs_card_root);
-
-	debugfs_create_file("state", 0444, rtd->debugfs_dpcm_root,
-			    rtd, &dpcm_state_fops);
-}
-#endif

sound/soc/soc-topology.c

···
 {
 	int ret = 0;
 
-	if (tplg->comp && tplg->ops && tplg->ops->vendor_load)
+	if (tplg->ops && tplg->ops->vendor_load)
 		ret = tplg->ops->vendor_load(tplg->comp, tplg->index, hdr);
 	else {
 		dev_err(tplg->dev, "ASoC: no vendor load callback for ID %d\n",
···
 static int soc_tplg_widget_load(struct soc_tplg *tplg,
 	struct snd_soc_dapm_widget *w, struct snd_soc_tplg_dapm_widget *tplg_w)
 {
-	if (tplg->comp && tplg->ops && tplg->ops->widget_load)
+	if (tplg->ops && tplg->ops->widget_load)
 		return tplg->ops->widget_load(tplg->comp, tplg->index, w,
 			tplg_w);
 
···
 static int soc_tplg_widget_ready(struct soc_tplg *tplg,
 	struct snd_soc_dapm_widget *w, struct snd_soc_tplg_dapm_widget *tplg_w)
 {
-	if (tplg->comp && tplg->ops && tplg->ops->widget_ready)
+	if (tplg->ops && tplg->ops->widget_ready)
 		return tplg->ops->widget_ready(tplg->comp, tplg->index, w,
 			tplg_w);
 
···
 	struct snd_soc_dai_driver *dai_drv,
 	struct snd_soc_tplg_pcm *pcm, struct snd_soc_dai *dai)
 {
-	if (tplg->comp && tplg->ops && tplg->ops->dai_load)
+	if (tplg->ops && tplg->ops->dai_load)
 		return tplg->ops->dai_load(tplg->comp, tplg->index, dai_drv,
 			pcm, dai);
 
···
 static int soc_tplg_dai_link_load(struct soc_tplg *tplg,
 	struct snd_soc_dai_link *link, struct snd_soc_tplg_link_config *cfg)
 {
-	if (tplg->comp && tplg->ops && tplg->ops->link_load)
+	if (tplg->ops && tplg->ops->link_load)
 		return tplg->ops->link_load(tplg->comp, tplg->index, link, cfg);
 
 	return 0;
···
 /* tell the component driver that all firmware has been loaded in this request */
 static void soc_tplg_complete(struct soc_tplg *tplg)
 {
-	if (tplg->comp && tplg->ops && tplg->ops->complete)
+	if (tplg->ops && tplg->ops->complete)
 		tplg->ops->complete(tplg->comp);
 }
 
···
 static int soc_tplg_init_kcontrol(struct soc_tplg *tplg,
 	struct snd_kcontrol_new *k, struct snd_soc_tplg_ctl_hdr *hdr)
 {
-	if (tplg->comp && tplg->ops && tplg->ops->control_load)
+	if (tplg->ops && tplg->ops->control_load)
 		return tplg->ops->control_load(tplg->comp, tplg->index, k,
 			hdr);
 
···
 static int soc_tplg_add_route(struct soc_tplg *tplg,
 	struct snd_soc_dapm_route *route)
 {
-	if (tplg->comp && tplg->ops && tplg->ops->dapm_route_load)
+	if (tplg->ops && tplg->ops->dapm_route_load)
 		return tplg->ops->dapm_route_load(tplg->comp, tplg->index,
 			route);
 
···
 	}
 
 	/* pass control to component driver for optional further init */
-	if (tplg->comp && tplg->ops && tplg->ops->manifest)
+	if (tplg->ops && tplg->ops->manifest)
 		ret = tplg->ops->manifest(tplg->comp, tplg->index, _manifest);
 
 	if (!abi_match)	/* free the duplicated one */
···
 	struct soc_tplg tplg;
 	int ret;
 
+	/* component needs to exist to keep and reference data while parsing */
+	if (!comp)
+		return -EINVAL;
+
 	/* setup parsing context */
 	memset(&tplg, 0, sizeof(tplg));
 	tplg.fw = fw;
···
 {
 	struct snd_soc_dapm_widget *w, *next_w;
 
-	list_for_each_entry_safe(w, next_w, &dapm->card->widgets, list) {
+	for_each_card_widgets_safe(dapm->card, w, next_w) {
 
 		/* make sure we are a widget with correct context */
 		if (w->dobj.type != SND_SOC_DOBJ_WIDGET || w->dapm != dapm)

sound/soc/sof/Kconfig

···
 	  required to enable i.MX8 devices.
 	  Say Y if you need this option. If unsure select "N".
 
+config SND_SOC_SOF_DEBUG_PROBES
+	bool "SOF enable data probing"
+	select SND_SOC_COMPRESS
+	help
+	  This option enables the data probing feature that can be used to
+	  gather data directly from specific points of the audio pipeline.
+	  Say Y if you want to enable probes.
+	  If unsure, select "N".
+
 config SND_SOC_SOF_DEVELOPER_SUPPORT
 	bool "SOF developer options support"
 	depends on EXPERT

sound/soc/sof/Makefile

···
 
 snd-sof-objs := core.o ops.o loader.o ipc.o pcm.o pm.o debug.o topology.o\
 		control.o trace.o utils.o sof-audio.o
+snd-sof-$(CONFIG_SND_SOC_SOF_DEBUG_PROBES) += probe.o compress.o
 
 snd-sof-pci-objs := sof-pci-dev.o
 snd-sof-acpi-objs := sof-acpi-dev.o

sound/soc/sof/compress.c

···
+// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause)
+//
+// This file is provided under a dual BSD/GPLv2 license.  When using or
+// redistributing this file, you may do so under either license.
+//
+// Copyright(c) 2019-2020 Intel Corporation. All rights reserved.
+//
+// Author: Cezary Rojewski <[e-mail]>
+//
+
+#include <sound/soc.h>
+#include "compress.h"
+#include "ops.h"
+#include "probe.h"
+
+struct snd_compr_ops sof_probe_compressed_ops = {
+	.copy		= sof_probe_compr_copy,
+};
+EXPORT_SYMBOL(sof_probe_compressed_ops);
+
+int sof_probe_compr_open(struct snd_compr_stream *cstream,
+		struct snd_soc_dai *dai)
+{
+	struct snd_sof_dev *sdev =
+				snd_soc_component_get_drvdata(dai->component);
+	int ret;
+
+	ret = snd_sof_probe_compr_assign(sdev, cstream, dai);
+	if (ret < 0) {
+		dev_err(dai->dev, "Failed to assign probe stream: %d\n", ret);
+		return ret;
+	}
+
+	sdev->extractor_stream_tag = ret;
+	return 0;
+}
+EXPORT_SYMBOL(sof_probe_compr_open);
+
+int sof_probe_compr_free(struct snd_compr_stream *cstream,
+		struct snd_soc_dai *dai)
+{
+	struct snd_sof_dev *sdev =
+				snd_soc_component_get_drvdata(dai->component);
+	struct sof_probe_point_desc *desc;
+	size_t num_desc;
+	int i, ret;
+
+	/* disconnect all probe points */
+	ret = sof_ipc_probe_points_info(sdev, &desc, &num_desc);
+	if (ret < 0) {
+		dev_err(dai->dev, "Failed to get probe points: %d\n", ret);
+		goto exit;
+	}
+
+	for (i = 0; i < num_desc; i++)
+		sof_ipc_probe_points_remove(sdev, &desc[i].buffer_id, 1);
+	kfree(desc);
+
+exit:
+	ret = sof_ipc_probe_deinit(sdev);
+	if (ret < 0)
+		dev_err(dai->dev, "Failed to deinit probe: %d\n", ret);
+
+	sdev->extractor_stream_tag = SOF_PROBE_INVALID_NODE_ID;
+	snd_compr_free_pages(cstream);
+
+	return snd_sof_probe_compr_free(sdev, cstream, dai);
+}
+EXPORT_SYMBOL(sof_probe_compr_free);
+
+int sof_probe_compr_set_params(struct snd_compr_stream *cstream,
+		struct snd_compr_params *params, struct snd_soc_dai *dai)
+{
+	struct snd_compr_runtime *rtd = cstream->runtime;
+	struct snd_sof_dev *sdev =
+				snd_soc_component_get_drvdata(dai->component);
+	int ret;
+
+	cstream->dma_buffer.dev.type = SNDRV_DMA_TYPE_DEV_SG;
+	cstream->dma_buffer.dev.dev = sdev->dev;
+	ret = snd_compr_malloc_pages(cstream, rtd->buffer_size);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_sof_probe_compr_set_params(sdev, cstream, params, dai);
+	if (ret < 0)
+		return ret;
+
+	ret = sof_ipc_probe_init(sdev, sdev->extractor_stream_tag,
+				 rtd->dma_bytes);
+	if (ret < 0) {
+		dev_err(dai->dev, "Failed to init probe: %d\n", ret);
+		return ret;
+	}
+
+	return 0;
+}
+EXPORT_SYMBOL(sof_probe_compr_set_params);
+
+int sof_probe_compr_trigger(struct snd_compr_stream *cstream, int cmd,
+		struct snd_soc_dai *dai)
+{
+	struct snd_sof_dev *sdev =
+				snd_soc_component_get_drvdata(dai->component);
+
+	return snd_sof_probe_compr_trigger(sdev, cstream, cmd, dai);
+}
+EXPORT_SYMBOL(sof_probe_compr_trigger);
+
+int sof_probe_compr_pointer(struct snd_compr_stream *cstream,
+		struct snd_compr_tstamp *tstamp, struct snd_soc_dai *dai)
+{
+	struct snd_sof_dev *sdev =
+				snd_soc_component_get_drvdata(dai->component);
+
+	return snd_sof_probe_compr_pointer(sdev, cstream, tstamp, dai);
+}
+EXPORT_SYMBOL(sof_probe_compr_pointer);
+
+int sof_probe_compr_copy(struct snd_compr_stream *cstream,
+		char __user *buf, size_t count)
+{
+	struct snd_compr_runtime *rtd = cstream->runtime;
+	unsigned int offset, n;
+	void *ptr;
+	int ret;
+
+	if (count > rtd->buffer_size)
+		count = rtd->buffer_size;
+
+	div_u64_rem(rtd->total_bytes_transferred, rtd->buffer_size, &offset);
+	ptr = rtd->dma_area + offset;
+	n = rtd->buffer_size - offset;
+
+	if (count < n) {
+		ret = copy_to_user(buf, ptr, count);
+	} else {
+		ret = copy_to_user(buf, ptr, n);
+		ret += copy_to_user(buf + n, rtd->dma_area, count - n);
+	}
+
+	if (ret)
+		return count - ret;
+	return count;
+}
+EXPORT_SYMBOL(sof_probe_compr_copy);

sound/soc/sof/compress.h

···
+/* SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) */
+/*
+ * This file is provided under a dual BSD/GPLv2 license.  When using or
+ * redistributing this file, you may do so under either license.
+ *
+ * Copyright(c) 2019-2020 Intel Corporation. All rights reserved.
+ *
+ * Author: Cezary Rojewski <[e-mail]>
+ */
+
+#ifndef __SOF_COMPRESS_H
+#define __SOF_COMPRESS_H
+
+#include <sound/compress_driver.h>
+
+extern struct snd_compr_ops sof_probe_compressed_ops;
+
+int sof_probe_compr_open(struct snd_compr_stream *cstream,
+		struct snd_soc_dai *dai);
+int sof_probe_compr_free(struct snd_compr_stream *cstream,
+		struct snd_soc_dai *dai);
+int sof_probe_compr_set_params(struct snd_compr_stream *cstream,
+		struct snd_compr_params *params, struct snd_soc_dai *dai);
+int sof_probe_compr_trigger(struct snd_compr_stream *cstream, int cmd,
+		struct snd_soc_dai *dai);
+int sof_probe_compr_pointer(struct snd_compr_stream *cstream,
+		struct snd_compr_tstamp *tstamp, struct snd_soc_dai *dai);
+int sof_probe_compr_copy(struct snd_compr_stream *cstream,
+		char __user *buf, size_t count);
+
+#endif

sound/soc/sof/core.c

···
 #include <sound/sof.h>
 #include "sof-priv.h"
 #include "ops.h"
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES)
+#include "probe.h"
+#endif
 
 /* see SOF_DBG_ flags */
 int sof_core_debug;
···
 	/* initialize sof device */
 	sdev->dev = dev;
 
-	/* initialize default D0 sub-state */
-	sdev->d0_substate = SOF_DSP_D0I0;
+	/* initialize default DSP power state */
+	sdev->dsp_power_state.state = SOF_DSP_PM_D0;
 
 	sdev->pdata = plat_data;
 	sdev->first_boot = true;
 	sdev->fw_state = SOF_FW_BOOT_NOT_STARTED;
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES)
+	sdev->extractor_stream_tag = SOF_PROBE_INVALID_NODE_ID;
+#endif
 	dev_set_drvdata(dev, sdev);
 
 	/* check all mandatory ops */

sound/soc/sof/debug.c

···
 #include "sof-priv.h"
 #include "ops.h"
 
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES)
+#include "probe.h"
+
+/**
+ * strsplit_u32 - Split string into sequence of u32 tokens
+ * @buf:	String to split into tokens.
+ * @delim:	String containing delimiter characters.
+ * @tkns:	Returned u32 sequence pointer.
+ * @num_tkns:	Returned number of tokens obtained.
+ */
+static int
+strsplit_u32(char **buf, const char *delim, u32 **tkns, size_t *num_tkns)
+{
+	char *s;
+	u32 *data, *tmp;
+	size_t count = 0;
+	size_t cap = 32;
+	int ret = 0;
+
+	*tkns = NULL;
+	*num_tkns = 0;
+	data = kcalloc(cap, sizeof(*data), GFP_KERNEL);
+	if (!data)
+		return -ENOMEM;
+
+	while ((s = strsep(buf, delim)) != NULL) {
+		ret = kstrtouint(s, 0, data + count);
+		if (ret)
+			goto exit;
+		if (++count >= cap) {
+			cap *= 2;
+			tmp = krealloc(data, cap * sizeof(*data), GFP_KERNEL);
+			if (!tmp) {
+				ret = -ENOMEM;
+				goto exit;
+			}
+			data = tmp;
+		}
+	}
+
+	if (!count)
+		goto exit;
+	*tkns = kmemdup(data, count * sizeof(*data), GFP_KERNEL);
+	if (*tkns == NULL) {
+		ret = -ENOMEM;
+		goto exit;
+	}
+	*num_tkns = count;
+
+exit:
+	kfree(data);
+	return ret;
+}
+
+static int tokenize_input(const char __user *from, size_t count,
+		loff_t *ppos, u32 **tkns, size_t *num_tkns)
+{
+	char *buf;
+	int ret;
+
+	buf = kmalloc(count + 1, GFP_KERNEL);
+	if (!buf)
+		return -ENOMEM;
+
+	ret = simple_write_to_buffer(buf, count, ppos, from, count);
+	if (ret != count) {
+		ret = ret >= 0 ? -EIO : ret;
+		goto exit;
+	}
+
+	buf[count] = '\0';
+	ret = strsplit_u32((char **)&buf, ",", tkns, num_tkns);
+exit:
+	kfree(buf);
+	return ret;
+}
+
+static ssize_t probe_points_read(struct file *file,
+		char __user *to, size_t count, loff_t *ppos)
+{
+	struct snd_sof_dfsentry *dfse = file->private_data;
+	struct snd_sof_dev *sdev = dfse->sdev;
+	struct sof_probe_point_desc *desc;
+	size_t num_desc, len = 0;
+	char *buf;
+	int i, ret;
+
+	if (sdev->extractor_stream_tag == SOF_PROBE_INVALID_NODE_ID) {
+		dev_warn(sdev->dev, "no extractor stream running\n");
+		return -ENOENT;
+	}
+
+	buf = kzalloc(PAGE_SIZE, GFP_KERNEL);
+	if (!buf)
+		return -ENOMEM;
+
+	ret = sof_ipc_probe_points_info(sdev, &desc, &num_desc);
+	if (ret < 0)
+		goto exit;
+
+	for (i = 0; i < num_desc; i++) {
+		ret = snprintf(buf + len, PAGE_SIZE - len,
+			"Id: %#010x  Purpose: %d  Node id: %#x\n",
+			desc[i].buffer_id, desc[i].purpose, desc[i].stream_tag);
+		if (ret < 0)
+			goto free_desc;
+		len += ret;
+	}
+
+	ret = simple_read_from_buffer(to, count, ppos, buf, len);
+free_desc:
+	kfree(desc);
+exit:
+	kfree(buf);
+	return ret;
+}
+
+static ssize_t probe_points_write(struct file *file,
+		const char __user *from, size_t count, loff_t *ppos)
+{
+	struct snd_sof_dfsentry *dfse = file->private_data;
+	struct snd_sof_dev *sdev = dfse->sdev;
+	struct sof_probe_point_desc *desc;
+	size_t num_tkns, bytes;
+	u32 *tkns;
+	int ret;
+
+	if (sdev->extractor_stream_tag == SOF_PROBE_INVALID_NODE_ID) {
+		dev_warn(sdev->dev, "no extractor stream running\n");
+		return -ENOENT;
+	}
+
+	ret = tokenize_input(from, count, ppos, &tkns, &num_tkns);
+	if (ret < 0)
+		return ret;
+	bytes = sizeof(*tkns) * num_tkns;
+	if (!num_tkns || (bytes % sizeof(*desc))) {
+		ret = -EINVAL;
+		goto exit;
+	}
+
+	desc = (struct sof_probe_point_desc *)tkns;
+	ret = sof_ipc_probe_points_add(sdev,
+			desc, bytes / sizeof(*desc));
+	if (!ret)
+		ret = count;
+exit:
+	kfree(tkns);
+	return ret;
+}
+
+static const struct file_operations probe_points_fops = {
+	.open = simple_open,
+	.read = probe_points_read,
+	.write = probe_points_write,
+	.llseek = default_llseek,
+};
+
+static ssize_t probe_points_remove_write(struct file *file,
+		const char __user *from, size_t count, loff_t *ppos)
+{
+	struct snd_sof_dfsentry *dfse = file->private_data;
+	struct snd_sof_dev *sdev = dfse->sdev;
+	size_t num_tkns;
+	u32 *tkns;
+	int ret;
+
+	if (sdev->extractor_stream_tag == SOF_PROBE_INVALID_NODE_ID) {
+		dev_warn(sdev->dev, "no extractor stream running\n");
+		return -ENOENT;
+	}
+
+	ret = tokenize_input(from, count, ppos, &tkns, &num_tkns);
+	if (ret < 0)
+		return ret;
+	if (!num_tkns) {
+		ret = -EINVAL;
+		goto exit;
+	}
+
+	ret = sof_ipc_probe_points_remove(sdev, tkns, num_tkns);
+	if (!ret)
+		ret = count;
+exit:
+	kfree(tkns);
+	return ret;
+}
+
+static const struct file_operations probe_points_remove_fops = {
+	.open = simple_open,
+	.write = probe_points_remove_write,
+	.llseek = default_llseek,
+};
+
+static int snd_sof_debugfs_probe_item(struct snd_sof_dev *sdev,
+				 const char *name, mode_t mode,
+				 const struct file_operations *fops)
+{
+	struct snd_sof_dfsentry *dfse;
+
+	dfse = devm_kzalloc(sdev->dev, sizeof(*dfse), GFP_KERNEL);
+	if (!dfse)
+		return -ENOMEM;
+
+	dfse->type = SOF_DFSENTRY_TYPE_BUF;
+	dfse->sdev = sdev;
+
+	debugfs_create_file(name, mode, sdev->debugfs_root, dfse, fops);
+	/* add to dfsentry list */
+	list_add(&dfse->list, &sdev->dfsentry_list);
+
+	return 0;
+}
+#endif
+
 #if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_IPC_FLOOD_TEST)
 #define MAX_IPC_FLOOD_DURATION_MS 1000
 #define MAX_IPC_FLOOD_COUNT 10000
···
 		if (err < 0)
 			return err;
 	}
+
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES)
+	err = snd_sof_debugfs_probe_item(sdev, "probe_points",
+			0644, &probe_points_fops);
+	if (err < 0)
+		return err;
+	err = snd_sof_debugfs_probe_item(sdev, "probe_points_remove",
+			0200, &probe_points_remove_fops);
+	if (err < 0)
+		return err;
+#endif
 
 #if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_IPC_FLOOD_TEST)
 	/* create read-write ipc_flood_count debugfs entry */

sound/soc/sof/imx/imx8.c

···
 /*
  * DSP control.
  */
-static int imx8_run(struct snd_sof_dev *sdev)
+static int imx8x_run(struct snd_sof_dev *sdev)
 {
 	struct imx8_priv *dsp_priv = (struct imx8_priv *)sdev->private;
 	int ret;
···
 				      IMX_SC_C_OFS_IRQ, 0x51);
 	if (ret < 0) {
 		dev_err(sdev->dev, "Error system address offset of IRQ\n");
+		return ret;
+	}
+
+	imx_sc_pm_cpu_start(dsp_priv->sc_ipc, IMX_SC_R_DSP, true,
+			    RESET_VECTOR_VADDR);
+
+	return 0;
+}
+
+static int imx8_run(struct snd_sof_dev *sdev)
+{
+	struct imx8_priv *dsp_priv = (struct imx8_priv *)sdev->private;
+	int ret;
+
+	ret = imx_sc_misc_set_control(dsp_priv->sc_ipc, IMX_SC_R_DSP,
+				      IMX_SC_C_OFS_SEL, 0);
+	if (ret < 0) {
+		dev_err(sdev->dev, "Error system address offset source select\n");
 		return ret;
 	}
 
···
 },
 };
 
-/* i.MX8  ops */
+/* i.MX8 ops */
 struct snd_sof_dsp_ops sof_imx8_ops = {
 	/* probe and remove */
 	.probe		= imx8_probe,
 	.remove		= imx8_remove,
 	/* DSP core boot */
 	.run		= imx8_run,
+
+	/* Block IO */
+	.block_read	= sof_block_read,
+	.block_write	= sof_block_write,
+
+	/* ipc */
+	.send_msg	= imx8_send_msg,
+	.fw_ready	= sof_fw_ready,
+	.get_mailbox_offset	= imx8_get_mailbox_offset,
+	.get_window_offset	= imx8_get_window_offset,
+
+	.ipc_msg_data	= imx8_ipc_msg_data,
+	.ipc_pcm_params	= imx8_ipc_pcm_params,
+
+	/* module loading */
+	.load_module	= snd_sof_parse_module_memcpy,
+	.get_bar_index	= imx8_get_bar_index,
+	/* firmware loading */
+	.load_firmware	= snd_sof_load_firmware_memcpy,
+
+	/* DAI drivers */
+	.drv = imx8_dai,
+	.num_drv = 1, /* we have only 1 ESAI interface on i.MX8 */
+};
+EXPORT_SYMBOL(sof_imx8_ops);
+
+/* i.MX8X ops */
+struct snd_sof_dsp_ops sof_imx8x_ops = {
+	/* probe and remove */
+	.probe		= imx8_probe,
+	.remove		= imx8_remove,
+	/* DSP core boot */
+	.run		= imx8x_run,
 
 	/* Block IO */
 	.block_read	= sof_block_read,
···
 			SNDRV_PCM_INFO_PAUSE |
 			SNDRV_PCM_INFO_NO_PERIOD_WAKEUP
 };
-EXPORT_SYMBOL(sof_imx8_ops);
+EXPORT_SYMBOL(sof_imx8x_ops);
 
 MODULE_LICENSE("Dual BSD/GPL");

sound/soc/sof/intel/Kconfig

···
 	  Say Y if you want to enable HDAudio codecs with SOF.
 	  If unsure select "N".
 
+config SND_SOC_SOF_HDA_PROBES
+	bool "SOF enable probes over HDA"
+	depends on SND_SOC_SOF_DEBUG_PROBES
+	help
+	  This option enables the data probing for Intel(R).
+		  Intel(R) Skylake and newer platforms.
+	  Say Y if you want to enable probes.
+	  If unsure, select "N".
+
 config SND_SOC_SOF_HDA_ALWAYS_ENABLE_DMI_L1
 	bool "SOF enable DMI Link L1"
 	help
···
 	  affected by DMI L1 issues. This option is not recommended.
 	  Say Y if you want to enable DMI Link L1
 	  If unsure, select "N".
-
-config SND_SOC_SOF_HDA_COMMON_HDMI_CODEC
-	bool "SOF common HDA HDMI codec driver"
-	depends on SND_SOC_SOF_HDA_LINK
-	depends on SND_HDA_CODEC_HDMI
-	default SND_HDA_CODEC_HDMI
-	help
-	  This adds support for HDMI audio by using the common HDA
-	  HDMI/DisplayPort codec driver.
-	  Say Y if you want to use the common codec driver with SOF.
-	  If unsure select "Y".
 
 endif ## SND_SOC_SOF_HDA_COMMON
 

sound/soc/sof/intel/Makefile

···
 				 hda-dsp.o hda-ipc.o hda-ctrl.o hda-pcm.o \
 				 hda-dai.o hda-bus.o \
 				 apl.o cnl.o
+snd-sof-intel-hda-common-$(CONFIG_SND_SOC_SOF_HDA_PROBES) += hda-compress.o
 
 snd-sof-intel-hda-objs := hda-codec.o
 

sound/soc/sof/intel/apl.c

···
 	.pcm_trigger	= hda_dsp_pcm_trigger,
 	.pcm_pointer	= hda_dsp_pcm_pointer,
 
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_PROBES)
+	/* probe callbacks */
+	.probe_assign	= hda_probe_compr_assign,
+	.probe_free	= hda_probe_compr_free,
+	.probe_set_params	= hda_probe_compr_set_params,
+	.probe_trigger	= hda_probe_compr_trigger,
+	.probe_pointer	= hda_probe_compr_pointer,
+#endif
+
 	/* firmware loading */
 	.load_firmware = snd_sof_load_firmware_raw,
 

sound/soc/sof/intel/cnl.c

···
 		hda_dsp_ipc_get_reply(sdev);
 		snd_sof_ipc_reply(sdev, msg);
 
-		if (sdev->code_loading)	{
-			sdev->code_loading = 0;
-			wake_up(&sdev->waitq);
-		}
-
 		cnl_ipc_dsp_done(sdev);
 
 		spin_unlock_irq(&sdev->ipc_lock);
···
 static int cnl_ipc_send_msg(struct snd_sof_dev *sdev,
 			    struct snd_sof_ipc_msg *msg)
 {
+	struct sof_intel_hda_dev *hdev = sdev->pdata->hw_pdata;
+	struct sof_ipc_cmd_hdr *hdr;
 	u32 dr = 0;
 	u32 dd = 0;
 
+	/*
+	 * Currently the only compact IPC supported is the PM_GATE
+	 * IPC which is used for transitioning the DSP between the
+	 * D0I0 and D0I3 states. And these are sent only during the
+	 * set_power_state() op. Therefore, there will never be a case
+	 * that a compact IPC results in the DSP exiting D0I3 without
+	 * the host and FW being in sync.
+	 */
 	if (cnl_compact_ipc_compress(msg, &dr, &dd)) {
 		/* send the message via IPC registers */
 		snd_sof_dsp_write(sdev, HDA_DSP_BAR, CNL_DSP_REG_HIPCIDD,
 				  dd);
 		snd_sof_dsp_write(sdev, HDA_DSP_BAR, CNL_DSP_REG_HIPCIDR,
 				  CNL_DSP_REG_HIPCIDR_BUSY | dr);
-	} else {
-		/* send the message via mailbox */
-		sof_mailbox_write(sdev, sdev->host_box.offset, msg->msg_data,
-				  msg->msg_size);
-		snd_sof_dsp_write(sdev, HDA_DSP_BAR, CNL_DSP_REG_HIPCIDR,
-				  CNL_DSP_REG_HIPCIDR_BUSY);
+		return 0;
 	}
+
+	/* send the message via mailbox */
+	sof_mailbox_write(sdev, sdev->host_box.offset, msg->msg_data,
+			  msg->msg_size);
+	snd_sof_dsp_write(sdev, HDA_DSP_BAR, CNL_DSP_REG_HIPCIDR,
+			  CNL_DSP_REG_HIPCIDR_BUSY);
+
+	hdr = msg->msg_data;
+
+	/*
+	 * Use mod_delayed_work() to schedule the delayed work
+	 * to avoid scheduling multiple workqueue items when
+	 * IPCs are sent at a high-rate. mod_delayed_work()
+	 * modifies the timer if the work is pending.
+	 * Also, a new delayed work should not be queued after the
+	 * the CTX_SAVE IPC, which is sent before the DSP enters D3.
+	 */
+	if (hdr->cmd != (SOF_IPC_GLB_PM_MSG | SOF_IPC_PM_CTX_SAVE))
+		mod_delayed_work(system_wq, &hdev->d0i3_work,
+				 msecs_to_jiffies(SOF_HDA_D0I3_WORK_DELAY_MS));
 
 	return 0;
 }
···
 	.pcm_hw_free	= hda_dsp_stream_hw_free,
 	.pcm_trigger	= hda_dsp_pcm_trigger,
 	.pcm_pointer	= hda_dsp_pcm_pointer,
+
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_PROBES)
+	/* probe callbacks */
+	.probe_assign	= hda_probe_compr_assign,
+	.probe_free	= hda_probe_compr_free,
+	.probe_set_params	= hda_probe_compr_set_params,
+	.probe_trigger	= hda_probe_compr_trigger,
+	.probe_pointer	= hda_probe_compr_pointer,
+#endif
 
 	/* firmware loading */
 	.load_firmware = snd_sof_load_firmware_raw,

sound/soc/sof/intel/hda-codec.c

···
 	if (ret < 0)
 		return ret;
 
-	if ((resp & 0xFFFF0000) == IDISP_VID_INTEL)
+	if ((resp & 0xFFFF0000) == IDISP_VID_INTEL) {
+		if (!hdev->bus->audio_component) {
+			dev_dbg(sdev->dev,
+				"iDisp hw present but no driver\n");
+			return -ENOENT;
+		}
 		hda_priv->need_display_power = true;
+	}
 
 	/*
 	 * if common HDMI codec driver is not used, codec load
···
 {
 	struct hdac_bus *bus = sof_to_bus(sdev);
 	int ret;
+
+	if (!bus->audio_component)
+		return 0;
 
 	/* power down unconditionally */
 	snd_hdac_display_power(bus, HDA_CODEC_IDX_CONTROLLER, false);

sound/soc/sof/intel/hda-compress.c

···
+// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause)
+//
+// This file is provided under a dual BSD/GPLv2 license.  When using or
+// redistributing this file, you may do so under either license.
+//
+// Copyright(c) 2019-2020 Intel Corporation. All rights reserved.
+//
+// Author: Cezary Rojewski <[e-mail]>
+//
+
+#include <sound/hdaudio_ext.h>
+#include <sound/soc.h>
+#include "../sof-priv.h"
+#include "hda.h"
+
+static inline struct hdac_ext_stream *
+hda_compr_get_stream(struct snd_compr_stream *cstream)
+{
+	return cstream->runtime->private_data;
+}
+
+int hda_probe_compr_assign(struct snd_sof_dev *sdev,
+			   struct snd_compr_stream *cstream,
+			   struct snd_soc_dai *dai)
+{
+	struct hdac_ext_stream *stream;
+
+	stream = hda_dsp_stream_get(sdev, cstream->direction);
+	if (!stream)
+		return -EBUSY;
+
+	hdac_stream(stream)->curr_pos = 0;
+	hdac_stream(stream)->cstream = cstream;
+	cstream->runtime->private_data = stream;
+
+	return hdac_stream(stream)->stream_tag;
+}
+
+int hda_probe_compr_free(struct snd_sof_dev *sdev,
+			 struct snd_compr_stream *cstream,
+			 struct snd_soc_dai *dai)
+{
+	struct hdac_ext_stream *stream = hda_compr_get_stream(cstream);
+	int ret;
+
+	ret = hda_dsp_stream_put(sdev, cstream->direction,
+				 hdac_stream(stream)->stream_tag);
+	if (ret < 0) {
+		dev_dbg(sdev->dev, "stream put failed: %d\n", ret);
+		return ret;
+	}
+
+	hdac_stream(stream)->cstream = NULL;
+	cstream->runtime->private_data = NULL;
+
+	return 0;
+}
+
+int hda_probe_compr_set_params(struct snd_sof_dev *sdev,
+			       struct snd_compr_stream *cstream,
+			       struct snd_compr_params *params,
+			       struct snd_soc_dai *dai)
+{
+	struct hdac_ext_stream *stream = hda_compr_get_stream(cstream);
+	struct hdac_stream *hstream = hdac_stream(stream);
+	struct snd_dma_buffer *dmab;
+	u32 bits, rate;
+	int bps, ret;
+
+	dmab = cstream->runtime->dma_buffer_p;
+	/* compr params do not store bit depth, default to S32_LE */
+	bps = snd_pcm_format_physical_width(SNDRV_PCM_FORMAT_S32_LE);
+	if (bps < 0)
+		return bps;
+	bits = hda_dsp_get_bits(sdev, bps);
+	rate = hda_dsp_get_mult_div(sdev, params->codec.sample_rate);
+
+	hstream->format_val = rate | bits | (params->codec.ch_out - 1);
+	hstream->bufsize = cstream->runtime->buffer_size;
+	hstream->period_bytes = cstream->runtime->fragment_size;
+	hstream->no_period_wakeup = 0;
+
+	ret = hda_dsp_stream_hw_params(sdev, stream, dmab, NULL);
+	if (ret < 0) {
+		dev_err(sdev->dev, "error: hdac prepare failed: %x\n", ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+int hda_probe_compr_trigger(struct snd_sof_dev *sdev,
+			    struct snd_compr_stream *cstream, int cmd,
+			    struct snd_soc_dai *dai)
+{
+	struct hdac_ext_stream *stream = hda_compr_get_stream(cstream);
+
+	return hda_dsp_stream_trigger(sdev, stream, cmd);
+}
+
+int hda_probe_compr_pointer(struct snd_sof_dev *sdev,
+			    struct snd_compr_stream *cstream,
+			    struct snd_compr_tstamp *tstamp,
+			    struct snd_soc_dai *dai)
+{
+	struct hdac_ext_stream *stream = hda_compr_get_stream(cstream);
+	struct snd_soc_pcm_stream *pstream;
+
+	pstream = &dai->driver->capture;
+	tstamp->copied_total = hdac_stream(stream)->curr_pos;
+	tstamp->sampling_rate = snd_pcm_rate_bit_to_rate(pstream->rates);
+
+	return 0;
+}

sound/soc/sof/intel/hda-ctrl.c

···
 #include <linux/module.h>
 #include <sound/hdaudio_ext.h>
 #include <sound/hda_register.h>
+#include <sound/hda_component.h>
 #include "../ops.h"
 #include "hda.h"
 
···
 	struct hdac_bus *bus = sof_to_bus(sdev);
 	u32 cap, offset, feature;
 	int count = 0;
+	int ret;
+
+	/*
+	 * On some devices, one reset cycle is necessary before reading
+	 * capabilities
+	 */
+	ret = hda_dsp_ctrl_link_reset(sdev, true);
+	if (ret < 0)
+		return ret;
+	ret = hda_dsp_ctrl_link_reset(sdev, false);
+	if (ret < 0)
+		return ret;
 
 	offset = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, SOF_HDA_LLCH);
 
 	do {
-		cap = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, offset);
-
 		dev_dbg(sdev->dev, "checking for capabilities at offset 0x%x\n",
 			offset & SOF_HDA_CAP_NEXT_MASK);
+
+		cap = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, offset);
+
+		if (cap == -1) {
+			dev_dbg(bus->dev, "Invalid capability reg read\n");
+			break;
+		}
 
 		feature = (cap & SOF_HDA_CAP_ID_MASK) >> SOF_HDA_CAP_ID_OFF;
 
···
 			bus->mlcap = bus->remap_addr + offset;
 			break;
 		default:
-			dev_vdbg(sdev->dev, "found capability %d at 0x%x\n",
-				 feature, offset);
+			dev_dbg(sdev->dev, "found capability %d at 0x%x\n",
+				feature, offset);
 			break;
 		}
 
···
 	if (bus->chip_init)
 		return 0;
 
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA)
+	snd_hdac_set_codec_wakeup(bus, true);
+#endif
 	hda_dsp_ctrl_misc_clock_gating(sdev, false);
 
 	if (full_reset) {
···
 		ret = hda_dsp_ctrl_link_reset(sdev, true);
 		if (ret < 0) {
 			dev_err(sdev->dev, "error: failed to reset HDA controller\n");
-			return ret;
+			goto err;
 		}
 
 		usleep_range(500, 1000);
···
 		ret = hda_dsp_ctrl_link_reset(sdev, false);
 		if (ret < 0) {
 			dev_err(sdev->dev, "error: failed to exit HDA controller reset\n");
-			return ret;
+			goto err;
 		}
 
 		usleep_range(1000, 1200);
···
 	/* check to see if controller is ready */
 	if (!snd_hdac_chip_readb(bus, GCTL)) {
 		dev_dbg(bus->dev, "controller not ready!\n");
-		return -EBUSY;
+		ret = -EBUSY;
+		goto err;
 	}
 
 	/* Accept unsolicited responses */
···
 
 	bus->chip_init = true;
 
+err:
 	hda_dsp_ctrl_misc_clock_gating(sdev, true);
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA)
+	snd_hdac_set_codec_wakeup(bus, false);
+#endif
 
 	return ret;
 }

sound/soc/sof/intel/hda-dai.c

···
 	struct hdac_bus *bus = hstream->bus;
 	struct hdac_ext_stream *link_dev;
 	struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream);
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	struct sof_intel_hda_stream *hda_stream;
 	struct hda_pipe_params p_params = {0};
 	struct hdac_ext_link *link;
···
 	bus = hstream->bus;
 	rtd = snd_pcm_substream_chip(substream);
 
-	link = snd_hdac_ext_bus_get_link(bus, rtd->codec_dai->component->name);
+	link = snd_hdac_ext_bus_get_link(bus, asoc_rtd_to_codec(rtd, 0)->component->name);
 	if (!link)
 		return -EINVAL;
 
···
 	if (ret < 0)
 		return ret;
 
-	link = snd_hdac_ext_bus_get_link(bus, rtd->codec_dai->component->name);
+	link = snd_hdac_ext_bus_get_link(bus, asoc_rtd_to_codec(rtd, 0)->component->name);
 	if (!link)
 		return -EINVAL;
 
···
 	.trigger = hda_link_pcm_trigger,
 	.prepare = hda_link_pcm_prepare,
 };
+
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_PROBES)
+#include "../compress.h"
+
+static struct snd_soc_cdai_ops sof_probe_compr_ops = {
+	.startup	= sof_probe_compr_open,
+	.shutdown	= sof_probe_compr_free,
+	.set_params	= sof_probe_compr_set_params,
+	.trigger	= sof_probe_compr_trigger,
+	.pointer	= sof_probe_compr_pointer,
+};
+
+#endif
 #endif
 
 /*
···
 struct snd_soc_dai_driver skl_dai[] = {
 {
 	.name = "SSP0 Pin",
+	.playback = {
+		.channels_min = 1,
+		.channels_max = 8,
+	},
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 8,
+	},
 },
 {
 	.name = "SSP1 Pin",
+	.playback = {
+		.channels_min = 1,
+		.channels_max = 8,
+	},
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 8,
+	},
 },
 {
 	.name = "SSP2 Pin",
+	.playback = {
+		.channels_min = 1,
+		.channels_max = 8,
+	},
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 8,
+	},
 },
 {
 	.name = "SSP3 Pin",
+	.playback = {
+		.channels_min = 1,
+		.channels_max = 8,
+	},
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 8,
+	},
 },
 {
 	.name = "SSP4 Pin",
+	.playback = {
+		.channels_min = 1,
+		.channels_max = 8,
+	},
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 8,
+	},
 },
 {
 	.name = "SSP5 Pin",
+	.playback = {
+		.channels_min = 1,
+		.channels_max = 8,
+	},
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 8,
+	},
 },
 {
 	.name = "DMIC01 Pin",
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 4,
+	},
 },
 {
 	.name = "DMIC16k Pin",
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 4,
+	},
 },
 #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA)
 {
 	.name = "iDisp1 Pin",
 	.ops = &hda_link_dai_ops,
+	.playback = {
+		.channels_min = 1,
+		.channels_max = 8,
+	},
 },
 {
 	.name = "iDisp2 Pin",
 	.ops = &hda_link_dai_ops,
+	.playback = {
+		.channels_min = 1,
+		.channels_max = 8,
+	},
 },
 {
 	.name = "iDisp3 Pin",
 	.ops = &hda_link_dai_ops,
+	.playback = {
+		.channels_min = 1,
+		.channels_max = 8,
+	},
 },
 {
 	.name = "iDisp4 Pin",
 	.ops = &hda_link_dai_ops,
+	.playback = {
+		.channels_min = 1,
+		.channels_max = 8,
+	},
 },
 {
 	.name = "Analog CPU DAI",
 	.ops = &hda_link_dai_ops,
+	.playback = {
+		.channels_min = 1,
+		.channels_max = 16,
+	},
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 16,
+	},
 },
 {
 	.name = "Digital CPU DAI",
 	.ops = &hda_link_dai_ops,
+	.playback = {
+		.channels_min = 1,
+		.channels_max = 16,
+	},
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 16,
+	},
 },
 {
 	.name = "Alt Analog CPU DAI",
 	.ops = &hda_link_dai_ops,
+	.playback = {
+		.channels_min = 1,
+		.channels_max = 16,
+	},
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 16,
+	},
 },
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_PROBES)
+{
+	.name = "Probe Extraction CPU DAI",
+	.compress_new = snd_soc_new_compress,
+	.cops = &sof_probe_compr_ops,
+	.capture = {
+		.stream_name = "Probe Extraction",
+		.channels_min = 1,
+		.channels_max = 8,
+		.rates = SNDRV_PCM_RATE_48000,
+		.rate_min = 48000,
+		.rate_max = 48000,
+	},
+},
+#endif
 #endif
 };

sound/soc/sof/intel/hda-dsp.c

···
  * Hardware interface for generic Intel audio DSP HDA IP
  */
 
+#include <linux/module.h>
 #include <sound/hdaudio_ext.h>
 #include <sound/hda_register.h>
+#include "../sof-audio.h"
 #include "../ops.h"
 #include "hda.h"
 #include "hda-ipc.h"
+
+static bool hda_enable_trace_D0I3_S0;
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG)
+module_param_named(enable_trace_D0I3_S0, hda_enable_trace_D0I3_S0, bool, 0444);
+MODULE_PARM_DESC(enable_trace_D0I3_S0,
+		 "SOF HDA enable trace when the DSP is in D0I3 in S0");
+#endif
 
 /*
  * DSP Core control.
···
 	pm_gate.flags = flags;
 
 	/* send pm_gate ipc to dsp */
-	return sof_ipc_tx_message(sdev->ipc, pm_gate.hdr.cmd, &pm_gate,
-				  sizeof(pm_gate), &reply, sizeof(reply));
+	return sof_ipc_tx_message_no_pm(sdev->ipc, pm_gate.hdr.cmd,
+					&pm_gate, sizeof(pm_gate), &reply,
+					sizeof(reply));
 }
 
-int hda_dsp_set_power_state(struct snd_sof_dev *sdev,
-			    enum sof_d0_substate d0_substate)
+static int hda_dsp_update_d0i3c_register(struct snd_sof_dev *sdev, u8 value)
 {
 	struct hdac_bus *bus = sof_to_bus(sdev);
-	u32 flags;
 	int ret;
-	u8 value;
 
 	/* Write to D0I3C after Command-In-Progress bit is cleared */
 	ret = hda_dsp_wait_d0i3c_done(sdev);
···
 	}
 
 	/* Update D0I3C register */
-	value = d0_substate == SOF_DSP_D0I3 ? SOF_HDA_VS_D0I3C_I3 : 0;
 	snd_hdac_chip_updateb(bus, VS_D0I3C, SOF_HDA_VS_D0I3C_I3, value);
 
 	/* Wait for cmd in progress to be cleared before exiting the function */
···
 	dev_vdbg(bus->dev, "D0I3C updated, register = 0x%x\n",
 		 snd_hdac_chip_readb(bus, VS_D0I3C));
 
-	if (d0_substate == SOF_DSP_D0I0)
-		flags = HDA_PM_PPG;/* prevent power gating in D0 */
-	else
-		flags = HDA_PM_NO_DMA_TRACE;/* disable DMA trace in D0I3*/
+	return 0;
+}
 
-	/* sending pm_gate IPC */
-	ret = hda_dsp_send_pm_gate_ipc(sdev, flags);
+static int hda_dsp_set_D0_state(struct snd_sof_dev *sdev,
+				const struct sof_dsp_power_state *target_state)
+{
+	u32 flags = 0;
+	int ret;
+	u8 value = 0;
+
+	/*
+	 * Sanity check for illegal state transitions
+	 * The only allowed transitions are:
+	 * 1. D3 -> D0I0
+	 * 2. D0I0 -> D0I3
+	 * 3. D0I3 -> D0I0
+	 */
+	switch (sdev->dsp_power_state.state) {
+	case SOF_DSP_PM_D0:
+		/* Follow the sequence below for D0 substate transitions */
+		break;
+	case SOF_DSP_PM_D3:
+		/* Follow regular flow for D3 -> D0 transition */
+		return 0;
+	default:
+		dev_err(sdev->dev, "error: transition from %d to %d not allowed\n",
+			sdev->dsp_power_state.state, target_state->state);
+		return -EINVAL;
+	}
+
+	/* Set flags and register value for D0 target substate */
+	if (target_state->substate == SOF_HDA_DSP_PM_D0I3) {
+		value = SOF_HDA_VS_D0I3C_I3;
+
+		/*
+		 * Trace DMA is disabled by default when the DSP enters D0I3.
+		 * But it can be kept enabled when the DSP enters D0I3 while the
+		 * system is in S0 for debug.
+		 */
+		if (hda_enable_trace_D0I3_S0 &&
+		    sdev->system_suspend_target != SOF_SUSPEND_NONE)
+			flags = HDA_PM_NO_DMA_TRACE;
+	} else {
+		/* prevent power gating in D0I0 */
+		flags = HDA_PM_PPG;
+	}
+
+	/* update D0I3C register */
+	ret = hda_dsp_update_d0i3c_register(sdev, value);
 	if (ret < 0)
+		return ret;
+
+	/*
+	 * Notify the DSP of the state change.
+	 * If this IPC fails, revert the D0I3C register update in order
+	 * to prevent partial state change.
+	 */
+	ret = hda_dsp_send_pm_gate_ipc(sdev, flags);
+	if (ret < 0) {
 		dev_err(sdev->dev,
 			"error: PM_GATE ipc error %d\n", ret);
+		goto revert;
+	}
+
+	return ret;
+
+revert:
+	/* fallback to the previous register value */
+	value = value ? 0 : SOF_HDA_VS_D0I3C_I3;
+
+	/*
+	 * This can fail but return the IPC error to signal that
+	 * the state change failed.
+	 */
+	hda_dsp_update_d0i3c_register(sdev, value);
 
 	return ret;
 }
+
+/* helper to log DSP state */
+static void hda_dsp_state_log(struct snd_sof_dev *sdev)
+{
+	switch (sdev->dsp_power_state.state) {
+	case SOF_DSP_PM_D0:
+		switch (sdev->dsp_power_state.substate) {
+		case SOF_HDA_DSP_PM_D0I0:
+			dev_dbg(sdev->dev, "Current DSP power state: D0I0\n");
+			break;
+		case SOF_HDA_DSP_PM_D0I3:
+			dev_dbg(sdev->dev, "Current DSP power state: D0I3\n");
+			break;
+		default:
+			dev_dbg(sdev->dev, "Unknown DSP D0 substate: %d\n",
+				sdev->dsp_power_state.substate);
+			break;
+		}
+		break;
+	case SOF_DSP_PM_D1:
+		dev_dbg(sdev->dev, "Current DSP power state: D1\n");
+		break;
+	case SOF_DSP_PM_D2:
+		dev_dbg(sdev->dev, "Current DSP power state: D2\n");
+		break;
+	case SOF_DSP_PM_D3_HOT:
+		dev_dbg(sdev->dev, "Current DSP power state: D3_HOT\n");
+		break;
+	case SOF_DSP_PM_D3:
+		dev_dbg(sdev->dev, "Current DSP power state: D3\n");
+		break;
+	case SOF_DSP_PM_D3_COLD:
+		dev_dbg(sdev->dev, "Current DSP power state: D3_COLD\n");
+		break;
+	default:
+		dev_dbg(sdev->dev, "Unknown DSP power state: %d\n",
+			sdev->dsp_power_state.state);
+		break;
+	}
+}
+
+/*
+ * All DSP power state transitions are initiated by the driver.
+ * If the requested state change fails, the error is simply returned.
+ * Further state transitions are attempted only when the set_power_save() op
+ * is called again either because of a new IPC sent to the DSP or
+ * during system suspend/resume.
+ */
+int hda_dsp_set_power_state(struct snd_sof_dev *sdev,
+			    const struct sof_dsp_power_state *target_state)
+{
+	int ret = 0;
+
+	/*
+	 * When the DSP is already in D0I3 and the target state is D0I3,
+	 * it could be the case that the DSP is in D0I3 during S0
+	 * and the system is suspending to S0Ix. Therefore,
+	 * hda_dsp_set_D0_state() must be called to disable trace DMA
+	 * by sending the PM_GATE IPC to the FW.
+	 */
+	if (target_state->substate == SOF_HDA_DSP_PM_D0I3 &&
+	    sdev->system_suspend_target == SOF_SUSPEND_S0IX)
+		goto set_state;
+
+	/*
+	 * For all other cases, return without doing anything if
+	 * the DSP is already in the target state.
+	 */
+	if (target_state->state == sdev->dsp_power_state.state &&
+	    target_state->substate == sdev->dsp_power_state.substate)
+		return 0;
+
+set_state:
+	switch (target_state->state) {
+	case SOF_DSP_PM_D0:
+		ret = hda_dsp_set_D0_state(sdev, target_state);
+		break;
+	case SOF_DSP_PM_D3:
+		/* The only allowed transition is: D0I0 -> D3 */
+		if (sdev->dsp_power_state.state == SOF_DSP_PM_D0 &&
+		    sdev->dsp_power_state.substate == SOF_HDA_DSP_PM_D0I0)
+			break;
+
+		dev_err(sdev->dev,
+			"error: transition from %d to %d not allowed\n",
+			sdev->dsp_power_state.state, target_state->state);
+		return -EINVAL;
+	default:
+		dev_err(sdev->dev, "error: target state unsupported %d\n",
+			target_state->state);
+		return -EINVAL;
+	}
+	if (ret < 0) {
+		dev_err(sdev->dev,
+			"failed to set requested target DSP state %d substate %d\n",
+			target_state->state, target_state->substate);
+		return ret;
+	}
+
+	sdev->dsp_power_state = *target_state;
+	hda_dsp_state_log(sdev);
+	return ret;
+}
+
+/*
+ * Audio DSP states may transform as below:-
+ *
+ *                                         Opportunistic D0I3 in S0
+ *     Runtime    +---------------------+  Delayed D0i3 work timeout
+ *     suspend    |                     +--------------------+
+ *   +------------+       D0I0(active)  |                    |
+ *   |            |                     <---------------+    |
+ *   |   +-------->                     |    New IPC	|    |
+ *   |   |Runtime +--^--+---------^--+--+ (via mailbox)	|    |
+ *   |   |resume     |  |         |  |			|    |
+ *   |   |           |  |         |  |			|    |
+ *   |   |     System|  |         |  |			|    |
+ *   |   |     resume|  | S3/S0IX |  |                  |    |
+ *   |   |	     |  | suspend |  | S0IX             |    |
+ *   |   |           |  |         |  |suspend           |    |
+ *   |   |           |  |         |  |                  |    |
+ *   |   |           |  |         |  |                  |    |
+ * +-v---+-----------+--v-------+ |  |           +------+----v----+
+ * |                            | |  +----------->                |
+ * |       D3 (suspended)       | |              |      D0I3      |
+ * |                            | +--------------+                |
+ * |                            |  System resume |                |
+ * +----------------------------+		 +----------------+
+ *
+ * S0IX suspend: The DSP is in D0I3 if any D0I3-compatible streams
+ *		 ignored the suspend trigger. Otherwise the DSP
+ *		 is in D3.
+ */
 
 static int hda_suspend(struct snd_sof_dev *sdev, bool runtime_suspend)
 {
···
 	struct hdac_bus *bus = sof_to_bus(sdev);
 #endif
 	int ret;
+
+	hda_sdw_int_enable(sdev, false);
 
 	/* disable IPC interrupts */
 	hda_dsp_ipc_int_disable(sdev);
···
 {
 	struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata;
 	struct pci_dev *pci = to_pci_dev(sdev->dev);
+	const struct sof_dsp_power_state target_state = {
+		.state = SOF_DSP_PM_D0,
+		.substate = SOF_HDA_DSP_PM_D0I0,
+	};
+	int ret;
 
-	if (sdev->s0_suspend) {
+	/* resume from D0I3 */
+	if (sdev->dsp_power_state.state == SOF_DSP_PM_D0) {
 		hda_codec_i915_display_power(sdev, true);
+
+		/* Set DSP power state */
+		ret = snd_sof_dsp_set_power_state(sdev, &target_state);
+		if (ret < 0) {
+			dev_err(sdev->dev, "error: setting dsp state %d substate %d\n",
+				target_state.state, target_state.substate);
+			return ret;
+		}
 
 		/* restore L1SEN bit */
 		if (hda->l1_support_changed)
···
 	}
 
 	/* init hda controller. DSP cores will be powered up during fw boot */
-	return hda_resume(sdev, false);
+	ret = hda_resume(sdev, false);
+	if (ret < 0)
+		return ret;
+
+	return snd_sof_dsp_set_power_state(sdev, &target_state);
 }
 
 int hda_dsp_runtime_resume(struct snd_sof_dev *sdev)
 {
+	const struct sof_dsp_power_state target_state = {
+		.state = SOF_DSP_PM_D0,
+	};
+	int ret;
+
 	/* init hda controller. DSP cores will be powered up during fw boot */
-	return hda_resume(sdev, true);
+	ret = hda_resume(sdev, true);
+	if (ret < 0)
+		return ret;
+
+	return snd_sof_dsp_set_power_state(sdev, &target_state);
 }
 
 int hda_dsp_runtime_idle(struct snd_sof_dev *sdev)
···
 
 int hda_dsp_runtime_suspend(struct snd_sof_dev *sdev)
 {
+	const struct sof_dsp_power_state target_state = {
+		.state = SOF_DSP_PM_D3,
+	};
+	int ret;
+
 	/* stop hda controller and power dsp off */
-	return hda_suspend(sdev, true);
+	ret = hda_suspend(sdev, true);
+	if (ret < 0)
+		return ret;
+
+	return snd_sof_dsp_set_power_state(sdev, &target_state);
 }
 
-int hda_dsp_suspend(struct snd_sof_dev *sdev)
+int hda_dsp_suspend(struct snd_sof_dev *sdev, u32 target_state)
 {
 	struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata;
 	struct hdac_bus *bus = sof_to_bus(sdev);
 	struct pci_dev *pci = to_pci_dev(sdev->dev);
+	const struct sof_dsp_power_state target_dsp_state = {
+		.state = target_state,
+		.substate = target_state == SOF_DSP_PM_D0 ?
+				SOF_HDA_DSP_PM_D0I3 : 0,
+	};
 	int ret;
 
-	if (sdev->s0_suspend) {
+	/* cancel any attempt for DSP D0I3 */
+	cancel_delayed_work_sync(&hda->d0i3_work);
+
+	if (target_state == SOF_DSP_PM_D0) {
 		/* we can't keep a wakeref to display driver at suspend */
 		hda_codec_i915_display_power(sdev, false);
+
+		/* Set DSP power state */
+		ret = snd_sof_dsp_set_power_state(sdev, &target_dsp_state);
+		if (ret < 0) {
+			dev_err(sdev->dev, "error: setting dsp state %d substate %d\n",
+				target_dsp_state.state,
+				target_dsp_state.substate);
+			return ret;
+		}
 
 		/* enable L1SEN to make sure the system can enter S0Ix */
 		hda->l1_support_changed =
···
 		return ret;
 	}
 
-	return 0;
+	return snd_sof_dsp_set_power_state(sdev, &target_dsp_state);
 }
 
 int hda_dsp_set_hw_params_upon_resume(struct snd_sof_dev *sdev)
···
 		 */
 		if (stream->link_substream) {
 			rtd = snd_pcm_substream_chip(stream->link_substream);
-			name = rtd->codec_dai->component->name;
+			name = asoc_rtd_to_codec(rtd, 0)->component->name;
 			link = snd_hdac_ext_bus_get_link(bus, name);
 			if (!link)
 				return -EINVAL;
···
 	}
 #endif
 	return 0;
+}
+
+void hda_dsp_d0i3_work(struct work_struct *work)
+{
+	struct sof_intel_hda_dev *hdev = container_of(work,
+						      struct sof_intel_hda_dev,
+						      d0i3_work.work);
+	struct hdac_bus *bus = &hdev->hbus.core;
+	struct snd_sof_dev *sdev = dev_get_drvdata(bus->dev);
+	struct sof_dsp_power_state target_state;
+	int ret;
+
+	target_state.state = SOF_DSP_PM_D0;
+
+	/* DSP can enter D0I3 iff only D0I3-compatible streams are active */
+	if (snd_sof_dsp_only_d0i3_compatible_stream_active(sdev))
+		target_state.substate = SOF_HDA_DSP_PM_D0I3;
+	else
+		target_state.substate = SOF_HDA_DSP_PM_D0I0;
+
+	/* remain in D0I0 */
+	if (target_state.substate == SOF_HDA_DSP_PM_D0I0)
+		return;
+
+	/* This can fail but error cannot be propagated */
+	ret = snd_sof_dsp_set_power_state(sdev, &target_state);
+	if (ret < 0)
+		dev_err_ratelimited(sdev->dev,
+				    "error: failed to set DSP state %d substate %d\n",
+				    target_state.state, target_state.substate);
 }

sound/soc/sof/intel/hda-ipc.c

···
 		ret = reply.error;
 	} else {
 		/* reply correct size ? */
-		if (reply.hdr.size != msg->reply_size) {
+		if (reply.hdr.size != msg->reply_size &&
+			/* getter payload is never known upfront */
+			!(reply.hdr.cmd & SOF_IPC_GLB_PROBE)) {
 			dev_err(sdev->dev, "error: reply expected %zu got %u bytes\n",
 				msg->reply_size, reply.hdr.size);
 			ret = -EINVAL;
···
 out:
 	msg->reply_error = ret;
 
-}
-
-static bool hda_dsp_ipc_is_sof(uint32_t msg)
-{
-	return (msg & (HDA_DSP_IPC_PURGE_FW | 0xf << 9)) != msg ||
-		(msg & HDA_DSP_IPC_PURGE_FW) != HDA_DSP_IPC_PURGE_FW;
 }
 
 /* IPC handler thread */
···
 		 */
 		spin_lock_irq(&sdev->ipc_lock);
 
-		/* handle immediate reply from DSP core - ignore ROM messages */
-		if (hda_dsp_ipc_is_sof(msg)) {
-			hda_dsp_ipc_get_reply(sdev);
-			snd_sof_ipc_reply(sdev, msg);
-		}
-
-		/* wake up sleeper if we are loading code */
-		if (sdev->code_loading)	{
-			sdev->code_loading = 0;
-			wake_up(&sdev->waitq);
-		}
+		/* handle immediate reply from DSP core */
+		hda_dsp_ipc_get_reply(sdev);
+		snd_sof_ipc_reply(sdev, msg);
 
 		/* set the done bit */
 		hda_dsp_ipc_dsp_done(sdev);

sound/soc/sof/intel/hda-loader.c

···
 		goto err;
 	}
 
+	/* set DONE bit to clear the reply IPC message */
+	snd_sof_dsp_update_bits_forced(sdev, HDA_DSP_BAR,
+				       chip->ipc_ack,
+				       chip->ipc_ack_mask,
+				       chip->ipc_ack_mask);
+
 	/* step 5: power down corex */
 	ret = hda_dsp_core_power_down(sdev,
 				  chip->cores_mask & ~(HDA_DSP_CORE_MASK(0)));
···
 	/* code loader is special case that reuses stream ops */
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
-		wait_event_timeout(sdev->waitq, !sdev->code_loading,
-				   HDA_DSP_CL_TRIGGER_TIMEOUT);
-
 		snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, SOF_HDA_INTCTL,
 					1 << hstream->index,
 					1 << hstream->index);
···
 	}
 
 	/*
+	 * When a SoundWire link is in clock stop state, a Slave
+	 * device may trigger in-band wakes for events such as jack
+	 * insertion or acoustic event detection. This event will lead
+	 * to a WAKEEN interrupt, handled by the PCI device and routed
+	 * to PME if the PCI device is in D3. The resume function in
+	 * audio PCI driver will be invoked by ACPI for PME event and
+	 * initialize the device and process WAKEEN interrupt.
+	 *
+	 * The WAKEEN interrupt should be processed ASAP to prevent an
+	 * interrupt flood, otherwise other interrupts, such IPC,
+	 * cannot work normally.  The WAKEEN is handled after the ROM
+	 * is initialized successfully, which ensures power rails are
+	 * enabled before accessing the SoundWire SHIM registers
+	 */
+	if (!sdev->first_boot)
+		hda_sdw_process_wakeen(sdev);
+
+	/*
 	 * at this point DSP ROM has been initialized and
 	 * should be ready for code loading and firmware boot
 	 */
···
 /* post fw run operations */
 int hda_dsp_post_fw_run(struct snd_sof_dev *sdev)
 {
+	int ret;
+
+	if (sdev->first_boot) {
+		ret = hda_sdw_startup(sdev);
+		if (ret < 0) {
+			dev_err(sdev->dev,
+				"error: could not startup SoundWire links\n");
+			return ret;
+		}
+	}
+
+	hda_sdw_int_enable(sdev, true);
+
 	/* re-enable clock gating and power gating */
 	return hda_dsp_ctrl_clock_power_gating(sdev, true);
 }

sound/soc/sof/intel/hda-pcm.c

···
 #define SDnFMT_BITS(x)	((x) << 4)
 #define SDnFMT_CHAN(x)	((x) << 0)
 
-static inline u32 get_mult_div(struct snd_sof_dev *sdev, int rate)
+u32 hda_dsp_get_mult_div(struct snd_sof_dev *sdev, int rate)
 {
 	switch (rate) {
 	case 8000:
···
 	}
 };
 
-static inline u32 get_bits(struct snd_sof_dev *sdev, int sample_bits)
+u32 hda_dsp_get_bits(struct snd_sof_dev *sdev, int sample_bits)
 {
 	switch (sample_bits) {
 	case 8:
···
 	u32 size, rate, bits;
 
 	size = params_buffer_bytes(params);
-	rate = get_mult_div(sdev, params_rate(params));
-	bits = get_bits(sdev, params_width(params));
+	rate = hda_dsp_get_mult_div(sdev, params_rate(params));
+	bits = hda_dsp_get_bits(sdev, params_width(params));
 
 	hstream->substream = substream;
 

sound/soc/sof/intel/hda-stream.c

···
 					SOF_HDA_REG_PP_PPCTL, mask, 0);
 	spin_unlock_irq(&bus->reg_lock);
 
+	stream->substream = NULL;
+
 	return 0;
 }
 
···
 	return ret;
 }
 
+static void
+hda_dsp_set_bytes_transferred(struct hdac_stream *hstream, u64 buffer_size)
+{
+	u64 prev_pos, pos, num_bytes;
+
+	div64_u64_rem(hstream->curr_pos, buffer_size, &prev_pos);
+	pos = snd_hdac_stream_get_pos_posbuf(hstream);
+
+	if (pos < prev_pos)
+		num_bytes = (buffer_size - prev_pos) +  pos;
+	else
+		num_bytes = pos - prev_pos;
+
+	hstream->curr_pos += num_bytes;
+}
+
 static bool hda_dsp_stream_check(struct hdac_bus *bus, u32 status)
 {
 	struct sof_intel_hda_dev *sof_hda = bus_to_sof_hda(bus);
···
 			snd_hdac_stream_writeb(s, SD_STS, sd_status);
 
 			active = true;
-			if (!s->substream ||
+			if ((!s->substream && !s->cstream) ||
 			    !s->running ||
 			    (sd_status & SOF_HDA_CL_DMA_SD_INT_COMPLETE) == 0)
 				continue;
 
 			/* Inform ALSA only in case not do that with IPC */
-			if (sof_hda->no_ipc_position)
+			if (s->substream && sof_hda->no_ipc_position) {
 				snd_sof_pcm_period_elapsed(s->substream);
+			} else if (s->cstream) {
+				hda_dsp_set_bytes_transferred(s,
+					s->cstream->runtime->buffer_size);
+				snd_compr_fragment_elapsed(s->cstream);
+			}
 		}
 	}
 

sound/soc/sof/intel/hda.c

···
 #include <sound/hdaudio_ext.h>
 #include <sound/hda_register.h>
 
+#include <linux/acpi.h>
 #include <linux/module.h>
+#include <linux/soundwire/sdw.h>
+#include <linux/soundwire/sdw_intel.h>
 #include <sound/intel-nhlt.h>
 #include <sound/sof.h>
 #include <sound/sof/xtensa.h>
+#include "../sof-audio.h"
 #include "../ops.h"
 #include "hda.h"
 
···
 #include "shim.h"
 
 #define EXCEPT_MAX_HDR_SIZE	0x400
+
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE)
+
+/*
+ * The default for SoundWire clock stop quirks is to power gate the IP
+ * and do a Bus Reset, this will need to be modified when the DSP
+ * needs to remain in D0i3 so that the Master does not lose context
+ * and enumeration is not required on clock restart
+ */
+static int sdw_clock_stop_quirks = SDW_INTEL_CLK_STOP_BUS_RESET;
+module_param(sdw_clock_stop_quirks, int, 0444);
+MODULE_PARM_DESC(sdw_clock_stop_quirks, "SOF SoundWire clock stop quirks");
+
+static int sdw_params_stream(struct device *dev,
+			     struct sdw_intel_stream_params_data *params_data)
+{
+	struct snd_sof_dev *sdev = dev_get_drvdata(dev);
+	struct snd_soc_dai *d = params_data->dai;
+	struct sof_ipc_dai_config config;
+	struct sof_ipc_reply reply;
+	int link_id = params_data->link_id;
+	int alh_stream_id = params_data->alh_stream_id;
+	int ret;
+	u32 size = sizeof(config);
+
+	memset(&config, 0, size);
+	config.hdr.size = size;
+	config.hdr.cmd = SOF_IPC_GLB_DAI_MSG | SOF_IPC_DAI_CONFIG;
+	config.type = SOF_DAI_INTEL_ALH;
+	config.dai_index = (link_id << 8) | (d->id);
+	config.alh.stream_id = alh_stream_id;
+
+	/* send message to DSP */
+	ret = sof_ipc_tx_message(sdev->ipc,
+				 config.hdr.cmd, &config, size, &reply,
+				 sizeof(reply));
+	if (ret < 0) {
+		dev_err(sdev->dev,
+			"error: failed to set DAI hw_params for link %d dai->id %d ALH %d\n",
+			link_id, d->id, alh_stream_id);
+	}
+
+	return ret;
+}
+
+static int sdw_free_stream(struct device *dev,
+			   struct sdw_intel_stream_free_data *free_data)
+{
+	struct snd_sof_dev *sdev = dev_get_drvdata(dev);
+	struct snd_soc_dai *d = free_data->dai;
+	struct sof_ipc_dai_config config;
+	struct sof_ipc_reply reply;
+	int link_id = free_data->link_id;
+	int ret;
+	u32 size = sizeof(config);
+
+	memset(&config, 0, size);
+	config.hdr.size = size;
+	config.hdr.cmd = SOF_IPC_GLB_DAI_MSG | SOF_IPC_DAI_CONFIG;
+	config.type = SOF_DAI_INTEL_ALH;
+	config.dai_index = (link_id << 8) | d->id;
+	config.alh.stream_id = 0xFFFF; /* invalid value on purpose */
+
+	/* send message to DSP */
+	ret = sof_ipc_tx_message(sdev->ipc,
+				 config.hdr.cmd, &config, size, &reply,
+				 sizeof(reply));
+	if (ret < 0) {
+		dev_err(sdev->dev,
+			"error: failed to free stream for link %d dai->id %d\n",
+			link_id, d->id);
+	}
+
+	return ret;
+}
+
+static const struct sdw_intel_ops sdw_callback = {
+	.params_stream = sdw_params_stream,
+	.free_stream = sdw_free_stream,
+};
+
+void hda_sdw_int_enable(struct snd_sof_dev *sdev, bool enable)
+{
+	sdw_intel_enable_irq(sdev->bar[HDA_DSP_BAR], enable);
+}
+
+static int hda_sdw_acpi_scan(struct snd_sof_dev *sdev)
+{
+	struct sof_intel_hda_dev *hdev;
+	acpi_handle handle;
+	int ret;
+
+	handle = ACPI_HANDLE(sdev->dev);
+
+	/* save ACPI info for the probe step */
+	hdev = sdev->pdata->hw_pdata;
+
+	ret = sdw_intel_acpi_scan(handle, &hdev->info);
+	if (ret < 0) {
+		dev_err(sdev->dev, "%s failed\n", __func__);
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int hda_sdw_probe(struct snd_sof_dev *sdev)
+{
+	struct sof_intel_hda_dev *hdev;
+	struct sdw_intel_res res;
+	void *sdw;
+
+	hdev = sdev->pdata->hw_pdata;
+
+	memset(&res, 0, sizeof(res));
+
+	res.mmio_base = sdev->bar[HDA_DSP_BAR];
+	res.irq = sdev->ipc_irq;
+	res.handle = hdev->info.handle;
+	res.parent = sdev->dev;
+	res.ops = &sdw_callback;
+	res.dev = sdev->dev;
+	res.clock_stop_quirks = sdw_clock_stop_quirks;
+
+	/*
+	 * ops and arg fields are not populated for now,
+	 * they will be needed when the DAI callbacks are
+	 * provided
+	 */
+
+	/* we could filter links here if needed, e.g for quirks */
+	res.count = hdev->info.count;
+	res.link_mask = hdev->info.link_mask;
+
+	sdw = sdw_intel_probe(&res);
+	if (!sdw) {
+		dev_err(sdev->dev, "error: SoundWire probe failed\n");
+		return -EINVAL;
+	}
+
+	/* save context */
+	hdev->sdw = sdw;
+
+	return 0;
+}
+
+int hda_sdw_startup(struct snd_sof_dev *sdev)
+{
+	struct sof_intel_hda_dev *hdev;
+
+	hdev = sdev->pdata->hw_pdata;
+
+	if (!hdev->sdw)
+		return 0;
+
+	return sdw_intel_startup(hdev->sdw);
+}
+
+static int hda_sdw_exit(struct snd_sof_dev *sdev)
+{
+	struct sof_intel_hda_dev *hdev;
+
+	hdev = sdev->pdata->hw_pdata;
+
+	hda_sdw_int_enable(sdev, false);
+
+	if (hdev->sdw)
+		sdw_intel_exit(hdev->sdw);
+	hdev->sdw = NULL;
+
+	return 0;
+}
+
+static bool hda_dsp_check_sdw_irq(struct snd_sof_dev *sdev)
+{
+	struct sof_intel_hda_dev *hdev;
+	bool ret = false;
+	u32 irq_status;
+
+	hdev = sdev->pdata->hw_pdata;
+
+	if (!hdev->sdw)
+		return ret;
+
+	/* store status */
+	irq_status = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_ADSPIS2);
+
+	/* invalid message ? */
+	if (irq_status == 0xffffffff)
+		goto out;
+
+	/* SDW message ? */
+	if (irq_status & HDA_DSP_REG_ADSPIS2_SNDW)
+		ret = true;
+
+out:
+	return ret;
+}
+
+static irqreturn_t hda_dsp_sdw_thread(int irq, void *context)
+{
+	return sdw_intel_thread(irq, context);
+}
+
+static bool hda_sdw_check_wakeen_irq(struct snd_sof_dev *sdev)
+{
+	struct sof_intel_hda_dev *hdev;
+
+	hdev = sdev->pdata->hw_pdata;
+	if (hdev->sdw &&
+	    snd_sof_dsp_read(sdev, HDA_DSP_BAR,
+			     HDA_DSP_REG_SNDW_WAKE_STS))
+		return true;
+
+	return false;
+}
+
+void hda_sdw_process_wakeen(struct snd_sof_dev *sdev)
+{
+	struct sof_intel_hda_dev *hdev;
+
+	hdev = sdev->pdata->hw_pdata;
+	if (!hdev->sdw)
+		return;
+
+	sdw_intel_process_wakeen_event(hdev->sdw);
+}
+
+#endif
 
 /*
  * Debug
···
 module_param_named(dmic_num, hda_dmic_num, int, 0444);
 MODULE_PARM_DESC(dmic_num, "SOF HDA DMIC number");
 
-static bool hda_codec_use_common_hdmi =
-	IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_COMMON_HDMI_CODEC);
+static bool hda_codec_use_common_hdmi = IS_ENABLED(CONFIG_SND_HDA_CODEC_HDMI);
 module_param_named(use_common_hdmi, hda_codec_use_common_hdmi, bool, 0444);
 MODULE_PARM_DESC(use_common_hdmi, "SOF HDA use common HDMI codec driver");
 #endif
···
 
 	/* init i915 and HDMI codecs */
 	ret = hda_codec_i915_init(sdev);
-	if (ret < 0) {
-		dev_err(sdev->dev, "error: init i915 and HDMI codec failed\n");
-		return ret;
-	}
+	if (ret < 0)
+		dev_warn(sdev->dev, "init of i915 and HDMI codec failed\n");
 
 	/* get controller capabilities */
 	ret = hda_dsp_ctrl_get_caps(sdev);
···
 static int hda_init_caps(struct snd_sof_dev *sdev)
 {
 	struct hdac_bus *bus = sof_to_bus(sdev);
+	struct snd_sof_pdata *pdata = sdev->pdata;
 #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA)
 	struct hdac_ext_link *hlink;
 #endif
+	struct sof_intel_hda_dev *hdev = pdata->hw_pdata;
+	u32 link_mask;
 	int ret = 0;
 
 	device_disable_async_suspend(bus->dev);
···
 	if (ret < 0) {
 		dev_err(bus->dev, "error: init chip failed with ret: %d\n",
 			ret);
-#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA)
-		hda_codec_i915_exit(sdev);
-#endif
 		return ret;
 	}
+
+	/* scan SoundWire capabilities exposed by DSDT */
+	ret = hda_sdw_acpi_scan(sdev);
+	if (ret < 0) {
+		dev_dbg(sdev->dev, "skipping SoundWire, ACPI scan error\n");
+		goto skip_soundwire;
+	}
+
+	link_mask = hdev->info.link_mask;
+	if (!link_mask) {
+		dev_dbg(sdev->dev, "skipping SoundWire, no links enabled\n");
+		goto skip_soundwire;
+	}
+
+	/*
+	 * probe/allocate SoundWire resources.
+	 * The hardware configuration takes place in hda_sdw_startup
+	 * after power rails are enabled.
+	 * It's entirely possible to have a mix of I2S/DMIC/SoundWire
+	 * devices, so we allocate the resources in all cases.
+	 */
+	ret = hda_sdw_probe(sdev);
+	if (ret < 0) {
+		dev_err(sdev->dev, "error: SoundWire probe error\n");
+		return ret;
+	}
+
+skip_soundwire:
 
 #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA)
 	if (bus->mlcap)
···
 	hda_codec_probe_bus(sdev, hda_codec_use_common_hdmi);
 
 	if (!HDA_IDISP_CODEC(bus->codec_mask))
-		hda_codec_i915_exit(sdev);
+		hda_codec_i915_display_power(sdev, false);
 
 	/*
 	 * we are done probing so decrement link counts
···
 static irqreturn_t hda_dsp_interrupt_thread(int irq, void *context)
 {
 	struct snd_sof_dev *sdev = context;
+	struct sof_intel_hda_dev *hdev = sdev->pdata->hw_pdata;
 
 	/* deal with streams and controller first */
 	if (hda_dsp_check_stream_irq(sdev))
···
 
 	if (hda_dsp_check_ipc_irq(sdev))
 		sof_ops(sdev)->irq_thread(irq, sdev);
+
+	if (hda_dsp_check_sdw_irq(sdev))
+		hda_dsp_sdw_thread(irq, hdev->sdw);
+
+	if (hda_sdw_check_wakeen_irq(sdev))
+		hda_sdw_process_wakeen(sdev);
 
 	/* enable GIE interrupt */
 	snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR,
···
 	hda_dsp_ctrl_ppcap_enable(sdev, true);
 	hda_dsp_ctrl_ppcap_int_enable(sdev, true);
 
-	/* initialize waitq for code loading */
-	init_waitqueue_head(&sdev->waitq);
-
 	/* set default mailbox offset for FW ready message */
 	sdev->dsp_box.offset = HDA_DSP_MBOX_UPLINK_OFFSET;
+
+	INIT_DELAYED_WORK(&hdev->d0i3_work, hda_dsp_d0i3_work);
 
 	return 0;
 
···
 	struct pci_dev *pci = to_pci_dev(sdev->dev);
 	const struct sof_intel_dsp_desc *chip = hda->desc;
 
+	/* cancel any attempt for DSP D0I3 */
+	cancel_delayed_work_sync(&hda->d0i3_work);
+
 #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA)
 	/* codec removal, invoke bus_device_remove */
 	snd_hdac_ext_bus_device_remove(bus);
 #endif
+
+	hda_sdw_exit(sdev);
 
 	if (!IS_ERR_OR_NULL(hda->dmic_dev))
 		platform_device_unregister(hda->dmic_dev);
···
 		/*
 		 * If no machine driver is found, then:
 		 *
-		 * hda machine driver is used if :
-		 * 1. there is one HDMI codec and one external HDAudio codec
-		 * 2. only HDMI codec
+		 * generic hda machine driver can handle:
+		 *  - one HDMI codec, and/or
+		 *  - one external HDAudio codec
 		 */
-		if (!pdata->machine && codec_num <= 2 &&
-		    HDA_IDISP_CODEC(bus->codec_mask)) {
+		if (!pdata->machine && codec_num <= 2) {
 			hda_mach = snd_soc_acpi_intel_hda_machines;
 
 			/* topology: use the info from hda_machines */
···
 			dev_info(bus->dev, "using HDA machine driver %s now\n",
 				 hda_mach->drv_name);
 
-			if (codec_num == 1)
+			if (codec_num == 1 && HDA_IDISP_CODEC(bus->codec_mask))
 				idisp_str = "-idisp";
 			else
 				idisp_str = "";
···
 }
 #endif
 
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE)
+/* Check if all Slaves defined on the link can be found */
+static bool link_slaves_found(struct snd_sof_dev *sdev,
+			      const struct snd_soc_acpi_link_adr *link,
+			      struct sdw_intel_ctx *sdw)
+{
+	struct hdac_bus *bus = sof_to_bus(sdev);
+	struct sdw_intel_slave_id *ids = sdw->ids;
+	int num_slaves = sdw->num_slaves;
+	unsigned int part_id, link_id, unique_id, mfg_id;
+	int i, j;
+
+	for (i = 0; i < link->num_adr; i++) {
+		u64 adr = link->adr_d[i].adr;
+
+		mfg_id = SDW_MFG_ID(adr);
+		part_id = SDW_PART_ID(adr);
+		link_id = SDW_DISCO_LINK_ID(adr);
+		for (j = 0; j < num_slaves; j++) {
+			if (ids[j].link_id != link_id ||
+			    ids[j].id.part_id != part_id ||
+			    ids[j].id.mfg_id != mfg_id)
+				continue;
+			/*
+			 * we have to check unique id
+			 * if there is more than one
+			 * Slave on the link
+			 */
+			unique_id = SDW_UNIQUE_ID(adr);
+			if (link->num_adr == 1 ||
+			    ids[j].id.unique_id == SDW_IGNORED_UNIQUE_ID ||
+			    ids[j].id.unique_id == unique_id) {
+				dev_dbg(bus->dev,
+					"found %x at link %d\n",
+					part_id, link_id);
+				break;
+			}
+		}
+		if (j == num_slaves) {
+			dev_dbg(bus->dev,
+				"Slave %x not found\n",
+				part_id);
+			return false;
+		}
+	}
+	return true;
+}
+
+static int hda_sdw_machine_select(struct snd_sof_dev *sdev)
+{
+	struct snd_sof_pdata *pdata = sdev->pdata;
+	const struct snd_soc_acpi_link_adr *link;
+	struct hdac_bus *bus = sof_to_bus(sdev);
+	struct snd_soc_acpi_mach *mach;
+	struct sof_intel_hda_dev *hdev;
+	u32 link_mask;
+	int i;
+
+	hdev = pdata->hw_pdata;
+	link_mask = hdev->info.link_mask;
+
+	/*
+	 * Select SoundWire machine driver if needed using the
+	 * alternate tables. This case deals with SoundWire-only
+	 * machines, for mixed cases with I2C/I2S the detection relies
+	 * on the HID list.
+	 */
+	if (link_mask && !pdata->machine) {
+		for (mach = pdata->desc->alt_machines;
+		     mach && mach->link_mask; mach++) {
+			if (mach->link_mask != link_mask)
+				continue;
+
+			/* No need to match adr if there is no links defined */
+			if (!mach->links)
+				break;
+
+			link = mach->links;
+			for (i = 0; i < hdev->info.count && link->num_adr;
+			     i++, link++) {
+				/*
+				 * Try next machine if any expected Slaves
+				 * are not found on this link.
+				 */
+				if (!link_slaves_found(sdev, link, hdev->sdw))
+					break;
+			}
+			/* Found if all Slaves are checked */
+			if (i == hdev->info.count || !link->num_adr)
+				break;
+		}
+		if (mach && mach->link_mask) {
+			dev_dbg(bus->dev,
+				"SoundWire machine driver %s topology %s\n",
+				mach->drv_name,
+				mach->sof_tplg_filename);
+			pdata->machine = mach;
+			mach->mach_params.links = mach->links;
+			mach->mach_params.link_mask = mach->link_mask;
+			mach->mach_params.platform = dev_name(sdev->dev);
+			pdata->fw_filename = mach->sof_fw_filename;
+			pdata->tplg_filename = mach->sof_tplg_filename;
+		} else {
+			dev_info(sdev->dev,
+				 "No SoundWire machine driver found\n");
+		}
+	}
+
+	return 0;
+}
+#else
+static int hda_sdw_machine_select(struct snd_sof_dev *sdev)
+{
+	return 0;
+}
+#endif
+
 void hda_set_mach_params(const struct snd_soc_acpi_mach *mach,
 			 struct device *dev)
 {
···
 	if (mach) {
 		sof_pdata->tplg_filename = mach->sof_tplg_filename;
 		sof_pdata->machine = mach;
+
+		if (mach->link_mask) {
+			mach->mach_params.links = mach->links;
+			mach->mach_params.link_mask = mach->link_mask;
+		}
 	}
+
+	/*
+	 * If I2S fails, try SoundWire
+	 */
+	hda_sdw_machine_select(sdev);
 
 	/*
 	 * Choose HDA generic machine driver if mach is NULL.

sound/soc/sof/intel/hda.h

···
 #ifndef __SOF_INTEL_HDA_H
 #define __SOF_INTEL_HDA_H
 
+#include <linux/soundwire/sdw.h>
+#include <linux/soundwire/sdw_intel.h>
+#include <sound/compress_driver.h>
 #include <sound/hda_codec.h>
 #include <sound/hdaudio_ext.h>
 #include "shim.h"
···
  * value cannot be read back within the specified time.
  */
 #define HDA_DSP_STREAM_RUN_TIMEOUT		300
-#define HDA_DSP_CL_TRIGGER_TIMEOUT		300
 
 #define HDA_DSP_SPIB_ENABLE			1
 #define HDA_DSP_SPIB_DISABLE			0
···
 #define HDA_DSP_REG_ADSPIS		(HDA_DSP_GEN_BASE + 0x0C)
 #define HDA_DSP_REG_ADSPIC2		(HDA_DSP_GEN_BASE + 0x10)
 #define HDA_DSP_REG_ADSPIS2		(HDA_DSP_GEN_BASE + 0x14)
+
+#define HDA_DSP_REG_ADSPIS2_SNDW	BIT(5)
+#define HDA_DSP_REG_SNDW_WAKE_STS      0x2C192
 
 /* Intel HD Audio Inter-Processor Communication Registers */
 #define HDA_DSP_IPC_BASE		0x40
···
 
 /* Number of DAIs */
 #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA)
+
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_PROBES)
+#define SOF_SKL_NUM_DAIS		16
+#else
 #define SOF_SKL_NUM_DAIS		15
+#endif
+
 #else
 #define SOF_SKL_NUM_DAIS		8
 #endif
···
 #define SOF_HDA_PLAYBACK		0
 #define SOF_HDA_CAPTURE			1
 
+/*
+ * Time in ms for opportunistic D0I3 entry delay.
+ * This has been deliberately chosen to be long to avoid race conditions.
+ * Could be optimized in future.
+ */
+#define SOF_HDA_D0I3_WORK_DELAY_MS	5000
+
+/* HDA DSP D0 substate */
+enum sof_hda_D0_substate {
+	SOF_HDA_DSP_PM_D0I0,	/* default D0 substate */
+	SOF_HDA_DSP_PM_D0I3,	/* low power D0 substate */
+};
+
 /* represents DSP HDA controller frontend - i.e. host facing control */
 struct sof_intel_hda_dev {
 
···
 
 	/* DMIC device */
 	struct platform_device *dmic_dev;
+
+	/* delayed work to enter D0I3 opportunistically */
+	struct delayed_work d0i3_work;
+
+	/* ACPI information stored between scan and probe steps */
+	struct sdw_intel_acpi_info info;
+
+	/* sdw context allocated by SoundWire driver */
+	struct sdw_intel_ctx *sdw;
 };
 
 static inline struct hdac_bus *sof_to_bus(struct snd_sof_dev *s)
···
 void hda_dsp_ipc_int_disable(struct snd_sof_dev *sdev);
 
 int hda_dsp_set_power_state(struct snd_sof_dev *sdev,
-			    enum sof_d0_substate d0_substate);
+			    const struct sof_dsp_power_state *target_state);
 
-int hda_dsp_suspend(struct snd_sof_dev *sdev);
+int hda_dsp_suspend(struct snd_sof_dev *sdev, u32 target_state);
 int hda_dsp_resume(struct snd_sof_dev *sdev);
 int hda_dsp_runtime_suspend(struct snd_sof_dev *sdev);
 int hda_dsp_runtime_resume(struct snd_sof_dev *sdev);
···
 void hda_dsp_dump(struct snd_sof_dev *sdev, u32 flags);
 void hda_ipc_dump(struct snd_sof_dev *sdev);
 void hda_ipc_irq_dump(struct snd_sof_dev *sdev);
+void hda_dsp_d0i3_work(struct work_struct *work);
 
 /*
  * DSP PCM Operations.
  */
+u32 hda_dsp_get_mult_div(struct snd_sof_dev *sdev, int rate);
+u32 hda_dsp_get_bits(struct snd_sof_dev *sdev, int sample_bits);
 int hda_dsp_pcm_open(struct snd_sof_dev *sdev,
 		     struct snd_pcm_substream *substream);
 int hda_dsp_pcm_close(struct snd_sof_dev *sdev,
···
 int hda_ipc_pcm_params(struct snd_sof_dev *sdev,
 		       struct snd_pcm_substream *substream,
 		       const struct sof_ipc_pcm_params_reply *reply);
+
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_PROBES)
+/*
+ * Probe Compress Operations.
+ */
+int hda_probe_compr_assign(struct snd_sof_dev *sdev,
+			   struct snd_compr_stream *cstream,
+			   struct snd_soc_dai *dai);
+int hda_probe_compr_free(struct snd_sof_dev *sdev,
+			 struct snd_compr_stream *cstream,
+			 struct snd_soc_dai *dai);
+int hda_probe_compr_set_params(struct snd_sof_dev *sdev,
+			       struct snd_compr_stream *cstream,
+			       struct snd_compr_params *params,
+			       struct snd_soc_dai *dai);
+int hda_probe_compr_trigger(struct snd_sof_dev *sdev,
+			    struct snd_compr_stream *cstream, int cmd,
+			    struct snd_soc_dai *dai);
+int hda_probe_compr_pointer(struct snd_sof_dev *sdev,
+			    struct snd_compr_stream *cstream,
+			    struct snd_compr_tstamp *tstamp,
+			    struct snd_soc_dai *dai);
+#endif
 
 /*
  * DSP IPC Operations.
···
 int hda_dsp_trace_init(struct snd_sof_dev *sdev, u32 *stream_tag);
 int hda_dsp_trace_release(struct snd_sof_dev *sdev);
 int hda_dsp_trace_trigger(struct snd_sof_dev *sdev, int cmd);
+
+/*
+ * SoundWire support
+ */
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE)
+
+int hda_sdw_startup(struct snd_sof_dev *sdev);
+void hda_sdw_int_enable(struct snd_sof_dev *sdev, bool enable);
+void hda_sdw_process_wakeen(struct snd_sof_dev *sdev);
+
+#else
+
+static inline int hda_sdw_acpi_scan(struct snd_sof_dev *sdev)
+{
+	return 0;
+}
+
+static inline int hda_sdw_probe(struct snd_sof_dev *sdev)
+{
+	return 0;
+}
+
+static inline int hda_sdw_startup(struct snd_sof_dev *sdev)
+{
+	return 0;
+}
+
+static inline int hda_sdw_exit(struct snd_sof_dev *sdev)
+{
+	return 0;
+}
+
+static inline void hda_sdw_int_enable(struct snd_sof_dev *sdev, bool enable)
+{
+}
+
+static inline bool hda_dsp_check_sdw_irq(struct snd_sof_dev *sdev)
+{
+	return false;
+}
+
+static inline irqreturn_t hda_dsp_sdw_thread(int irq, void *context)
+{
+	return IRQ_HANDLED;
+}
+
+static inline bool hda_sdw_check_wakeen_irq(struct snd_sof_dev *sdev)
+{
+	return false;
+}
+
+static inline void hda_sdw_process_wakeen(struct snd_sof_dev *sdev)
+{
+}
+#endif
 
 /* common dai driver */
 extern struct snd_soc_dai_driver skl_dai[];

sound/soc/sof/ipc.c

···
 		snd_sof_handle_fw_exception(ipc->sdev);
 		ret = -ETIMEDOUT;
 	} else {
-		/* copy the data returned from DSP */
 		ret = msg->reply_error;
-		if (msg->reply_size)
-			memcpy(reply_data, msg->reply_data, msg->reply_size);
-		if (ret < 0)
+		if (ret < 0) {
 			dev_err(sdev->dev, "error: ipc error for 0x%x size %zu\n",
 				hdr->cmd, msg->reply_size);
-		else
+		} else {
 			ipc_log_header(sdev->dev, "ipc tx succeeded", hdr->cmd);
+			if (msg->reply_size)
+				/* copy the data returned from DSP */
+				memcpy(reply_data, msg->reply_data,
+				       msg->reply_size);
+		}
 	}
 
 	return ret;
···
 	spin_unlock_irq(&sdev->ipc_lock);
 
 	if (ret < 0) {
-		/* So far IPC TX never fails, consider making the above void */
 		dev_err_ratelimited(sdev->dev,
 				    "error: ipc tx failed with error %d\n",
 				    ret);
···
 		       void *msg_data, size_t msg_bytes, void *reply_data,
 		       size_t reply_bytes)
 {
+	const struct sof_dsp_power_state target_state = {
+		.state = SOF_DSP_PM_D0,
+	};
+	int ret;
+
+	/* ensure the DSP is in D0 before sending a new IPC */
+	ret = snd_sof_dsp_set_power_state(ipc->sdev, &target_state);
+	if (ret < 0) {
+		dev_err(ipc->sdev->dev, "error: resuming DSP %d\n", ret);
+		return ret;
+	}
+
+	return sof_ipc_tx_message_no_pm(ipc, header, msg_data, msg_bytes,
+					reply_data, reply_bytes);
+}
+EXPORT_SYMBOL(sof_ipc_tx_message);
+
+/*
+ * send IPC message from host to DSP without modifying the DSP state.
+ * This will be used for IPC's that can be handled by the DSP
+ * even in a low-power D0 substate.
+ */
+int sof_ipc_tx_message_no_pm(struct snd_sof_ipc *ipc, u32 header,
+			     void *msg_data, size_t msg_bytes,
+			     void *reply_data, size_t reply_bytes)
+{
 	int ret;
 
 	if (msg_bytes > SOF_IPC_MSG_MAX_SIZE ||
···
 
 	return ret;
 }
-EXPORT_SYMBOL(sof_ipc_tx_message);
+EXPORT_SYMBOL(sof_ipc_tx_message_no_pm);
 
 /* handle reply message from DSP */
 int snd_sof_ipc_reply(struct snd_sof_dev *sdev, u32 msg_id)

sound/soc/sof/loader.c

···
 
 		/* process structure data */
 		switch (ext_hdr->type) {
-		case SOF_IPC_EXT_DMA_BUFFER:
-			ret = 0;
-			break;
 		case SOF_IPC_EXT_WINDOW:
 			ret = get_ext_windows(sdev, ext_hdr);
 			break;
···
 	struct snd_sof_pdata *plat_data = sdev->pdata;
 	const char *fw_filename;
 	int ret;
-
-	/* set code loading condition to true */
-	sdev->code_loading = 1;
 
 	/* Don't request firmware again if firmware is already requested */
 	if (plat_data->fw)

sound/soc/sof/ops.h

···
 	return 0;
 }
 
-static inline int snd_sof_dsp_suspend(struct snd_sof_dev *sdev)
+static inline int snd_sof_dsp_suspend(struct snd_sof_dev *sdev,
+				      u32 target_state)
 {
 	if (sof_ops(sdev)->suspend)
-		return sof_ops(sdev)->suspend(sdev);
+		return sof_ops(sdev)->suspend(sdev, target_state);
 
 	return 0;
 }
···
 	return 0;
 }
 
-static inline int snd_sof_dsp_set_power_state(struct snd_sof_dev *sdev,
-					      enum sof_d0_substate substate)
+static inline int
+snd_sof_dsp_set_power_state(struct snd_sof_dev *sdev,
+			    const struct sof_dsp_power_state *target_state)
 {
 	if (sof_ops(sdev)->set_power_state)
-		return sof_ops(sdev)->set_power_state(sdev, substate);
+		return sof_ops(sdev)->set_power_state(sdev, target_state);
 
-	/* D0 substate is not supported */
-	return -ENOTSUPP;
+	/* D0 substate is not supported, do nothing here. */
+	return 0;
 }
 
 /* debug */
···
 
 	return 0;
 }
+
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES)
+static inline int
+snd_sof_probe_compr_assign(struct snd_sof_dev *sdev,
+		struct snd_compr_stream *cstream, struct snd_soc_dai *dai)
+{
+	return sof_ops(sdev)->probe_assign(sdev, cstream, dai);
+}
+
+static inline int
+snd_sof_probe_compr_free(struct snd_sof_dev *sdev,
+		struct snd_compr_stream *cstream, struct snd_soc_dai *dai)
+{
+	return sof_ops(sdev)->probe_free(sdev, cstream, dai);
+}
+
+static inline int
+snd_sof_probe_compr_set_params(struct snd_sof_dev *sdev,
+		struct snd_compr_stream *cstream,
+		struct snd_compr_params *params, struct snd_soc_dai *dai)
+{
+	return sof_ops(sdev)->probe_set_params(sdev, cstream, params, dai);
+}
+
+static inline int
+snd_sof_probe_compr_trigger(struct snd_sof_dev *sdev,
+		struct snd_compr_stream *cstream, int cmd,
+		struct snd_soc_dai *dai)
+{
+	return sof_ops(sdev)->probe_trigger(sdev, cstream, cmd, dai);
+}
+
+static inline int
+snd_sof_probe_compr_pointer(struct snd_sof_dev *sdev,
+		struct snd_compr_stream *cstream,
+		struct snd_compr_tstamp *tstamp, struct snd_soc_dai *dai)
+{
+	if (sof_ops(sdev) && sof_ops(sdev)->probe_pointer)
+		return sof_ops(sdev)->probe_pointer(sdev, cstream, tstamp, dai);
+
+	return 0;
+}
+#endif
 
 /* machine driver */
 static inline int

sound/soc/sof/pcm.c

···
 #include "sof-priv.h"
 #include "sof-audio.h"
 #include "ops.h"
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES)
+#include "compress.h"
+#endif
 
 /* Create DMA buffer page table for DSP */
 static int create_page_table(struct snd_soc_component *component,
···
 /*
  * sof pcm period elapse work
  */
-static void sof_pcm_period_elapsed_work(struct work_struct *work)
+void snd_sof_pcm_period_elapsed_work(struct work_struct *work)
 {
 	struct snd_sof_pcm_stream *sps =
 		container_of(work, struct snd_sof_pcm_stream,
···
 		stream.hdr.cmd |= SOF_IPC_STREAM_TRIG_START;
 		break;
 	case SNDRV_PCM_TRIGGER_SUSPEND:
-		if (sdev->s0_suspend &&
+		if (sdev->system_suspend_target == SOF_SUSPEND_S0IX &&
 		    spcm->stream[substream->stream].d0i3_compatible) {
 			/*
 			 * trap the event, not sending trigger stop to
···
 	dev_dbg(component->dev, "pcm: open stream %d dir %d\n",
 		spcm->pcm.pcm_id, substream->stream);
 
-	INIT_WORK(&spcm->stream[substream->stream].period_elapsed_work,
-		  sof_pcm_period_elapsed_work);
 
 	caps = &spcm->pcm.caps[substream->stream];
 
···
 
 	snd_pcm_set_managed_buffer(pcm->streams[stream].substream,
 				   SNDRV_DMA_TYPE_DEV_SG, sdev->dev,
-				   le32_to_cpu(caps->buffer_size_min),
-				   le32_to_cpu(caps->buffer_size_max));
+				   0, le32_to_cpu(caps->buffer_size_max));
 capture:
 	stream = SNDRV_PCM_STREAM_CAPTURE;
 
···
 
 	snd_pcm_set_managed_buffer(pcm->streams[stream].substream,
 				   SNDRV_DMA_TYPE_DEV_SG, sdev->dev,
-				   le32_to_cpu(caps->buffer_size_min),
-				   le32_to_cpu(caps->buffer_size_max));
+				   0, le32_to_cpu(caps->buffer_size_max));
 
 	return 0;
 }
···
 
 #if IS_ENABLED(CONFIG_SND_SOC_SOF_COMPRESS)
 	pd->compr_ops = &sof_compressed_ops;
+#endif
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES)
+	/* override cops when probe support is enabled */
+	pd->compr_ops = &sof_probe_compressed_ops;
 #endif
 	pd->pcm_construct = sof_pcm_new;
 	pd->ignore_machine = drv_name;

sound/soc/sof/pm.c

···
 #include "sof-priv.h"
 #include "sof-audio.h"
 
+/*
+ * Helper function to determine the target DSP state during
+ * system suspend. This function only cares about the device
+ * D-states. Platform-specific substates, if any, should be
+ * handled by the platform-specific parts.
+ */
+static u32 snd_sof_dsp_power_target(struct snd_sof_dev *sdev)
+{
+	u32 target_dsp_state;
+
+	switch (sdev->system_suspend_target) {
+	case SOF_SUSPEND_S3:
+		/* DSP should be in D3 if the system is suspending to S3 */
+		target_dsp_state = SOF_DSP_PM_D3;
+		break;
+	case SOF_SUSPEND_S0IX:
+		/*
+		 * Currently, the only criterion for retaining the DSP in D0
+		 * is that there are streams that ignored the suspend trigger.
+		 * Additional criteria such Soundwire clock-stop mode and
+		 * device suspend latency considerations will be added later.
+		 */
+		if (snd_sof_stream_suspend_ignored(sdev))
+			target_dsp_state = SOF_DSP_PM_D0;
+		else
+			target_dsp_state = SOF_DSP_PM_D3;
+		break;
+	default:
+		/* This case would be during runtime suspend */
+		target_dsp_state = SOF_DSP_PM_D3;
+		break;
+	}
+
+	return target_dsp_state;
+}
+
 static int sof_send_pm_ctx_ipc(struct snd_sof_dev *sdev, int cmd)
 {
 	struct sof_ipc_pm_ctx pm_ctx;
···
 static int sof_resume(struct device *dev, bool runtime_resume)
 {
 	struct snd_sof_dev *sdev = dev_get_drvdata(dev);
+	u32 old_state = sdev->dsp_power_state.state;
 	int ret;
 
 	/* do nothing if dsp resume callbacks are not set */
···
 			"error: failed to power up DSP after resume\n");
 		return ret;
 	}
+
+	/* Nothing further to do if resuming from a low-power D0 substate */
+	if (!runtime_resume && old_state == SOF_DSP_PM_D0)
+		return 0;
 
 	sdev->fw_state = SOF_FW_BOOT_PREPARE;
 
···
 			"error: ctx_restore ipc error during resume %d\n",
 			ret);
 
-	/* initialize default D0 sub-state */
-	sdev->d0_substate = SOF_DSP_D0I0;
-
 	return ret;
 }
 
 static int sof_suspend(struct device *dev, bool runtime_suspend)
 {
 	struct snd_sof_dev *sdev = dev_get_drvdata(dev);
+	u32 target_state = 0;
 	int ret;
 
 	/* do nothing if dsp suspend callback is not set */
···
 		return 0;
 
 	if (sdev->fw_state != SOF_FW_BOOT_COMPLETE)
-		goto power_down;
-
-	/* release trace */
-	snd_sof_release_trace(sdev);
+		goto suspend;
 
 	/* set restore_stream for all streams during system suspend */
 	if (!runtime_suspend) {
···
 			return ret;
 		}
 	}
+
+	target_state = snd_sof_dsp_power_target(sdev);
+
+	/* Skip to platform-specific suspend if DSP is entering D0 */
+	if (target_state == SOF_DSP_PM_D0)
+		goto suspend;
+
+	/* release trace */
+	snd_sof_release_trace(sdev);
 
 #if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_ENABLE_DEBUGFS_CACHE)
 	/* cache debugfs contents during runtime suspend */
···
 			 ret);
 	}
 
-power_down:
+suspend:
 
 	/* return if the DSP was not probed successfully */
 	if (sdev->fw_state == SOF_FW_BOOT_NOT_STARTED)
 		return 0;
 
-	/* power down all DSP cores */
+	/* platform-specific suspend */
 	if (runtime_suspend)
 		ret = snd_sof_dsp_runtime_suspend(sdev);
 	else
-		ret = snd_sof_dsp_suspend(sdev);
+		ret = snd_sof_dsp_suspend(sdev, target_state);
 	if (ret < 0)
 		dev_err(sdev->dev,
 			"error: failed to power down DSP during suspend %d\n",
 			ret);
+
+	/* Do not reset FW state if DSP is in D0 */
+	if (target_state == SOF_DSP_PM_D0)
+		return ret;
 
 	/* reset FW state */
 	sdev->fw_state = SOF_FW_BOOT_NOT_STARTED;
···
 }
 EXPORT_SYMBOL(snd_sof_runtime_resume);
 
-int snd_sof_set_d0_substate(struct snd_sof_dev *sdev,
-			    enum sof_d0_substate d0_substate)
-{
-	int ret;
-
-	if (sdev->d0_substate == d0_substate)
-		return 0;
-
-	/* do platform specific set_state */
-	ret = snd_sof_dsp_set_power_state(sdev, d0_substate);
-	if (ret < 0)
-		return ret;
-
-	/* update dsp D0 sub-state */
-	sdev->d0_substate = d0_substate;
-
-	return 0;
-}
-EXPORT_SYMBOL(snd_sof_set_d0_substate);
-
-/*
- * Audio DSP states may transform as below:-
- *
- *                                         D0I3 compatible stream
- *     Runtime    +---------------------+   opened only, timeout
- *     suspend    |                     +--------------------+
- *   +------------+       D0(active)    |                    |
- *   |            |                     <---------------+    |
- *   |   +-------->                     |               |    |
- *   |   |Runtime +--^--+---------^--+--+ The last      |    |
- *   |   |resume     |  |         |  |    opened D0I3   |    |
- *   |   |           |  |         |  |    compatible    |    |
- *   |   |     resume|  |         |  |    stream closed |    |
- *   |   |      from |  | D3      |  |                  |    |
- *   |   |       D3  |  |suspend  |  | d0i3             |    |
- *   |   |           |  |         |  |suspend           |    |
- *   |   |           |  |         |  |                  |    |
- *   |   |           |  |         |  |                  |    |
- * +-v---+-----------+--v-------+ |  |           +------+----v----+
- * |                            | |  +----------->                |
- * |       D3 (suspended)       | |              |      D0I3      +-----+
- * |                            | +--------------+                |     |
- * |                            |  resume from   |                |     |
- * +-------------------^--------+  d0i3 suspend  +----------------+     |
- *                     |                                                |
- *                     |                       D3 suspend               |
- *                     +------------------------------------------------+
- *
- * d0i3_suspend = s0_suspend && D0I3 stream opened,
- * D3 suspend = !d0i3_suspend,
- */
-
 int snd_sof_resume(struct device *dev)
 {
-	struct snd_sof_dev *sdev = dev_get_drvdata(dev);
-	int ret;
-
-	if (snd_sof_dsp_d0i3_on_suspend(sdev)) {
-		/* resume from D0I3 */
-		dev_dbg(sdev->dev, "DSP will exit from D0i3...\n");
-		ret = snd_sof_set_d0_substate(sdev, SOF_DSP_D0I0);
-		if (ret == -ENOTSUPP) {
-			/* fallback to resume from D3 */
-			dev_dbg(sdev->dev, "D0i3 not supported, fall back to resume from D3...\n");
-			goto d3_resume;
-		} else if (ret < 0) {
-			dev_err(sdev->dev, "error: failed to exit from D0I3 %d\n",
-				ret);
-			return ret;
-		}
-
-		/* platform-specific resume from D0i3 */
-		return snd_sof_dsp_resume(sdev);
-	}
-
-d3_resume:
-	/* resume from D3 */
 	return sof_resume(dev, false);
 }
 EXPORT_SYMBOL(snd_sof_resume);
 
 int snd_sof_suspend(struct device *dev)
 {
-	struct snd_sof_dev *sdev = dev_get_drvdata(dev);
-	int ret;
-
-	if (snd_sof_dsp_d0i3_on_suspend(sdev)) {
-		/* suspend to D0i3 */
-		dev_dbg(sdev->dev, "DSP is trying to enter D0i3...\n");
-		ret = snd_sof_set_d0_substate(sdev, SOF_DSP_D0I3);
-		if (ret == -ENOTSUPP) {
-			/* fallback to D3 suspend */
-			dev_dbg(sdev->dev, "D0i3 not supported, fall back to D3...\n");
-			goto d3_suspend;
-		} else if (ret < 0) {
-			dev_err(sdev->dev, "error: failed to enter D0I3, %d\n",
-				ret);
-			return ret;
-		}
-
-		/* platform-specific suspend to D0i3 */
-		return snd_sof_dsp_suspend(sdev);
-	}
-
-d3_suspend:
-	/* suspend to D3 */
 	return sof_suspend(dev, false);
 }
 EXPORT_SYMBOL(snd_sof_suspend);
···
 	struct snd_sof_dev *sdev = dev_get_drvdata(dev);
 
 #if defined(CONFIG_ACPI)
-	sdev->s0_suspend = acpi_target_system_state() == ACPI_STATE_S0;
+	if (acpi_target_system_state() == ACPI_STATE_S0)
+		sdev->system_suspend_target = SOF_SUSPEND_S0IX;
+	else
+		sdev->system_suspend_target = SOF_SUSPEND_S3;
 #else
 	/* will suspend to S3 by default */
-	sdev->s0_suspend = false;
+	sdev->system_suspend_target = SOF_SUSPEND_S3;
 #endif
 
 	return 0;
···
 {
 	struct snd_sof_dev *sdev = dev_get_drvdata(dev);
 
-	sdev->s0_suspend = false;
+	sdev->system_suspend_target = SOF_SUSPEND_NONE;
 }
 EXPORT_SYMBOL(snd_sof_complete);

sound/soc/sof/probe.c

···
+// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause)
+//
+// This file is provided under a dual BSD/GPLv2 license.  When using or
+// redistributing this file, you may do so under either license.
+//
+// Copyright(c) 2019-2020 Intel Corporation. All rights reserved.
+//
+// Author: Cezary Rojewski <[e-mail]>
+//
+
+#include "sof-priv.h"
+#include "probe.h"
+
+/**
+ * sof_ipc_probe_init - initialize data probing
+ * @sdev:		SOF sound device
+ * @stream_tag:		Extractor stream tag
+ * @buffer_size:	DMA buffer size to set for extractor
+ *
+ * Host chooses whether extraction is supported or not by providing
+ * valid stream tag to DSP. Once specified, stream described by that
+ * tag will be tied to DSP for extraction for the entire lifetime of
+ * probe.
+ *
+ * Probing is initialized only once and each INIT request must be
+ * matched by DEINIT call.
+ */
+int sof_ipc_probe_init(struct snd_sof_dev *sdev,
+		u32 stream_tag, size_t buffer_size)
+{
+	struct sof_ipc_probe_dma_add_params *msg;
+	struct sof_ipc_reply reply;
+	size_t size = struct_size(msg, dma, 1);
+	int ret;
+
+	msg = kmalloc(size, GFP_KERNEL);
+	if (!msg)
+		return -ENOMEM;
+	msg->hdr.size = size;
+	msg->hdr.cmd = SOF_IPC_GLB_PROBE | SOF_IPC_PROBE_INIT;
+	msg->num_elems = 1;
+	msg->dma[0].stream_tag = stream_tag;
+	msg->dma[0].dma_buffer_size = buffer_size;
+
+	ret = sof_ipc_tx_message(sdev->ipc, msg->hdr.cmd, msg, msg->hdr.size,
+			&reply, sizeof(reply));
+	kfree(msg);
+	return ret;
+}
+EXPORT_SYMBOL(sof_ipc_probe_init);
+
+/**
+ * sof_ipc_probe_deinit - cleanup after data probing
+ * @sdev:	SOF sound device
+ *
+ * Host sends DEINIT request to free previously initialized probe
+ * on DSP side once it is no longer needed. DEINIT only when there
+ * are no probes connected and with all injectors detached.
+ */
+int sof_ipc_probe_deinit(struct snd_sof_dev *sdev)
+{
+	struct sof_ipc_cmd_hdr msg;
+	struct sof_ipc_reply reply;
+
+	msg.size = sizeof(msg);
+	msg.cmd = SOF_IPC_GLB_PROBE | SOF_IPC_PROBE_DEINIT;
+
+	return sof_ipc_tx_message(sdev->ipc, msg.cmd, &msg, msg.size,
+			&reply, sizeof(reply));
+}
+EXPORT_SYMBOL(sof_ipc_probe_deinit);
+
+static int sof_ipc_probe_info(struct snd_sof_dev *sdev, unsigned int cmd,
+		void **params, size_t *num_params)
+{
+	struct sof_ipc_probe_info_params msg = {{{0}}};
+	struct sof_ipc_probe_info_params *reply;
+	size_t bytes;
+	int ret;
+
+	*params = NULL;
+	*num_params = 0;
+
+	reply = kzalloc(SOF_IPC_MSG_MAX_SIZE, GFP_KERNEL);
+	if (!reply)
+		return -ENOMEM;
+	msg.rhdr.hdr.size = sizeof(msg);
+	msg.rhdr.hdr.cmd = SOF_IPC_GLB_PROBE | cmd;
+
+	ret = sof_ipc_tx_message(sdev->ipc, msg.rhdr.hdr.cmd, &msg,
+			msg.rhdr.hdr.size, reply, SOF_IPC_MSG_MAX_SIZE);
+	if (ret < 0 || reply->rhdr.error < 0)
+		goto exit;
+
+	if (!reply->num_elems)
+		goto exit;
+
+	if (cmd == SOF_IPC_PROBE_DMA_INFO)
+		bytes = sizeof(reply->dma[0]);
+	else
+		bytes = sizeof(reply->desc[0]);
+	bytes *= reply->num_elems;
+	*params = kmemdup(&reply->dma[0], bytes, GFP_KERNEL);
+	if (!*params) {
+		ret = -ENOMEM;
+		goto exit;
+	}
+	*num_params = reply->num_elems;
+
+exit:
+	kfree(reply);
+	return ret;
+}
+
+/**
+ * sof_ipc_probe_dma_info - retrieve list of active injection dmas
+ * @sdev:	SOF sound device
+ * @dma:	Returned list of active dmas
+ * @num_dma:	Returned count of active dmas
+ *
+ * Host sends DMA_INFO request to obtain list of injection dmas it
+ * can use to transfer data over with.
+ *
+ * Note that list contains only injection dmas as there is only one
+ * extractor (dma) and it is always assigned on probing init.
+ * DSP knows exactly where data from extraction probes is going to,
+ * which is not the case for injection where multiple streams
+ * could be engaged.
+ */
+int sof_ipc_probe_dma_info(struct snd_sof_dev *sdev,
+		struct sof_probe_dma **dma, size_t *num_dma)
+{
+	return sof_ipc_probe_info(sdev, SOF_IPC_PROBE_DMA_INFO,
+			(void **)dma, num_dma);
+}
+EXPORT_SYMBOL(sof_ipc_probe_dma_info);
+
+/**
+ * sof_ipc_probe_dma_add - attach to specified dmas
+ * @sdev:	SOF sound device
+ * @dma:	List of streams (dmas) to attach to
+ * @num_dma:	Number of elements in @dma
+ *
+ * Contrary to extraction, injection streams are never assigned
+ * on init. Before attempting any data injection, host is responsible
+ * for specifying streams which will be later used to transfer data
+ * to connected probe points.
+ */
+int sof_ipc_probe_dma_add(struct snd_sof_dev *sdev,
+		struct sof_probe_dma *dma, size_t num_dma)
+{
+	struct sof_ipc_probe_dma_add_params *msg;
+	struct sof_ipc_reply reply;
+	size_t size = struct_size(msg, dma, num_dma);
+	int ret;
+
+	msg = kmalloc(size, GFP_KERNEL);
+	if (!msg)
+		return -ENOMEM;
+	msg->hdr.size = size;
+	msg->num_elems = num_dma;
+	msg->hdr.cmd = SOF_IPC_GLB_PROBE | SOF_IPC_PROBE_DMA_ADD;
+	memcpy(&msg->dma[0], dma, size - sizeof(*msg));
+
+	ret = sof_ipc_tx_message(sdev->ipc, msg->hdr.cmd, msg, msg->hdr.size,
+			&reply, sizeof(reply));
+	kfree(msg);
+	return ret;
+}
+EXPORT_SYMBOL(sof_ipc_probe_dma_add);
+
+/**
+ * sof_ipc_probe_dma_remove - detach from specified dmas
+ * @sdev:		SOF sound device
+ * @stream_tag:		List of stream tags to detach from
+ * @num_stream_tag:	Number of elements in @stream_tag
+ *
+ * Host sends DMA_REMOVE request to free previously attached stream
+ * from being occupied for injection. Each detach operation should
+ * match equivalent DMA_ADD. Detach only when all probes tied to
+ * given stream have been disconnected.
+ */
+int sof_ipc_probe_dma_remove(struct snd_sof_dev *sdev,
+		unsigned int *stream_tag, size_t num_stream_tag)
+{
+	struct sof_ipc_probe_dma_remove_params *msg;
+	struct sof_ipc_reply reply;
+	size_t size = struct_size(msg, stream_tag, num_stream_tag);
+	int ret;
+
+	msg = kmalloc(size, GFP_KERNEL);
+	if (!msg)
+		return -ENOMEM;
+	msg->hdr.size = size;
+	msg->num_elems = num_stream_tag;
+	msg->hdr.cmd = SOF_IPC_GLB_PROBE | SOF_IPC_PROBE_DMA_REMOVE;
+	memcpy(&msg->stream_tag[0], stream_tag, size - sizeof(*msg));
+
+	ret = sof_ipc_tx_message(sdev->ipc, msg->hdr.cmd, msg, msg->hdr.size,
+			&reply, sizeof(reply));
+	kfree(msg);
+	return ret;
+}
+EXPORT_SYMBOL(sof_ipc_probe_dma_remove);
+
+/**
+ * sof_ipc_probe_points_info - retrieve list of active probe points
+ * @sdev:	SOF sound device
+ * @desc:	Returned list of active probes
+ * @num_desc:	Returned count of active probes
+ *
+ * Host sends PROBE_POINT_INFO request to obtain list of active probe
+ * points, valid for disconnection when given probe is no longer
+ * required.
+ */
+int sof_ipc_probe_points_info(struct snd_sof_dev *sdev,
+		struct sof_probe_point_desc **desc, size_t *num_desc)
+{
+	return sof_ipc_probe_info(sdev, SOF_IPC_PROBE_POINT_INFO,
+				 (void **)desc, num_desc);
+}
+EXPORT_SYMBOL(sof_ipc_probe_points_info);
+
+/**
+ * sof_ipc_probe_points_add - connect specified probes
+ * @sdev:	SOF sound device
+ * @desc:	List of probe points to connect
+ * @num_desc:	Number of elements in @desc
+ *
+ * Dynamically connects to provided set of endpoints. Immediately
+ * after connection is established, host must be prepared to
+ * transfer data from or to target stream given the probing purpose.
+ *
+ * Each probe point should be removed using PROBE_POINT_REMOVE
+ * request when no longer needed.
+ */
+int sof_ipc_probe_points_add(struct snd_sof_dev *sdev,
+		struct sof_probe_point_desc *desc, size_t num_desc)
+{
+	struct sof_ipc_probe_point_add_params *msg;
+	struct sof_ipc_reply reply;
+	size_t size = struct_size(msg, desc, num_desc);
+	int ret;
+
+	msg = kmalloc(size, GFP_KERNEL);
+	if (!msg)
+		return -ENOMEM;
+	msg->hdr.size = size;
+	msg->num_elems = num_desc;
+	msg->hdr.cmd = SOF_IPC_GLB_PROBE | SOF_IPC_PROBE_POINT_ADD;
+	memcpy(&msg->desc[0], desc, size - sizeof(*msg));
+
+	ret = sof_ipc_tx_message(sdev->ipc, msg->hdr.cmd, msg, msg->hdr.size,
+			&reply, sizeof(reply));
+	kfree(msg);
+	return ret;
+}
+EXPORT_SYMBOL(sof_ipc_probe_points_add);
+
+/**
+ * sof_ipc_probe_points_remove - disconnect specified probes
+ * @sdev:		SOF sound device
+ * @buffer_id:		List of probe points to disconnect
+ * @num_buffer_id:	Number of elements in @desc
+ *
+ * Removes previously connected probes from list of active probe
+ * points and frees all resources on DSP side.
+ */
+int sof_ipc_probe_points_remove(struct snd_sof_dev *sdev,
+		unsigned int *buffer_id, size_t num_buffer_id)
+{
+	struct sof_ipc_probe_point_remove_params *msg;
+	struct sof_ipc_reply reply;
+	size_t size = struct_size(msg, buffer_id, num_buffer_id);
+	int ret;
+
+	msg = kmalloc(size, GFP_KERNEL);
+	if (!msg)
+		return -ENOMEM;
+	msg->hdr.size = size;
+	msg->num_elems = num_buffer_id;
+	msg->hdr.cmd = SOF_IPC_GLB_PROBE | SOF_IPC_PROBE_POINT_REMOVE;
+	memcpy(&msg->buffer_id[0], buffer_id, size - sizeof(*msg));
+
+	ret = sof_ipc_tx_message(sdev->ipc, msg->hdr.cmd, msg, msg->hdr.size,
+			&reply, sizeof(reply));
+	kfree(msg);
+	return ret;
+}
+EXPORT_SYMBOL(sof_ipc_probe_points_remove);

sound/soc/sof/probe.h

···
+/* SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) */
+/*
+ * This file is provided under a dual BSD/GPLv2 license.  When using or
+ * redistributing this file, you may do so under either license.
+ *
+ * Copyright(c) 2019-2020 Intel Corporation. All rights reserved.
+ *
+ * Author: Cezary Rojewski <[e-mail]>
+ */
+
+#ifndef __SOF_PROBE_H
+#define __SOF_PROBE_H
+
+#include <sound/sof/header.h>
+
+struct snd_sof_dev;
+
+#define SOF_PROBE_INVALID_NODE_ID UINT_MAX
+
+struct sof_probe_dma {
+	unsigned int stream_tag;
+	unsigned int dma_buffer_size;
+} __packed;
+
+enum sof_connection_purpose {
+	SOF_CONNECTION_PURPOSE_EXTRACT = 1,
+	SOF_CONNECTION_PURPOSE_INJECT,
+};
+
+struct sof_probe_point_desc {
+	unsigned int buffer_id;
+	unsigned int purpose;
+	unsigned int stream_tag;
+} __packed;
+
+struct sof_ipc_probe_dma_add_params {
+	struct sof_ipc_cmd_hdr hdr;
+	unsigned int num_elems;
+	struct sof_probe_dma dma[0];
+} __packed;
+
+struct sof_ipc_probe_info_params {
+	struct sof_ipc_reply rhdr;
+	unsigned int num_elems;
+	union {
+		struct sof_probe_dma dma[0];
+		struct sof_probe_point_desc desc[0];
+	};
+} __packed;
+
+struct sof_ipc_probe_dma_remove_params {
+	struct sof_ipc_cmd_hdr hdr;
+	unsigned int num_elems;
+	unsigned int stream_tag[0];
+} __packed;
+
+struct sof_ipc_probe_point_add_params {
+	struct sof_ipc_cmd_hdr hdr;
+	unsigned int num_elems;
+	struct sof_probe_point_desc desc[0];
+} __packed;
+
+struct sof_ipc_probe_point_remove_params {
+	struct sof_ipc_cmd_hdr hdr;
+	unsigned int num_elems;
+	unsigned int buffer_id[0];
+} __packed;
+
+int sof_ipc_probe_init(struct snd_sof_dev *sdev,
+		u32 stream_tag, size_t buffer_size);
+int sof_ipc_probe_deinit(struct snd_sof_dev *sdev);
+int sof_ipc_probe_dma_info(struct snd_sof_dev *sdev,
+		struct sof_probe_dma **dma, size_t *num_dma);
+int sof_ipc_probe_dma_add(struct snd_sof_dev *sdev,
+		struct sof_probe_dma *dma, size_t num_dma);
+int sof_ipc_probe_dma_remove(struct snd_sof_dev *sdev,
+		unsigned int *stream_tag, size_t num_stream_tag);
+int sof_ipc_probe_points_info(struct snd_sof_dev *sdev,
+		struct sof_probe_point_desc **desc, size_t *num_desc);
+int sof_ipc_probe_points_add(struct snd_sof_dev *sdev,
+		struct sof_probe_point_desc *desc, size_t num_desc);
+int sof_ipc_probe_points_remove(struct snd_sof_dev *sdev,
+		unsigned int *buffer_id, size_t num_buffer_id);
+
+#endif

sound/soc/sof/sof-audio.c

···
 #include "sof-audio.h"
 #include "ops.h"
 
-bool snd_sof_dsp_d0i3_on_suspend(struct snd_sof_dev *sdev)
+/*
+ * helper to determine if there are only D0i3 compatible
+ * streams active
+ */
+bool snd_sof_dsp_only_d0i3_compatible_stream_active(struct snd_sof_dev *sdev)
+{
+	struct snd_pcm_substream *substream;
+	struct snd_sof_pcm *spcm;
+	bool d0i3_compatible_active = false;
+	int dir;
+
+	list_for_each_entry(spcm, &sdev->pcm_list, list) {
+		for_each_pcm_streams(dir) {
+			substream = spcm->stream[dir].substream;
+			if (!substream || !substream->runtime)
+				continue;
+
+			/*
+			 * substream->runtime being not NULL indicates that
+			 * that the stream is open. No need to check the
+			 * stream state.
+			 */
+			if (!spcm->stream[dir].d0i3_compatible)
+				return false;
+
+			d0i3_compatible_active = true;
+		}
+	}
+
+	return d0i3_compatible_active;
+}
+EXPORT_SYMBOL(snd_sof_dsp_only_d0i3_compatible_stream_active);
+
+bool snd_sof_stream_suspend_ignored(struct snd_sof_dev *sdev)
 {
 	struct snd_sof_pcm *spcm;
 
···
 	 * have been suspended.
 	 */
 	list_for_each_entry(spcm, &sdev->pcm_list, list) {
-		for (dir = 0; dir <= SNDRV_PCM_STREAM_CAPTURE; dir++) {
+		for_each_pcm_streams(dir) {
+			/*
+			 * do not reset hw_params upon resume for streams that
+			 * were kept running during suspend
+			 */
+			if (spcm->stream[dir].suspend_ignored)
+				continue;
+
 			substream = spcm->stream[dir].substream;
 			if (!substream || !substream->runtime)
 				continue;
···
 	int dir;
 
 	list_for_each_entry(spcm, &sdev->pcm_list, list) {
-		dir = SNDRV_PCM_STREAM_PLAYBACK;
-		if (spcm->stream[dir].comp_id == comp_id) {
-			*direction = dir;
-			return spcm;
-		}
-
-		dir = SNDRV_PCM_STREAM_CAPTURE;
-		if (spcm->stream[dir].comp_id == comp_id) {
-			*direction = dir;
-			return spcm;
+		for_each_pcm_streams(dir) {
+			if (spcm->stream[dir].comp_id == comp_id) {
+				*direction = dir;
+				return spcm;
+			}
 		}
 	}
 

sound/soc/sof/sof-audio.h

···
 #ifndef __SOUND_SOC_SOF_AUDIO_H
 #define __SOUND_SOC_SOF_AUDIO_H
 
+#include <linux/workqueue.h>
+
 #include <sound/soc.h>
 #include <sound/control.h>
 #include <sound/sof/stream.h> /* needs to be included before control.h */
···
 struct snd_sof_pcm *snd_sof_find_spcm_pcm_id(struct snd_soc_component *scomp,
 					     unsigned int pcm_id);
 void snd_sof_pcm_period_elapsed(struct snd_pcm_substream *substream);
+void snd_sof_pcm_period_elapsed_work(struct work_struct *work);
 
 /*
  * Mixer IPC
···
 /* PM */
 int sof_restore_pipelines(struct device *dev);
 int sof_set_hw_params_upon_resume(struct device *dev);
-bool snd_sof_dsp_d0i3_on_suspend(struct snd_sof_dev *sdev);
+bool snd_sof_stream_suspend_ignored(struct snd_sof_dev *sdev);
+bool snd_sof_dsp_only_d0i3_compatible_stream_active(struct snd_sof_dev *sdev);
 
 /* Machine driver enumeration */
 int sof_machine_register(struct snd_sof_dev *sdev, void *pdata);

sound/soc/sof/sof-of-dev.c

···
 #include "ops.h"
 
 extern struct snd_sof_dsp_ops sof_imx8_ops;
+extern struct snd_sof_dsp_ops sof_imx8x_ops;
 
 /* platform specific devices */
 #if IS_ENABLED(CONFIG_SND_SOC_SOF_IMX8)
 static struct sof_dev_desc sof_of_imx8qxp_desc = {
+	.default_fw_path = "imx/sof",
+	.default_tplg_path = "imx/sof-tplg",
+	.default_fw_filename = "sof-imx8x.ri",
+	.nocodec_tplg_filename = "sof-imx8-nocodec.tplg",
+	.ops = &sof_imx8x_ops,
+};
+
+static struct sof_dev_desc sof_of_imx8qm_desc = {
 	.default_fw_path = "imx/sof",
 	.default_tplg_path = "imx/sof-tplg",
 	.default_fw_filename = "sof-imx8.ri",
···
 static const struct of_device_id sof_of_ids[] = {
 #if IS_ENABLED(CONFIG_SND_SOC_SOF_IMX8)
 	{ .compatible = "fsl,imx8qxp-dsp", .data = &sof_of_imx8qxp_desc},
+	{ .compatible = "fsl,imx8qm-dsp", .data = &sof_of_imx8qm_desc},
 #endif
 	{ }
 };

sound/soc/sof/sof-priv.h

···
 	(IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_ENABLE_DEBUGFS_CACHE) || \
 	 IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_IPC_FLOOD_TEST))
 
-/* DSP D0ix sub-state */
-enum sof_d0_substate {
-	SOF_DSP_D0I0 = 0,	/* DSP default D0 substate */
-	SOF_DSP_D0I3,		/* DSP D0i3(low power) substate*/
+/* DSP power state */
+enum sof_dsp_power_states {
+	SOF_DSP_PM_D0,
+	SOF_DSP_PM_D1,
+	SOF_DSP_PM_D2,
+	SOF_DSP_PM_D3_HOT,
+	SOF_DSP_PM_D3,
+	SOF_DSP_PM_D3_COLD,
+};
+
+struct sof_dsp_power_state {
+	u32 state;
+	u32 substate; /* platform-specific */
+};
+
+/* System suspend target state */
+enum sof_system_suspend_state {
+	SOF_SUSPEND_NONE = 0,
+	SOF_SUSPEND_S0IX,
+	SOF_SUSPEND_S3,
 };
 
 struct snd_sof_dev;
···
 	snd_pcm_uframes_t (*pcm_pointer)(struct snd_sof_dev *sdev,
 					 struct snd_pcm_substream *substream); /* optional */
 
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES)
+	/* Except for probe_pointer, all probe ops are mandatory */
+	int (*probe_assign)(struct snd_sof_dev *sdev,
+			struct snd_compr_stream *cstream,
+			struct snd_soc_dai *dai); /* mandatory */
+	int (*probe_free)(struct snd_sof_dev *sdev,
+			struct snd_compr_stream *cstream,
+			struct snd_soc_dai *dai); /* mandatory */
+	int (*probe_set_params)(struct snd_sof_dev *sdev,
+			struct snd_compr_stream *cstream,
+			struct snd_compr_params *params,
+			struct snd_soc_dai *dai); /* mandatory */
+	int (*probe_trigger)(struct snd_sof_dev *sdev,
+			struct snd_compr_stream *cstream, int cmd,
+			struct snd_soc_dai *dai); /* mandatory */
+	int (*probe_pointer)(struct snd_sof_dev *sdev,
+			struct snd_compr_stream *cstream,
+			struct snd_compr_tstamp *tstamp,
+			struct snd_soc_dai *dai); /* optional */
+#endif
+
 	/* host read DSP stream data */
 	void (*ipc_msg_data)(struct snd_sof_dev *sdev,
 			     struct snd_pcm_substream *substream,
···
 	int (*post_fw_run)(struct snd_sof_dev *sof_dev); /* optional */
 
 	/* DSP PM */
-	int (*suspend)(struct snd_sof_dev *sof_dev); /* optional */
+	int (*suspend)(struct snd_sof_dev *sof_dev,
+		       u32 target_state); /* optional */
 	int (*resume)(struct snd_sof_dev *sof_dev); /* optional */
 	int (*runtime_suspend)(struct snd_sof_dev *sof_dev); /* optional */
 	int (*runtime_resume)(struct snd_sof_dev *sof_dev); /* optional */
 	int (*runtime_idle)(struct snd_sof_dev *sof_dev); /* optional */
 	int (*set_hw_params_upon_resume)(struct snd_sof_dev *sdev); /* optional */
 	int (*set_power_state)(struct snd_sof_dev *sdev,
-			       enum sof_d0_substate d0_substate); /* optional */
+			       const struct sof_dsp_power_state *target_state); /* optional */
 
 	/* DSP clocking */
 	int (*set_clk)(struct snd_sof_dev *sof_dev, u32 freq); /* optional */
···
 	 */
 	struct snd_soc_component_driver plat_drv;
 
-	/* power states related */
-	enum sof_d0_substate d0_substate;
-	/* flag to track if the intended power target of suspend is S0ix */
-	bool s0_suspend;
+	/* current DSP power state */
+	struct sof_dsp_power_state dsp_power_state;
+
+	/* Intended power target of system suspend */
+	enum sof_system_suspend_state system_suspend_target;
 
 	/* DSP firmware boot */
 	wait_queue_head_t boot_wait;
···
 	u32 enabled_cores_mask; /* keep track of enabled cores */
 
 	/* FW configuration */
-	struct sof_ipc_dma_buffer_data *info_buffer;
 	struct sof_ipc_window *info_window;
 
 	/* IPC timeouts in ms */
 	int ipc_timeout;
 	int boot_timeout;
 
-	/* Wait queue for code loading */
-	wait_queue_head_t waitq;
-	int code_loading;
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES)
+	unsigned int extractor_stream_tag;
+#endif
 
 	/* DMA for Trace */
 	struct snd_dma_buffer dmatb;
···
 int snd_sof_suspend(struct device *dev);
 int snd_sof_prepare(struct device *dev);
 void snd_sof_complete(struct device *dev);
-int snd_sof_set_d0_substate(struct snd_sof_dev *sdev,
-			    enum sof_d0_substate d0_substate);
 
 void snd_sof_new_platform_drv(struct snd_sof_dev *sdev);
 
···
 int sof_ipc_tx_message(struct snd_sof_ipc *ipc, u32 header,
 		       void *msg_data, size_t msg_bytes, void *reply_data,
 		       size_t reply_bytes);
+int sof_ipc_tx_message_no_pm(struct snd_sof_ipc *ipc, u32 header,
+			     void *msg_data, size_t msg_bytes,
+			     void *reply_data, size_t reply_bytes);
 
 /*
  * Trace/debug

sound/soc/sof/topology.c

···
 //
 
 #include <linux/firmware.h>
+#include <linux/workqueue.h>
 #include <sound/tlv.h>
 #include <sound/pcm_params.h>
 #include <uapi/sound/sof/tokens.h>
···
 {
 	struct snd_soc_card *card = scomp->card;
 	struct snd_soc_pcm_runtime *rtd;
+	struct snd_soc_dai *cpu_dai;
+	int i;
 
 	list_for_each_entry(rtd, &card->rtd_list, list) {
 		dev_vdbg(scomp->dev, "tplg: check widget: %s stream: %s dai stream: %s\n",
···
 
 		switch (w->id) {
 		case snd_soc_dapm_dai_out:
-			rtd->cpu_dai->capture_widget = w;
+			for_each_rtd_cpu_dais(rtd, i, cpu_dai)
+				cpu_dai->capture_widget = w;
 			dai->name = rtd->dai_link->name;
 			dev_dbg(scomp->dev, "tplg: connected widget %s -> DAI link %s\n",
 				w->name, rtd->dai_link->name);
 			break;
 		case snd_soc_dapm_dai_in:
-			rtd->cpu_dai->playback_widget = w;
+			for_each_rtd_cpu_dais(rtd, i, cpu_dai)
+				cpu_dai->playback_widget = w;
 			dai->name = rtd->dai_link->name;
 			dev_dbg(scomp->dev, "tplg: connected widget %s -> DAI link %s\n",
 				w->name, rtd->dai_link->name);
···
 	struct snd_soc_tplg_stream_caps *caps;
 	struct snd_soc_tplg_private *private = &pcm->priv;
 	struct snd_sof_pcm *spcm;
-	int stream = SNDRV_PCM_STREAM_PLAYBACK;
+	int stream;
 	int ret = 0;
 
 	/* nothing to do for BEs atm */
···
 		return -ENOMEM;
 
 	spcm->scomp = scomp;
-	spcm->stream[SNDRV_PCM_STREAM_PLAYBACK].comp_id = COMP_ID_UNASSIGNED;
-	spcm->stream[SNDRV_PCM_STREAM_CAPTURE].comp_id = COMP_ID_UNASSIGNED;
+
+	for_each_pcm_streams(stream) {
+		spcm->stream[stream].comp_id = COMP_ID_UNASSIGNED;
+		INIT_WORK(&spcm->stream[stream].period_elapsed_work,
+			  snd_sof_pcm_period_elapsed_work);
+	}
 
 	spcm->pcm = *pcm;
 	dev_dbg(scomp->dev, "tplg: load pcm %s\n", pcm->dai_name);
···
 	if (!spcm->pcm.playback)
 		goto capture;
 
+	stream = SNDRV_PCM_STREAM_PLAYBACK;
+
 	dev_vdbg(scomp->dev, "tplg: pcm %s stream tokens: playback d0i3:%d\n",
-		 spcm->pcm.pcm_name, spcm->stream[0].d0i3_compatible);
+		 spcm->pcm.pcm_name, spcm->stream[stream].d0i3_compatible);
 
 	caps = &spcm->pcm.caps[stream];
 
···
 		return ret;
 
 	dev_vdbg(scomp->dev, "tplg: pcm %s stream tokens: capture d0i3:%d\n",
-		 spcm->pcm.pcm_name, spcm->stream[1].d0i3_compatible);
+		 spcm->pcm.pcm_name, spcm->stream[stream].d0i3_compatible);
 
 	caps = &spcm->pcm.caps[stream];
 

sound/soc/sprd/Kconfig

···
 	  the Spreadtrum SoCs' Audio interfaces.
 
 config SND_SOC_SPRD_MCDT
-	bool "Spreadtrum multi-channel data transfer support"
+	tristate "Spreadtrum multi-channel data transfer support"
 	depends on SND_SOC_SPRD
 	help
 	  Say y here to enable multi-channel data transfer support. It

sound/soc/sprd/sprd-mcdt.h

···
 	struct list_head list;
 };
 
-#ifdef CONFIG_SND_SOC_SPRD_MCDT
+#if IS_ENABLED(CONFIG_SND_SOC_SPRD_MCDT)
 struct sprd_mcdt_chan *sprd_mcdt_request_chan(u8 channel,
 					      enum sprd_mcdt_channel_type type);
 void sprd_mcdt_free_chan(struct sprd_mcdt_chan *chan);

sound/soc/sprd/sprd-pcm-compress.c

···
 	struct snd_soc_component *component =
 		snd_soc_rtdcom_lookup(rtd, DRV_NAME);
 	struct device *dev = component->dev;
-	struct sprd_compr_data *data = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+	struct sprd_compr_data *data = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
 	struct sprd_pcm_dma_params *dma_params = data->dma_params;
 	struct sprd_compr_dma *dma = &stream->dma[channel];
 	struct dma_slave_config config = { };
···
 	struct snd_soc_component *component =
 		snd_soc_rtdcom_lookup(rtd, DRV_NAME);
 	struct device *dev = component->dev;
-	struct sprd_compr_data *data = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+	struct sprd_compr_data *data = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
 	struct sprd_compr_stream *stream;
 	struct sprd_compr_callback cb;
 	int stream_id = cstream->direction, ret;

sound/soc/sprd/sprd-pcm-dma.c

···
 	unsigned long flags;
 	int ret, i, j, sg_num;
 
-	dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+	dma_params = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream);
 	if (!dma_params) {
 		dev_warn(component->dev, "no dma parameters setting\n");
 		dma_private->params = NULL;

sound/soc/stm/stm32_adfsdm.c

···
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct stm32_adfsdm_priv *priv =
-		snd_soc_dai_get_drvdata(rtd->cpu_dai);
+		snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
 
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
···
 				 struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct stm32_adfsdm_priv *priv = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+	struct stm32_adfsdm_priv *priv = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
 	int ret;
 
 	ret =  snd_soc_set_runtime_hwparams(substream, &stm32_adfsdm_pcm_hw);
···
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct stm32_adfsdm_priv *priv =
-		snd_soc_dai_get_drvdata(rtd->cpu_dai);
+		snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
 
 	priv->substream = NULL;
 
···
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct stm32_adfsdm_priv *priv =
-		snd_soc_dai_get_drvdata(rtd->cpu_dai);
+		snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
 
 	return bytes_to_frames(substream->runtime, priv->pos);
 }
···
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct stm32_adfsdm_priv *priv =
-		snd_soc_dai_get_drvdata(rtd->cpu_dai);
+		snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
 
 	priv->pcm_buff = substream->runtime->dma_area;
 
···
 {
 	struct snd_pcm *pcm = rtd->pcm;
 	struct stm32_adfsdm_priv *priv =
-		snd_soc_dai_get_drvdata(rtd->cpu_dai);
+		snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
 	unsigned int size = DFSDM_MAX_PERIODS * DFSDM_MAX_PERIOD_SIZE;
 
 	snd_pcm_set_managed_buffer_all(pcm, SNDRV_DMA_TYPE_DEV,

sound/soc/stm/stm32_i2s.c

···
 	/* Get clocks */
 	i2s->pclk = devm_clk_get(&pdev->dev, "pclk");
 	if (IS_ERR(i2s->pclk)) {
-		dev_err(&pdev->dev, "Could not get pclk\n");
+		if (PTR_ERR(i2s->pclk) != -EPROBE_DEFER)
+			dev_err(&pdev->dev, "Could not get pclk: %ld\n",
+				PTR_ERR(i2s->pclk));
 		return PTR_ERR(i2s->pclk);
 	}
 
 	i2s->i2sclk = devm_clk_get(&pdev->dev, "i2sclk");
 	if (IS_ERR(i2s->i2sclk)) {
-		dev_err(&pdev->dev, "Could not get i2sclk\n");
+		if (PTR_ERR(i2s->i2sclk) != -EPROBE_DEFER)
+			dev_err(&pdev->dev, "Could not get i2sclk: %ld\n",
+				PTR_ERR(i2s->i2sclk));
 		return PTR_ERR(i2s->i2sclk);
 	}
 
 	i2s->x8kclk = devm_clk_get(&pdev->dev, "x8k");
 	if (IS_ERR(i2s->x8kclk)) {
-		dev_err(&pdev->dev, "missing x8k parent clock\n");
+		if (PTR_ERR(i2s->x8kclk) != -EPROBE_DEFER)
+			dev_err(&pdev->dev, "Could not get x8k parent clock: %ld\n",
+				PTR_ERR(i2s->x8kclk));
 		return PTR_ERR(i2s->x8kclk);
 	}
 
 	i2s->x11kclk = devm_clk_get(&pdev->dev, "x11k");
 	if (IS_ERR(i2s->x11kclk)) {
-		dev_err(&pdev->dev, "missing x11k parent clock\n");
+		if (PTR_ERR(i2s->x11kclk) != -EPROBE_DEFER)
+			dev_err(&pdev->dev, "Could not get x11k parent clock: %ld\n",
+				PTR_ERR(i2s->x11kclk));
 		return PTR_ERR(i2s->x11kclk);
 	}
 
···
 	}
 
 	/* Reset */
-	rst = devm_reset_control_get_exclusive(&pdev->dev, NULL);
-	if (!IS_ERR(rst)) {
-		reset_control_assert(rst);
-		udelay(2);
-		reset_control_deassert(rst);
+	rst = devm_reset_control_get_optional_exclusive(&pdev->dev, NULL);
+	if (IS_ERR(rst)) {
+		if (PTR_ERR(rst) != -EPROBE_DEFER)
+			dev_err(&pdev->dev, "Reset controller error %ld\n",
+				PTR_ERR(rst));
+		return PTR_ERR(rst);
 	}
+	reset_control_assert(rst);
+	udelay(2);
+	reset_control_deassert(rst);
+
+	return 0;
+}
+
+static int stm32_i2s_remove(struct platform_device *pdev)
+{
+	snd_dmaengine_pcm_unregister(&pdev->dev);
+	snd_soc_unregister_component(&pdev->dev);
 
 	return 0;
 }
···
 	i2s->regmap = devm_regmap_init_mmio_clk(&pdev->dev, "pclk",
 						i2s->base, i2s->regmap_conf);
 	if (IS_ERR(i2s->regmap)) {
-		dev_err(&pdev->dev, "regmap init failed\n");
+		if (PTR_ERR(i2s->regmap) != -EPROBE_DEFER)
+			dev_err(&pdev->dev, "Regmap init error %ld\n",
+				PTR_ERR(i2s->regmap));
 		return PTR_ERR(i2s->regmap);
 	}
 
-	ret = devm_snd_soc_register_component(&pdev->dev, &stm32_i2s_component,
-					      i2s->dai_drv, 1);
-	if (ret)
+	ret = snd_dmaengine_pcm_register(&pdev->dev, &stm32_i2s_pcm_config, 0);
+	if (ret) {
+		if (ret != -EPROBE_DEFER)
+			dev_err(&pdev->dev, "PCM DMA register error %d\n", ret);
 		return ret;
+	}
 
-	ret = devm_snd_dmaengine_pcm_register(&pdev->dev,
-					      &stm32_i2s_pcm_config, 0);
-	if (ret)
+	ret = snd_soc_register_component(&pdev->dev, &stm32_i2s_component,
+					 i2s->dai_drv, 1);
+	if (ret) {
+		snd_dmaengine_pcm_unregister(&pdev->dev);
 		return ret;
+	}
 
 	/* Set SPI/I2S in i2s mode */
 	ret = regmap_update_bits(i2s->regmap, STM32_I2S_CGFR_REG,
 				 I2S_CGFR_I2SMOD, I2S_CGFR_I2SMOD);
 	if (ret)
-		return ret;
+		goto error;
 
 	ret = regmap_read(i2s->regmap, STM32_I2S_IPIDR_REG, &val);
 	if (ret)
-		return ret;
+		goto error;
 
 	if (val == I2S_IPIDR_NUMBER) {
 		ret = regmap_read(i2s->regmap, STM32_I2S_HWCFGR_REG, &val);
 		if (ret)
-			return ret;
+			goto error;
 
 		if (!FIELD_GET(I2S_HWCFGR_I2S_SUPPORT_MASK, val)) {
 			dev_err(&pdev->dev,
 				"Device does not support i2s mode\n");
-			return -EPERM;
+			ret = -EPERM;
+			goto error;
 		}
 
 		ret = regmap_read(i2s->regmap, STM32_I2S_VERR_REG, &val);
+		if (ret)
+			goto error;
 
 		dev_dbg(&pdev->dev, "I2S version: %lu.%lu registered\n",
 			FIELD_GET(I2S_VERR_MAJ_MASK, val),
 			FIELD_GET(I2S_VERR_MIN_MASK, val));
 	}
+
+	return ret;
+
+error:
+	stm32_i2s_remove(pdev);
 
 	return ret;
 }
···
 		.pm = &stm32_i2s_pm_ops,
 	},
 	.probe = stm32_i2s_probe,
+	.remove = stm32_i2s_remove,
 };
 
 module_platform_driver(stm32_i2s_driver);

sound/soc/stm/stm32_sai.c

···
 	if (!STM_SAI_IS_F4(sai)) {
 		sai->pclk = devm_clk_get(&pdev->dev, "pclk");
 		if (IS_ERR(sai->pclk)) {
-			dev_err(&pdev->dev, "missing bus clock pclk\n");
+			if (PTR_ERR(sai->pclk) != -EPROBE_DEFER)
+				dev_err(&pdev->dev, "missing bus clock pclk: %ld\n",
+					PTR_ERR(sai->pclk));
 			return PTR_ERR(sai->pclk);
 		}
 	}
 
 	sai->clk_x8k = devm_clk_get(&pdev->dev, "x8k");
 	if (IS_ERR(sai->clk_x8k)) {
-		dev_err(&pdev->dev, "missing x8k parent clock\n");
+		if (PTR_ERR(sai->clk_x8k) != -EPROBE_DEFER)
+			dev_err(&pdev->dev, "missing x8k parent clock: %ld\n",
+				PTR_ERR(sai->clk_x8k));
 		return PTR_ERR(sai->clk_x8k);
 	}
 
 	sai->clk_x11k = devm_clk_get(&pdev->dev, "x11k");
 	if (IS_ERR(sai->clk_x11k)) {
-		dev_err(&pdev->dev, "missing x11k parent clock\n");
+		if (PTR_ERR(sai->clk_x11k) != -EPROBE_DEFER)
+			dev_err(&pdev->dev, "missing x11k parent clock: %ld\n",
+				PTR_ERR(sai->clk_x11k));
 		return PTR_ERR(sai->clk_x11k);
 	}
 
···
 		return sai->irq;
 
 	/* reset */
-	rst = devm_reset_control_get_exclusive(&pdev->dev, NULL);
-	if (!IS_ERR(rst)) {
-		reset_control_assert(rst);
-		udelay(2);
-		reset_control_deassert(rst);
+	rst = devm_reset_control_get_optional_exclusive(&pdev->dev, NULL);
+	if (IS_ERR(rst)) {
+		if (PTR_ERR(rst) != -EPROBE_DEFER)
+			dev_err(&pdev->dev, "Reset controller error %ld\n",
+				PTR_ERR(rst));
+		return PTR_ERR(rst);
 	}
+	reset_control_assert(rst);
+	udelay(2);
+	reset_control_deassert(rst);
 
 	/* Enable peripheral clock to allow register access */
 	ret = clk_prepare_enable(sai->pclk);

sound/soc/stm/stm32_sai_sub.c

···
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	struct stm32_sai_sub_data *sai = dev_get_drvdata(cpu_dai->dev);
 	int *ptr = (int *)(runtime->dma_area + hwoff +
 			   channel * (runtime->dma_bytes / runtime->channels));
···
 	sai->regmap = devm_regmap_init_mmio(&pdev->dev, base,
 					    sai->regmap_config);
 	if (IS_ERR(sai->regmap)) {
-		dev_err(&pdev->dev, "Failed to initialize MMIO\n");
+		if (PTR_ERR(sai->regmap) != -EPROBE_DEFER)
+			dev_err(&pdev->dev, "Regmap init error %ld\n",
+				PTR_ERR(sai->regmap));
 		return PTR_ERR(sai->regmap);
 	}
 
···
 	of_node_put(args.np);
 	sai->sai_ck = devm_clk_get(&pdev->dev, "sai_ck");
 	if (IS_ERR(sai->sai_ck)) {
-		dev_err(&pdev->dev, "Missing kernel clock sai_ck\n");
+		if (PTR_ERR(sai->sai_ck) != -EPROBE_DEFER)
+			dev_err(&pdev->dev, "Missing kernel clock sai_ck: %ld\n",
+				PTR_ERR(sai->sai_ck));
 		return PTR_ERR(sai->sai_ck);
 	}
 
···
 
 	ret = snd_dmaengine_pcm_register(&pdev->dev, conf, 0);
 	if (ret) {
-		dev_err(&pdev->dev, "Could not register pcm dma\n");
+		if (ret != -EPROBE_DEFER)
+			dev_err(&pdev->dev, "Could not register pcm dma\n");
 		return ret;
 	}
 

sound/soc/stm/stm32_spdifrx.c

···
 
 	spdifrx->ctrl_chan = dma_request_chan(dev, "rx-ctrl");
 	if (IS_ERR(spdifrx->ctrl_chan)) {
-		dev_err(dev, "dma_request_slave_channel failed\n");
+		if (PTR_ERR(spdifrx->ctrl_chan) != -EPROBE_DEFER)
+			dev_err(dev, "dma_request_slave_channel error %ld\n",
+				PTR_ERR(spdifrx->ctrl_chan));
 		return PTR_ERR(spdifrx->ctrl_chan);
 	}
 
···
 
 	spdifrx->kclk = devm_clk_get(&pdev->dev, "kclk");
 	if (IS_ERR(spdifrx->kclk)) {
-		dev_err(&pdev->dev, "Could not get kclk\n");
+		if (PTR_ERR(spdifrx->kclk) != -EPROBE_DEFER)
+			dev_err(&pdev->dev, "Could not get kclk: %ld\n",
+				PTR_ERR(spdifrx->kclk));
 		return PTR_ERR(spdifrx->kclk);
 	}
 
 	spdifrx->irq = platform_get_irq(pdev, 0);
 	if (spdifrx->irq < 0)
 		return spdifrx->irq;
+
+	return 0;
+}
+
+static int stm32_spdifrx_remove(struct platform_device *pdev)
+{
+	struct stm32_spdifrx_data *spdifrx = platform_get_drvdata(pdev);
+
+	if (spdifrx->ctrl_chan)
+		dma_release_channel(spdifrx->ctrl_chan);
+
+	if (spdifrx->dmab)
+		snd_dma_free_pages(spdifrx->dmab);
+
+	snd_dmaengine_pcm_unregister(&pdev->dev);
+	snd_soc_unregister_component(&pdev->dev);
 
 	return 0;
 }
···
 						    spdifrx->base,
 						    spdifrx->regmap_conf);
 	if (IS_ERR(spdifrx->regmap)) {
-		dev_err(&pdev->dev, "Regmap init failed\n");
+		if (PTR_ERR(spdifrx->regmap) != -EPROBE_DEFER)
+			dev_err(&pdev->dev, "Regmap init error %ld\n",
+				PTR_ERR(spdifrx->regmap));
 		return PTR_ERR(spdifrx->regmap);
 	}
 
···
 		return ret;
 	}
 
-	rst = devm_reset_control_get_exclusive(&pdev->dev, NULL);
-	if (!IS_ERR(rst)) {
-		reset_control_assert(rst);
-		udelay(2);
-		reset_control_deassert(rst);
+	rst = devm_reset_control_get_optional_exclusive(&pdev->dev, NULL);
+	if (IS_ERR(rst)) {
+		if (PTR_ERR(rst) != -EPROBE_DEFER)
+			dev_err(&pdev->dev, "Reset controller error %ld\n",
+				PTR_ERR(rst));
+		return PTR_ERR(rst);
+	}
+	reset_control_assert(rst);
+	udelay(2);
+	reset_control_deassert(rst);
+
+	pcm_config = &stm32_spdifrx_pcm_config;
+	ret = snd_dmaengine_pcm_register(&pdev->dev, pcm_config, 0);
+	if (ret) {
+		if (ret != -EPROBE_DEFER)
+			dev_err(&pdev->dev, "PCM DMA register error %d\n", ret);
+		return ret;
 	}
 
-	ret = devm_snd_soc_register_component(&pdev->dev,
-					      &stm32_spdifrx_component,
-					      stm32_spdifrx_dai,
-					      ARRAY_SIZE(stm32_spdifrx_dai));
-	if (ret)
+	ret = snd_soc_register_component(&pdev->dev,
+					 &stm32_spdifrx_component,
+					 stm32_spdifrx_dai,
+					 ARRAY_SIZE(stm32_spdifrx_dai));
+	if (ret) {
+		snd_dmaengine_pcm_unregister(&pdev->dev);
 		return ret;
+	}
 
 	ret = stm32_spdifrx_dma_ctrl_register(&pdev->dev, spdifrx);
 	if (ret)
 		goto error;
-
-	pcm_config = &stm32_spdifrx_pcm_config;
-	ret = devm_snd_dmaengine_pcm_register(&pdev->dev, pcm_config, 0);
-	if (ret) {
-		dev_err(&pdev->dev, "PCM DMA register returned %d\n", ret);
-		goto error;
-	}
 
 	ret = regmap_read(spdifrx->regmap, STM32_SPDIFRX_IDR, &idr);
 	if (ret)
···
 
 	if (idr == SPDIFRX_IPIDR_NUMBER) {
 		ret = regmap_read(spdifrx->regmap, STM32_SPDIFRX_VERR, &ver);
+		if (ret)
+			goto error;
 
 		dev_dbg(&pdev->dev, "SPDIFRX version: %lu.%lu registered\n",
 			FIELD_GET(SPDIFRX_VERR_MAJ_MASK, ver),
···
 	return ret;
 
 error:
-	if (!IS_ERR(spdifrx->ctrl_chan))
-		dma_release_channel(spdifrx->ctrl_chan);
-	if (spdifrx->dmab)
-		snd_dma_free_pages(spdifrx->dmab);
+	stm32_spdifrx_remove(pdev);
 
 	return ret;
-}
-
-static int stm32_spdifrx_remove(struct platform_device *pdev)
-{
-	struct stm32_spdifrx_data *spdifrx = platform_get_drvdata(pdev);
-
-	if (spdifrx->ctrl_chan)
-		dma_release_channel(spdifrx->ctrl_chan);
-
-	if (spdifrx->dmab)
-		snd_dma_free_pages(spdifrx->dmab);
-
-	return 0;
 }
 
 MODULE_DEVICE_TABLE(of, stm32_spdifrx_ids);

sound/soc/sunxi/sun4i-spdif.c

···
 			       struct snd_soc_dai *cpu_dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct sun4i_spdif_dev *host = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+	struct sun4i_spdif_dev *host = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
 
 	if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
 		return -EINVAL;

sound/soc/sunxi/sun8i-codec.c

···
 #define SUN8I_AIF1CLK_CTRL_AIF1_BCLK_DIV_MASK	GENMASK(12, 9)
 
 struct sun8i_codec {
-	struct device	*dev;
 	struct regmap	*regmap;
 	struct clk	*clk_module;
 	struct clk	*clk_bus;
···
 	scodec = devm_kzalloc(&pdev->dev, sizeof(*scodec), GFP_KERNEL);
 	if (!scodec)
 		return -ENOMEM;
-
-	scodec->dev = &pdev->dev;
 
 	scodec->clk_module = devm_clk_get(&pdev->dev, "mod");
 	if (IS_ERR(scodec->clk_module)) {

sound/soc/tegra/tegra_alc5632.c

···
 					struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	struct snd_soc_card *card = rtd->card;
 	struct tegra_alc5632 *alc5632 = snd_soc_card_get_drvdata(card);
 	int srate, mclk;

sound/soc/tegra/tegra_max98090.c

···
 					struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	struct snd_soc_card *card = rtd->card;
 	struct tegra_max98090 *machine = snd_soc_card_get_drvdata(card);
 	int srate, mclk;

sound/soc/tegra/tegra_rt5640.c

···
 					struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	struct snd_soc_card *card = rtd->card;
 	struct tegra_rt5640 *machine = snd_soc_card_get_drvdata(card);
 	int srate, mclk;

sound/soc/tegra/tegra_rt5677.c

···
 					struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	struct snd_soc_card *card = rtd->card;
 	struct tegra_rt5677 *machine = snd_soc_card_get_drvdata(card);
 	int srate, mclk, err;

sound/soc/tegra/tegra_sgtl5000.c

···
 					struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	struct snd_soc_card *card = rtd->card;
 	struct tegra_sgtl5000 *machine = snd_soc_card_get_drvdata(card);
 	int srate, mclk;

sound/soc/tegra/tegra_wm8753.c

···
 					struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	struct snd_soc_card *card = rtd->card;
 	struct tegra_wm8753 *machine = snd_soc_card_get_drvdata(card);
 	int srate, mclk;

sound/soc/tegra/tegra_wm8903.c

···
 					struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	struct snd_soc_card *card = rtd->card;
 	struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card);
 	int srate, mclk;
···
 	return 0;
 }
 
+static int tegra_wm8903_event_int_mic(struct snd_soc_dapm_widget *w,
+				      struct snd_kcontrol *k, int event)
+{
+	struct snd_soc_dapm_context *dapm = w->dapm;
+	struct snd_soc_card *card = dapm->card;
+	struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card);
+
+	if (!gpio_is_valid(machine->gpio_int_mic_en))
+		return 0;
+
+	gpio_set_value_cansleep(machine->gpio_int_mic_en,
+				SND_SOC_DAPM_EVENT_ON(event));
+
+	return 0;
+}
+
 static const struct snd_soc_dapm_widget tegra_wm8903_dapm_widgets[] = {
 	SND_SOC_DAPM_SPK("Int Spk", tegra_wm8903_event_int_spk),
 	SND_SOC_DAPM_HP("Headphone Jack", tegra_wm8903_event_hp),
 	SND_SOC_DAPM_MIC("Mic Jack", NULL),
+	SND_SOC_DAPM_MIC("Int Mic", tegra_wm8903_event_int_mic),
 };
 
 static const struct snd_kcontrol_new tegra_wm8903_controls[] = {
 	SOC_DAPM_PIN_SWITCH("Int Spk"),
+	SOC_DAPM_PIN_SWITCH("Int Mic"),
 };
 
 static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd)
 {
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	struct snd_soc_component *component = codec_dai->component;
 	struct snd_soc_card *card = rtd->card;
 	struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card);
···
 {
 	struct snd_soc_pcm_runtime *rtd =
 		snd_soc_get_pcm_runtime(card, &card->dai_link[0]);
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	struct snd_soc_component *component = codec_dai->component;
 
 	wm8903_mic_detect(component, NULL, 0, 0);

sound/soc/tegra/trimslice.c

···
 					struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	struct snd_soc_card *card = rtd->card;
 	struct tegra_trimslice *trimslice = snd_soc_card_get_drvdata(card);
 	int srate, mclk;

sound/soc/ti/Kconfig

···
 # SPDX-License-Identifier: GPL-2.0-only
 menu "Audio support for Texas Instruments SoCs"
-depends on DMA_OMAP || TI_EDMA || COMPILE_TEST
+depends on DMA_OMAP || TI_EDMA || TI_K3_UDMA || COMPILE_TEST
 
 config SND_SOC_TI_EDMA_PCM
 	tristate
 	select SND_SOC_GENERIC_DMAENGINE_PCM
 
 config SND_SOC_TI_SDMA_PCM
+	tristate
+	select SND_SOC_GENERIC_DMAENGINE_PCM
+
+config SND_SOC_TI_UDMA_PCM
 	tristate
 	select SND_SOC_GENERIC_DMAENGINE_PCM
 
···
 	tristate "Multichannel Audio Serial Port (McASP) support"
 	select SND_SOC_TI_EDMA_PCM
 	select SND_SOC_TI_SDMA_PCM
+	select SND_SOC_TI_UDMA_PCM
 	help
 	  Say Y or M here if you want to have support for McASP IP found in
 	  various Texas Instruments SoCs like:
···
 	  - Sitara line of SoCs (AM335x, AM438x, etc)
 	  - DRA7x devices
 	  - Keystone devices
+	  - K3 devices (am654, j721e)
 
 config SND_SOC_DAVINCI_VCIF
 	tristate "daVinci Voice Interface (VCIF) support"

sound/soc/ti/Makefile

···
 # Platform drivers
 snd-soc-ti-edma-objs := edma-pcm.o
 snd-soc-ti-sdma-objs := sdma-pcm.o
+snd-soc-ti-udma-objs := udma-pcm.o
 
 obj-$(CONFIG_SND_SOC_TI_EDMA_PCM) += snd-soc-ti-edma.o
 obj-$(CONFIG_SND_SOC_TI_SDMA_PCM) += snd-soc-ti-sdma.o
+obj-$(CONFIG_SND_SOC_TI_UDMA_PCM) += snd-soc-ti-udma.o
 
 # CPU DAI drivers
 snd-soc-davinci-asp-objs := davinci-i2s.o

sound/soc/ti/ams-delta.c

···
 
 static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd)
 {
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	struct snd_soc_card *card = rtd->card;
 	struct snd_soc_dapm_context *dapm = &card->dapm;
 	int ret;
 	/* Codec is ready, now add/activate board specific controls */
 
 	/* Store a pointer to the codec structure for tty ldisc use */
-	cx20442_codec = rtd->codec_dai->component;
+	cx20442_codec = asoc_rtd_to_codec(rtd, 0)->component;
 
 	/* Add hook switch - can be used to control the codec from userspace
 	 * even if line discipline fails */

sound/soc/ti/davinci-evm.c

···
 			 struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	struct snd_soc_card *soc_card = rtd->card;
 	int ret = 0;
 	unsigned sysclk = ((struct snd_soc_card_drvdata_davinci *)

sound/soc/ti/davinci-mcasp.c

···
 
 #include "edma-pcm.h"
 #include "sdma-pcm.h"
+#include "udma-pcm.h"
 #include "davinci-mcasp.h"
 
 #define MCASP_MAX_AFIFO_DEPTH	64
···
 	} else if (match) {
 		pdata = devm_kmemdup(&pdev->dev, match->data, sizeof(*pdata),
 				     GFP_KERNEL);
-		if (!pdata) {
-			ret = -ENOMEM;
-			return pdata;
-		}
+		if (!pdata)
+			return NULL;
 	} else {
 		/* control shouldn't reach here. something is wrong */
 		ret = -EINVAL;
···
 enum {
 	PCM_EDMA,
 	PCM_SDMA,
+	PCM_UDMA,
 };
 static const char *sdma_prefix = "ti,omap";
 
···
 	dev_dbg(mcasp->dev, "DMA controller compatible = \"%s\"\n", tmp);
 	if (!strncmp(tmp, sdma_prefix, strlen(sdma_prefix)))
 		return PCM_SDMA;
+	else if (strstr(tmp, "udmap"))
+		return PCM_UDMA;
 
 	return PCM_EDMA;
 }
···
 		break;
 	case PCM_SDMA:
 		ret = sdma_pcm_platform_register(&pdev->dev, "tx", "rx");
+		break;
+	case PCM_UDMA:
+		ret = udma_pcm_platform_register(&pdev->dev);
 		break;
 	default:
 		dev_err(&pdev->dev, "No DMA controller found (%d)\n", ret);

sound/soc/ti/davinci-vcif.c

···
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct davinci_vcif_dev *davinci_vcif_dev =
-			snd_soc_dai_get_drvdata(rtd->cpu_dai);
+			snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
 	struct davinci_vc *davinci_vc = davinci_vcif_dev->davinci_vc;
 	u32 w;
 
···
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct davinci_vcif_dev *davinci_vcif_dev =
-			snd_soc_dai_get_drvdata(rtd->cpu_dai);
+			snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
 	struct davinci_vc *davinci_vc = davinci_vcif_dev->davinci_vc;
 	u32 w;
 

sound/soc/ti/n810.c

···
 	struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	int err;
 
 	/* Set the codec system clock for DAC and ADC */

sound/soc/ti/omap-abe-twl6040.c

···
 	struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	struct snd_soc_card *card = rtd->card;
 	struct abe_twl6040 *priv = snd_soc_card_get_drvdata(card);
 	int clk_id, freq;
···
 	struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	int ret = 0;
 
 	ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_DMIC_SYSCLK_PAD_CLKS,
···
 
 static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd)
 {
-	struct snd_soc_component *component = rtd->codec_dai->component;
+	struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
 	struct snd_soc_card *card = rtd->card;
 	struct abe_twl6040 *priv = snd_soc_card_get_drvdata(card);
 	int hs_trim;

sound/soc/ti/omap-mcbsp-st.c

···
 
 int omap_mcbsp_st_add_controls(struct snd_soc_pcm_runtime *rtd, int port_id)
 {
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai);
 
 	if (!mcbsp->st_data) {

sound/soc/ti/omap-mcbsp.c

···
 		unsigned int packet_size)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai);
 	int words;
 
···
 			struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai);
 	u16 fifo_use;
 	snd_pcm_sframes_t delay;

sound/soc/ti/omap-mcpdm.c

···
 void omap_mcpdm_configure_dn_offsets(struct snd_soc_pcm_runtime *rtd,
 				    u8 rx1, u8 rx2)
 {
-	struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+	struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
 
 	mcpdm->dn_rx_offset = MCPDM_DNOFST_RX1(rx1) | MCPDM_DNOFST_RX2(rx2);
 }

sound/soc/ti/omap3pandora.c

···
 	struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	int ret;
 
 	/* Set the codec system clock for DAC and ADC */

sound/soc/ti/osk5912.c

···
 			 struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	int err;
 
 	/* Set the codec system clock for DAC and ADC */

sound/soc/ti/rx51.c

···
 	struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 
 	/* Set the codec system clock for DAC and ADC */
 	return snd_soc_dai_set_sysclk(codec_dai, 0, 19200000,

sound/soc/ti/udma-pcm.c

···
+// SPDX-License-Identifier: GPL-2.0
+/*
+ *  Copyright (C) 2020 Texas Instruments Incorporated - http://www.ti.com
+ *  Author: Peter Ujfalusi <[e-mail]>
+ */
+
+#include <linux/module.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/dmaengine_pcm.h>
+
+#include "udma-pcm.h"
+
+static const struct snd_pcm_hardware udma_pcm_hardware = {
+	.info			= SNDRV_PCM_INFO_MMAP |
+				  SNDRV_PCM_INFO_MMAP_VALID |
+				  SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME |
+				  SNDRV_PCM_INFO_NO_PERIOD_WAKEUP |
+				  SNDRV_PCM_INFO_INTERLEAVED,
+	.buffer_bytes_max	= SIZE_MAX,
+	.period_bytes_min	= 32,
+	.period_bytes_max	= SZ_64K,
+	.periods_min		= 2,
+	.periods_max		= UINT_MAX,
+};
+
+static const struct snd_dmaengine_pcm_config udma_dmaengine_pcm_config = {
+	.pcm_hardware = &udma_pcm_hardware,
+	.prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config,
+};
+
+int udma_pcm_platform_register(struct device *dev)
+{
+	return devm_snd_dmaengine_pcm_register(dev, &udma_dmaengine_pcm_config,
+					       0);
+}
+EXPORT_SYMBOL_GPL(udma_pcm_platform_register);
+
+MODULE_AUTHOR("Peter Ujfalusi <[e-mail]>");
+MODULE_DESCRIPTION("UDMA PCM ASoC platform driver");
+MODULE_LICENSE("GPL v2");

sound/soc/ti/udma-pcm.h

···
+/* SPDX-License-Identifier: GPL-2.0 */
+/*
+ *  Copyright (C) 2018 Texas Instruments Incorporated - http://www.ti.com
+ */
+
+#ifndef __UDMA_PCM_H__
+#define __UDMA_PCM_H__
+
+#if IS_ENABLED(CONFIG_SND_SOC_TI_UDMA_PCM)
+int udma_pcm_platform_register(struct device *dev);
+#else
+static inline int udma_pcm_platform_register(struct device *dev)
+{
+	return 0;
+}
+#endif /* CONFIG_SND_SOC_TI_UDMA_PCM */
+
+#endif /* __UDMA_PCM_H__ */

sound/soc/txx9/txx9aclc.c

···
 			    struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_card *card = rtd->card->snd_card;
-	struct snd_soc_dai *dai = rtd->cpu_dai;
+	struct snd_soc_dai *dai = asoc_rtd_to_cpu(rtd, 0);
 	struct snd_pcm *pcm = rtd->pcm;
 	struct platform_device *pdev = to_platform_device(component->dev);
 	struct txx9aclc_soc_device *dev;

sound/soc/uniphier/aio-compress.c

···
 {
 	struct snd_compr *compr = rtd->compr;
 	struct device *dev = compr->card->dev;
-	struct uniphier_aio *aio = uniphier_priv(rtd->cpu_dai);
+	struct uniphier_aio *aio = uniphier_priv(asoc_rtd_to_cpu(rtd, 0));
 	struct uniphier_aio_sub *sub = &aio->sub[compr->direction];
 	size_t size = AUD_RING_SIZE;
 	int dma_dir = DMA_FROM_DEVICE, ret;
···
 {
 	struct snd_compr *compr = rtd->compr;
 	struct device *dev = compr->card->dev;
-	struct uniphier_aio *aio = uniphier_priv(rtd->cpu_dai);
+	struct uniphier_aio *aio = uniphier_priv(asoc_rtd_to_cpu(rtd, 0));
 	struct uniphier_aio_sub *sub = &aio->sub[compr->direction];
 	int dma_dir = DMA_FROM_DEVICE;
 
···
 static int uniphier_aio_compr_open(struct snd_compr_stream *cstream)
 {
 	struct snd_soc_pcm_runtime *rtd = cstream->private_data;
-	struct uniphier_aio *aio = uniphier_priv(rtd->cpu_dai);
+	struct uniphier_aio *aio = uniphier_priv(asoc_rtd_to_cpu(rtd, 0));
 	struct uniphier_aio_sub *sub = &aio->sub[cstream->direction];
 	int ret;
 
···
 static int uniphier_aio_compr_free(struct snd_compr_stream *cstream)
 {
 	struct snd_soc_pcm_runtime *rtd = cstream->private_data;
-	struct uniphier_aio *aio = uniphier_priv(rtd->cpu_dai);
+	struct uniphier_aio *aio = uniphier_priv(asoc_rtd_to_cpu(rtd, 0));
 	struct uniphier_aio_sub *sub = &aio->sub[cstream->direction];
 	int ret;
 
···
 					 struct snd_codec *params)
 {
 	struct snd_soc_pcm_runtime *rtd = cstream->private_data;
-	struct uniphier_aio *aio = uniphier_priv(rtd->cpu_dai);
+	struct uniphier_aio *aio = uniphier_priv(asoc_rtd_to_cpu(rtd, 0));
 	struct uniphier_aio_sub *sub = &aio->sub[cstream->direction];
 
 	*params = sub->cparams.codec;
···
 					 struct snd_compr_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = cstream->private_data;
-	struct uniphier_aio *aio = uniphier_priv(rtd->cpu_dai);
+	struct uniphier_aio *aio = uniphier_priv(asoc_rtd_to_cpu(rtd, 0));
 	struct uniphier_aio_sub *sub = &aio->sub[cstream->direction];
 	struct device *dev = &aio->chip->pdev->dev;
 	int ret;
···
 static int uniphier_aio_compr_hw_free(struct snd_compr_stream *cstream)
 {
 	struct snd_soc_pcm_runtime *rtd = cstream->private_data;
-	struct uniphier_aio *aio = uniphier_priv(rtd->cpu_dai);
+	struct uniphier_aio *aio = uniphier_priv(asoc_rtd_to_cpu(rtd, 0));
 	struct uniphier_aio_sub *sub = &aio->sub[cstream->direction];
 
 	sub->setting = 0;
···
 {
 	struct snd_soc_pcm_runtime *rtd = cstream->private_data;
 	struct snd_compr_runtime *runtime = cstream->runtime;
-	struct uniphier_aio *aio = uniphier_priv(rtd->cpu_dai);
+	struct uniphier_aio *aio = uniphier_priv(asoc_rtd_to_cpu(rtd, 0));
 	struct uniphier_aio_sub *sub = &aio->sub[cstream->direction];
 	int bytes = runtime->fragment_size;
 	unsigned long flags;
···
 {
 	struct snd_soc_pcm_runtime *rtd = cstream->private_data;
 	struct snd_compr_runtime *runtime = cstream->runtime;
-	struct uniphier_aio *aio = uniphier_priv(rtd->cpu_dai);
+	struct uniphier_aio *aio = uniphier_priv(asoc_rtd_to_cpu(rtd, 0));
 	struct uniphier_aio_sub *sub = &aio->sub[cstream->direction];
 	struct device *dev = &aio->chip->pdev->dev;
 	int bytes = runtime->fragment_size, ret = 0;
···
 {
 	struct snd_soc_pcm_runtime *rtd = cstream->private_data;
 	struct snd_compr_runtime *runtime = cstream->runtime;
-	struct uniphier_aio *aio = uniphier_priv(rtd->cpu_dai);
+	struct uniphier_aio *aio = uniphier_priv(asoc_rtd_to_cpu(rtd, 0));
 	struct uniphier_aio_sub *sub = &aio->sub[cstream->direction];
 	int bytes = runtime->fragment_size;
 	unsigned long flags;
···
 	struct snd_soc_pcm_runtime *rtd = cstream->private_data;
 	struct snd_compr_runtime *runtime = cstream->runtime;
 	struct device *carddev = rtd->compr->card->dev;
-	struct uniphier_aio *aio = uniphier_priv(rtd->cpu_dai);
+	struct uniphier_aio *aio = uniphier_priv(asoc_rtd_to_cpu(rtd, 0));
 	struct uniphier_aio_sub *sub = &aio->sub[cstream->direction];
 	size_t cnt = min_t(size_t, count, aio_rb_space_to_end(sub) / 2);
 	int bytes = runtime->fragment_size;

sound/soc/uniphier/aio-dma.c

···
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream);
-	struct uniphier_aio *aio = uniphier_priv(rtd->cpu_dai);
+	struct uniphier_aio *aio = uniphier_priv(asoc_rtd_to_cpu(rtd, 0));
 	struct uniphier_aio_sub *sub = &aio->sub[substream->stream];
 	int bytes = runtime->period_size *
 		runtime->channels * samples_to_bytes(runtime, 1);
···
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream);
-	struct uniphier_aio *aio = uniphier_priv(rtd->cpu_dai);
+	struct uniphier_aio *aio = uniphier_priv(asoc_rtd_to_cpu(rtd, 0));
 	struct uniphier_aio_sub *sub = &aio->sub[substream->stream];
 	struct device *dev = &aio->chip->pdev->dev;
 	int bytes = runtime->period_size *
···
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream);
-	struct uniphier_aio *aio = uniphier_priv(rtd->cpu_dai);
+	struct uniphier_aio *aio = uniphier_priv(asoc_rtd_to_cpu(rtd, 0));
 	struct uniphier_aio_sub *sub = &aio->sub[substream->stream];
 	int bytes = runtime->period_size *
 		runtime->channels * samples_to_bytes(runtime, 1);

sound/soc/ux500/mop500_ab8500.c

···
 			struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	struct device *dev = rtd->card->dev;
 	unsigned int fmt;
 	int channels, ret = 0, driver_mode, slots;
···
 static int mop500_ab8500_hw_free(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 
 	mutex_lock(&mop500_ab8500_params_lock);
 	__clear_bit(cpu_dai->id, &mop500_ab8500_usage);

sound/soc/ux500/ux500_pcm.c

···
 static struct dma_chan *ux500_pcm_request_chan(struct snd_soc_pcm_runtime *rtd,
 	struct snd_pcm_substream *substream)
 {
-	struct snd_soc_dai *dai = rtd->cpu_dai;
+	struct snd_soc_dai *dai = asoc_rtd_to_cpu(rtd, 0);
 	u16 per_data_width, mem_data_width;
 	struct stedma40_chan_cfg *dma_cfg;
 	struct ux500_msp_dma_params *dma_params;
···
 		struct dma_slave_config *slave_config)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct msp_i2s_platform_data *pdata = rtd->cpu_dai->dev->platform_data;
+	struct msp_i2s_platform_data *pdata = asoc_rtd_to_cpu(rtd, 0)->dev->platform_data;
 	struct snd_dmaengine_dai_dma_data *snd_dma_params;
 	struct ux500_msp_dma_params *ste_dma_params;
 	dma_addr_t dma_addr;
···
 
 	if (pdata) {
 		ste_dma_params =
-			snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+			snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream);
 		dma_addr = ste_dma_params->tx_rx_addr;
 	} else {
 		snd_dma_params =
-			snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+			snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream);
 		dma_addr = snd_dma_params->addr;
 	}
 

sound/soc/xtensa/xtfpga-i2s.c

···
 	void *p;
 
 	snd_soc_set_runtime_hwparams(substream, &xtfpga_pcm_hardware);
-	p = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+	p = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream);
 	runtime->private_data = p;
 
 	return 0;

sound/soc/zte/zx-spdif.c

···
 	zx_spdif->mapbase = res->start;
 	zx_spdif->reg_base = devm_ioremap_resource(&pdev->dev, res);
 	if (IS_ERR(zx_spdif->reg_base)) {
-		dev_err(&pdev->dev, "ioremap failed!\n");
 		return PTR_ERR(zx_spdif->reg_base);
 	}
 

sound/soc/zte/zx-tdm.c

···
 
 static int zx_tdm_probe(struct platform_device *pdev)
 {
-	struct device *dev = &pdev->dev;
 	struct of_phandle_args out_args;
 	unsigned int dma_reg_offset;
 	struct zx_tdm_info *zx_tdm;
···
 	if (!zx_tdm)
 		return -ENOMEM;
 
-	zx_tdm->dev = dev;
+	zx_tdm->dev = &pdev->dev;
 
 	zx_tdm->dai_wclk = devm_clk_get(&pdev->dev, "wclk");
 	if (IS_ERR(zx_tdm->dai_wclk)) {

sound/usb/Makefile

···
 			mixer_scarlett.o \
 			mixer_scarlett_gen2.o \
 			mixer_us16x08.o \
+			mixer_s1810c.o \
 			pcm.o \
 			power.o \
 			proc.o \

sound/usb/card.c

···
 static bool ignore_ctl_error;
 static bool autoclock = true;
 static char *quirk_alias[SNDRV_CARDS];
+static char *delayed_register[SNDRV_CARDS];
 
 bool snd_usb_use_vmalloc = true;
 bool snd_usb_skip_validation;
···
 MODULE_PARM_DESC(autoclock, "Enable auto-clock selection for UAC2 devices (default: yes).");
 module_param_array(quirk_alias, charp, NULL, 0444);
 MODULE_PARM_DESC(quirk_alias, "Quirk aliases, e.g. 0123abcd:5678beef.");
+module_param_array(delayed_register, charp, NULL, 0444);
+MODULE_PARM_DESC(delayed_register, "Quirk for delayed registration, given by id:iface, e.g. 0123abcd:4.");
 module_param_named(use_vmalloc, snd_usb_use_vmalloc, bool, 0444);
 MODULE_PARM_DESC(use_vmalloc, "Use vmalloc for PCM intermediate buffers (default: yes).");
 module_param_named(skip_validation, snd_usb_skip_validation, bool, 0444);
···
 	return false;
 }
 
+static bool check_delayed_register_option(struct snd_usb_audio *chip, int iface)
+{
+	int i;
+	unsigned int id, inum;
+
+	for (i = 0; i < ARRAY_SIZE(delayed_register); i++) {
+		if (delayed_register[i] &&
+		    sscanf(delayed_register[i], "%x:%x", &id, &inum) == 2 &&
+		    id == chip->usb_id)
+			return inum != iface;
+	}
+
+	return false;
+}
+
 static const struct usb_device_id usb_audio_ids[]; /* defined below */
 
 /* look for the corresponding quirk */
···
 			goto __error;
 	}
 
-	/* we are allowed to call snd_card_register() many times */
-	err = snd_card_register(chip->card);
-	if (err < 0)
-		goto __error;
+	if (chip->need_delayed_register) {
+		dev_info(&dev->dev,
+			 "Found post-registration device assignment: %08x:%02x\n",
+			 chip->usb_id, ifnum);
+		chip->need_delayed_register = false; /* clear again */
+	}
+
+	/* we are allowed to call snd_card_register() many times, but first
+	 * check to see if a device needs to skip it or do anything special
+	 */
+	if (!snd_usb_registration_quirk(chip, ifnum) &&
+	    !check_delayed_register_option(chip, ifnum)) {
+		err = snd_card_register(chip->card);
+		if (err < 0)
+			goto __error;
+	}
 
 	if (quirk && quirk->shares_media_device) {
 		/* don't want to fail when snd_media_device_create() fails */

sound/usb/clock.c

···
 	return ret;
 }
 
-/*
- * Assume the clock is valid if clock source supports only one single sample
- * rate, the terminal is connected directly to it (there is no clock selector)
- * and clock type is internal. This is to deal with some Denon DJ controllers
- * that always reports that clock is invalid.
- */
 static bool uac_clock_source_is_valid_quirk(struct snd_usb_audio *chip,
 					    struct audioformat *fmt,
 					    int source_id)
 {
+	bool ret = false;
+	int count;
+	unsigned char data;
+	struct usb_device *dev = chip->dev;
+
 	if (fmt->protocol == UAC_VERSION_2) {
 		struct uac_clock_source_descriptor *cs_desc =
 			snd_usb_find_clock_source(chip->ctrl_intf, source_id);
···
 		if (!cs_desc)
 			return false;
 
-		return (fmt->nr_rates == 1 &&
-			(fmt->clock & 0xff) == cs_desc->bClockID &&
-			(cs_desc->bmAttributes & 0x3) !=
-				UAC_CLOCK_SOURCE_TYPE_EXT);
+		/*
+		 * Assume the clock is valid if clock source supports only one
+		 * single sample rate, the terminal is connected directly to it
+		 * (there is no clock selector) and clock type is internal.
+		 * This is to deal with some Denon DJ controllers that always
+		 * reports that clock is invalid.
+		 */
+		if (fmt->nr_rates == 1 &&
+		    (fmt->clock & 0xff) == cs_desc->bClockID &&
+		    (cs_desc->bmAttributes & 0x3) !=
+				UAC_CLOCK_SOURCE_TYPE_EXT)
+			return true;
 	}
 
-	return false;
+	/*
+	 * MOTU MicroBook IIc
+	 * Sample rate changes takes more than 2 seconds for this device. Clock
+	 * validity request returns false during that period.
+	 */
+	if (chip->usb_id == USB_ID(0x07fd, 0x0004)) {
+		count = 0;
+
+		while ((!ret) && (count < 50)) {
+			int err;
+
+			msleep(100);
+
+			err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR,
+					      USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN,
+					      UAC2_CS_CONTROL_CLOCK_VALID << 8,
+					      snd_usb_ctrl_intf(chip) | (source_id << 8),
+					      &data, sizeof(data));
+			if (err < 0) {
+				dev_warn(&dev->dev,
+					 "%s(): cannot get clock validity for id %d\n",
+					   __func__, source_id);
+				return false;
+			}
+
+			ret = !!data;
+			count++;
+		}
+	}
+
+	return ret;
 }
 
 static bool uac_clock_source_is_valid(struct snd_usb_audio *chip,

sound/usb/format.c

···
 	return 0;
 }
 
+
+/*
+ * Presonus Studio 1810c supports a limited set of sampling
+ * rates per altsetting but reports the full set each time.
+ * If we don't filter out the unsupported rates and attempt
+ * to configure the card, it will hang refusing to do any
+ * further audio I/O until a hard reset is performed.
+ *
+ * The list of supported rates per altsetting (set of available
+ * I/O channels) is described in the owner's manual, section 2.2.
+ */
+static bool s1810c_valid_sample_rate(struct audioformat *fp,
+				     unsigned int rate)
+{
+	switch (fp->altsetting) {
+	case 1:
+		/* All ADAT ports available */
+		return rate <= 48000;
+	case 2:
+		/* Half of ADAT ports available */
+		return (rate == 88200 || rate == 96000);
+	case 3:
+		/* Analog I/O only (no S/PDIF nor ADAT) */
+		return rate >= 176400;
+	default:
+		return false;
+	}
+	return false;
+}
+
 /*
  * Helper function to walk the array of sample rate triplets reported by
  * the device. The problem is that we need to parse whole array first to
···
 		}
 
 		for (rate = min; rate <= max; rate += res) {
+
+			/* Filter out invalid rates on Presonus Studio 1810c */
+			if (chip->usb_id == USB_ID(0x0194f, 0x010c) &&
+			    !s1810c_valid_sample_rate(fp, rate))
+				goto skip_rate;
+
 			if (fp->rate_table)
 				fp->rate_table[nr_rates] = rate;
 			if (!fp->rate_min || rate < fp->rate_min)
···
 				break;
 			}
 
+skip_rate:
 			/* avoid endless loop */
 			if (res == 0)
 				break;

sound/usb/midi.c

···
 	__u8  bDescriptorType;
 	__u8  bDescriptorSubtype;
 	__u8  bNumEmbMIDIJack;
-	__u8  baAssocJackID[0];
+	__u8  baAssocJackID[];
 } __attribute__ ((packed));
 
 struct snd_usb_midi_in_endpoint;
···
 	return 0;
 }
 
+static struct usb_ms_endpoint_descriptor *find_usb_ms_endpoint_descriptor(
+					struct usb_host_endpoint *hostep)
+{
+	unsigned char *extra = hostep->extra;
+	int extralen = hostep->extralen;
+
+	while (extralen > 3) {
+		struct usb_ms_endpoint_descriptor *ms_ep =
+				(struct usb_ms_endpoint_descriptor *)extra;
+
+		if (ms_ep->bLength > 3 &&
+		    ms_ep->bDescriptorType == USB_DT_CS_ENDPOINT &&
+		    ms_ep->bDescriptorSubtype == UAC_MS_GENERAL)
+			return ms_ep;
+		if (!extra[0])
+			break;
+		extralen -= extra[0];
+		extra += extra[0];
+	}
+	return NULL;
+}
+
 /*
  * Returns MIDIStreaming device capabilities.
  */
···
 		ep = get_ep_desc(hostep);
 		if (!usb_endpoint_xfer_bulk(ep) && !usb_endpoint_xfer_int(ep))
 			continue;
-		ms_ep = (struct usb_ms_endpoint_descriptor *)hostep->extra;
-		if (hostep->extralen < 4 ||
-		    ms_ep->bLength < 4 ||
-		    ms_ep->bDescriptorType != USB_DT_CS_ENDPOINT ||
-		    ms_ep->bDescriptorSubtype != UAC_MS_GENERAL)
+		ms_ep = find_usb_ms_endpoint_descriptor(hostep);
+		if (!ms_ep)
 			continue;
 		if (usb_endpoint_dir_out(ep)) {
 			if (endpoints[epidx].out_ep) {

sound/usb/mixer.c

···
  * retrieve a mixer value
  */
 
+static inline int mixer_ctrl_intf(struct usb_mixer_interface *mixer)
+{
+	return get_iface_desc(mixer->hostif)->bInterfaceNumber;
+}
+
 static int get_ctl_value_v1(struct usb_mixer_elem_info *cval, int request,
 			    int validx, int *value_ret)
 {
···
 		return -EIO;
 
 	while (timeout-- > 0) {
-		idx = snd_usb_ctrl_intf(chip) | (cval->head.id << 8);
+		idx = mixer_ctrl_intf(cval->head.mixer) | (cval->head.id << 8);
 		err = snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), request,
 				      USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN,
 				      validx, idx, buf, val_len);
···
 	if (ret)
 		goto error;
 
-	idx = snd_usb_ctrl_intf(chip) | (cval->head.id << 8);
+	idx = mixer_ctrl_intf(cval->head.mixer) | (cval->head.id << 8);
 	ret = snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), bRequest,
 			      USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN,
 			      validx, idx, buf, size);
···
 		return -EIO;
 
 	while (timeout-- > 0) {
-		idx = snd_usb_ctrl_intf(chip) | (cval->head.id << 8);
+		idx = mixer_ctrl_intf(cval->head.mixer) | (cval->head.id << 8);
 		err = snd_usb_ctl_msg(chip->dev,
 				      usb_sndctrlpipe(chip->dev, 0), request,
 				      USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT,
···
 				  struct usb_audio_term *term,
 				  void *p1, int id)
 {
+	struct uac2_effect_unit_descriptor *d = p1;
+	int err;
+
+	err = __check_input_term(state, d->bSourceID, term);
+	if (err < 0)
+		return err;
 	term->type = UAC3_EFFECT_UNIT << 16; /* virtual type */
 	term->id = id;
 	return 0;
···
 		    get_ctl_value(cval, UAC_GET_MIN, (cval->control << 8) | minchn, &cval->min) < 0) {
 			usb_audio_err(cval->head.mixer->chip,
 				      "%d:%d: cannot get min/max values for control %d (id %d)\n",
-				   cval->head.id, snd_usb_ctrl_intf(cval->head.mixer->chip),
+				   cval->head.id, mixer_ctrl_intf(cval->head.mixer),
 							       cval->control, cval->head.id);
 			return -EINVAL;
 		}
···
 	if (ret)
 		goto error;
 
-	idx = snd_usb_ctrl_intf(chip) | (cval->head.id << 8);
+	idx = mixer_ctrl_intf(cval->head.mixer) | (cval->head.id << 8);
 	if (cval->head.mixer->protocol == UAC_VERSION_2) {
 		struct uac2_connectors_ctl_blk uac2_conn;
 
···
 
 	/* get min/max values */
 	get_min_max_with_quirks(cval, 0, kctl);
+
+	/* skip a bogus volume range */
+	if (cval->max <= cval->min) {
+		usb_audio_dbg(mixer->chip,
+			      "[%d] FU [%s] skipped due to invalid volume\n",
+			      cval->head.id, kctl->id.name);
+		snd_ctl_free_one(kctl);
+		return;
+	}
+
 
 	if (control == UAC_FU_VOLUME) {
 		check_mapped_dB(map, cval);
···
 	list_for_each_entry(mixer, &chip->mixer_list, list) {
 		snd_iprintf(buffer,
 			"USB Mixer: usb_id=0x%08x, ctrlif=%i, ctlerr=%i\n",
-				chip->usb_id, snd_usb_ctrl_intf(chip),
+				chip->usb_id, mixer_ctrl_intf(mixer),
 				mixer->ignore_ctl_error);
 		snd_iprintf(buffer, "Card: %s\n", chip->card->longname);
 		for (unitid = 0; unitid < MAX_ID_ELEMS; unitid++) {

sound/usb/mixer_quirks.c

···
 #include "mixer_scarlett.h"
 #include "mixer_scarlett_gen2.h"
 #include "mixer_us16x08.h"
+#include "mixer_s1810c.h"
 #include "helper.h"
 
 struct std_mono_table {
···
 	case USB_ID(0x2a39, 0x3fd3): /* RME ADI-2 DAC */
 	case USB_ID(0x2a39, 0x3fd4): /* RME */
 		err = snd_rme_controls_create(mixer);
+		break;
+
+	case USB_ID(0x0194f, 0x010c): /* Presonus Studio 1810c */
+		err = snd_sc1810_init_mixer(mixer);
 		break;
 	}
 

sound/usb/mixer_s1810c.c

···
+// SPDX-License-Identifier: GPL-2.0
+/*
+ * Presonus Studio 1810c driver for ALSA
+ * Copyright (C) 2019 Nick Kossifidis <[e-mail]>
+ *
+ * Based on reverse engineering of the communication protocol
+ * between the windows driver / Univeral Control (UC) program
+ * and the device, through usbmon.
+ *
+ * For now this bypasses the mixer, with all channels split,
+ * so that the software can mix with greater flexibility.
+ * It also adds controls for the 4 buttons on the front of
+ * the device.
+ */
+
+#include <linux/usb.h>
+#include <linux/usb/audio-v2.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/control.h>
+
+#include "usbaudio.h"
+#include "mixer.h"
+#include "mixer_quirks.h"
+#include "helper.h"
+#include "mixer_s1810c.h"
+
+#define SC1810C_CMD_REQ	160
+#define SC1810C_CMD_REQTYPE \
+	(USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_OUT)
+#define SC1810C_CMD_F1		0x50617269
+#define SC1810C_CMD_F2		0x14
+
+/*
+ * DISCLAIMER: These are just guesses based on the
+ * dumps I got.
+ *
+ * It seems like a selects between
+ * device (0), mixer (0x64) and output (0x65)
+ *
+ * For mixer (0x64):
+ *  * b selects an input channel (see below).
+ *  * c selects an output channel pair (see below).
+ *  * d selects left (0) or right (1) of that pair.
+ *  * e 0-> disconnect, 0x01000000-> connect,
+ *	0x0109-> used for stereo-linking channels,
+ *	e is also used for setting volume levels
+ *	in which case b is also set so I guess
+ *	this way it is possible to set the volume
+ *	level from the specified input to the
+ *	specified output.
+ *
+ * IN Channels:
+ * 0  - 7  Mic/Inst/Line (Analog inputs)
+ * 8  - 9  S/PDIF
+ * 10 - 17 ADAT
+ * 18 - 35 DAW (Inputs from the host)
+ *
+ * OUT Channels (pairs):
+ * 0 -> Main out
+ * 1 -> Line1/2
+ * 2 -> Line3/4
+ * 3 -> S/PDIF
+ * 4 -> ADAT?
+ *
+ * For device (0):
+ *  * b and c are not used, at least not on the
+ *    dumps I got.
+ *  * d sets the control id to be modified
+ *    (see below).
+ *  * e sets the setting for that control.
+ *    (so for the switches I was interested
+ *    in it's 0/1)
+ *
+ * For output (0x65):
+ *   * b is the output channel (see above).
+ *   * c is zero.
+ *   * e I guess the same as with mixer except 0x0109
+ *	 which I didn't see in my dumps.
+ *
+ * The two fixed fields have the same values for
+ * mixer and output but a different set for device.
+ */
+struct s1810c_ctl_packet {
+	u32 a;
+	u32 b;
+	u32 fixed1;
+	u32 fixed2;
+	u32 c;
+	u32 d;
+	u32 e;
+};
+
+#define SC1810C_CTL_LINE_SW	0
+#define SC1810C_CTL_MUTE_SW	1
+#define SC1810C_CTL_AB_SW	3
+#define SC1810C_CTL_48V_SW	4
+
+#define SC1810C_SET_STATE_REQ	161
+#define SC1810C_SET_STATE_REQTYPE SC1810C_CMD_REQTYPE
+#define SC1810C_SET_STATE_F1	0x64656D73
+#define SC1810C_SET_STATE_F2	0xF4
+
+#define SC1810C_GET_STATE_REQ	162
+#define SC1810C_GET_STATE_REQTYPE \
+	(USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_IN)
+#define SC1810C_GET_STATE_F1	SC1810C_SET_STATE_F1
+#define SC1810C_GET_STATE_F2	SC1810C_SET_STATE_F2
+
+#define SC1810C_STATE_F1_IDX	2
+#define SC1810C_STATE_F2_IDX	3
+
+/*
+ * This packet includes mixer volumes and
+ * various other fields, it's an extended
+ * version of ctl_packet, with a and b
+ * being zero and different f1/f2.
+ */
+struct s1810c_state_packet {
+	u32 fields[63];
+};
+
+#define SC1810C_STATE_48V_SW	58
+#define SC1810C_STATE_LINE_SW	59
+#define SC1810C_STATE_MUTE_SW	60
+#define SC1810C_STATE_AB_SW	62
+
+struct s1810_mixer_state {
+	uint16_t seqnum;
+	struct mutex usb_mutex;
+	struct mutex data_mutex;
+};
+
+static int
+snd_s1810c_send_ctl_packet(struct usb_device *dev, u32 a,
+			   u32 b, u32 c, u32 d, u32 e)
+{
+	struct s1810c_ctl_packet pkt = { 0 };
+	int ret = 0;
+
+	pkt.fixed1 = SC1810C_CMD_F1;
+	pkt.fixed2 = SC1810C_CMD_F2;
+
+	pkt.a = a;
+	pkt.b = b;
+	pkt.c = c;
+	pkt.d = d;
+	/*
+	 * Value for settings 0/1 for this
+	 * output channel is always 0 (probably because
+	 * there is no ADAT output on 1810c)
+	 */
+	pkt.e = (c == 4) ? 0 : e;
+
+	ret = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0),
+			      SC1810C_CMD_REQ,
+			      SC1810C_CMD_REQTYPE, 0, 0, &pkt, sizeof(pkt));
+	if (ret < 0) {
+		dev_warn(&dev->dev, "could not send ctl packet\n");
+		return ret;
+	}
+	return 0;
+}
+
+/*
+ * When opening Universal Control the program periodicaly
+ * sends and receives state packets for syncinc state between
+ * the device and the host.
+ *
+ * Note that if we send only the request to get data back we'll
+ * get an error, we need to first send an empty state packet and
+ * then ask to receive a filled. Their seqnumbers must also match.
+ */
+static int
+snd_sc1810c_get_status_field(struct usb_device *dev,
+			     u32 *field, int field_idx, uint16_t *seqnum)
+{
+	struct s1810c_state_packet pkt_out = { { 0 } };
+	struct s1810c_state_packet pkt_in = { { 0 } };
+	int ret = 0;
+
+	pkt_out.fields[SC1810C_STATE_F1_IDX] = SC1810C_SET_STATE_F1;
+	pkt_out.fields[SC1810C_STATE_F2_IDX] = SC1810C_SET_STATE_F2;
+	ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0),
+			      SC1810C_SET_STATE_REQ,
+			      SC1810C_SET_STATE_REQTYPE,
+			      (*seqnum), 0, &pkt_out, sizeof(pkt_out));
+	if (ret < 0) {
+		dev_warn(&dev->dev, "could not send state packet (%d)\n", ret);
+		return ret;
+	}
+
+	ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0),
+			      SC1810C_GET_STATE_REQ,
+			      SC1810C_GET_STATE_REQTYPE,
+			      (*seqnum), 0, &pkt_in, sizeof(pkt_in));
+	if (ret < 0) {
+		dev_warn(&dev->dev, "could not get state field %u (%d)\n",
+			 field_idx, ret);
+		return ret;
+	}
+
+	(*field) = pkt_in.fields[field_idx];
+	(*seqnum)++;
+	return 0;
+}
+
+/*
+ * This is what I got when bypassing the mixer with
+ * all channels split. I'm not 100% sure of what's going
+ * on, I could probably clean this up based on my observations
+ * but I prefer to keep the same behavior as the windows driver.
+ */
+static int snd_s1810c_init_mixer_maps(struct snd_usb_audio *chip)
+{
+	u32 a, b, c, e, n, off;
+	struct usb_device *dev = chip->dev;
+
+	/* Set initial volume levels ? */
+	a = 0x64;
+	e = 0xbc;
+	for (n = 0; n < 2; n++) {
+		off = n * 18;
+		for (b = off, c = 0; b < 18 + off; b++) {
+			/* This channel to all outputs ? */
+			for (c = 0; c <= 8; c++) {
+				snd_s1810c_send_ctl_packet(dev, a, b, c, 0, e);
+				snd_s1810c_send_ctl_packet(dev, a, b, c, 1, e);
+			}
+			/* This channel to main output (again) */
+			snd_s1810c_send_ctl_packet(dev, a, b, 0, 0, e);
+			snd_s1810c_send_ctl_packet(dev, a, b, 0, 1, e);
+		}
+		/*
+		 * I noticed on UC that DAW channels have different
+		 * initial volumes, so this makes sense.
+		 */
+		e = 0xb53bf0;
+	}
+
+	/* Connect analog outputs ? */
+	a = 0x65;
+	e = 0x01000000;
+	for (b = 1; b < 3; b++) {
+		snd_s1810c_send_ctl_packet(dev, a, b, 0, 0, e);
+		snd_s1810c_send_ctl_packet(dev, a, b, 0, 1, e);
+	}
+	snd_s1810c_send_ctl_packet(dev, a, 0, 0, 0, e);
+	snd_s1810c_send_ctl_packet(dev, a, 0, 0, 1, e);
+
+	/* Set initial volume levels for S/PDIF mappings ? */
+	a = 0x64;
+	e = 0xbc;
+	c = 3;
+	for (n = 0; n < 2; n++) {
+		off = n * 18;
+		for (b = off; b < 18 + off; b++) {
+			snd_s1810c_send_ctl_packet(dev, a, b, c, 0, e);
+			snd_s1810c_send_ctl_packet(dev, a, b, c, 1, e);
+		}
+		e = 0xb53bf0;
+	}
+
+	/* Connect S/PDIF output ? */
+	a = 0x65;
+	e = 0x01000000;
+	snd_s1810c_send_ctl_packet(dev, a, 3, 0, 0, e);
+	snd_s1810c_send_ctl_packet(dev, a, 3, 0, 1, e);
+
+	/* Connect all outputs (again) ? */
+	a = 0x65;
+	e = 0x01000000;
+	for (b = 0; b < 4; b++) {
+		snd_s1810c_send_ctl_packet(dev, a, b, 0, 0, e);
+		snd_s1810c_send_ctl_packet(dev, a, b, 0, 1, e);
+	}
+
+	/* Basic routing to get sound out of the device */
+	a = 0x64;
+	e = 0x01000000;
+	for (c = 0; c < 4; c++) {
+		for (b = 0; b < 36; b++) {
+			if ((c == 0 && b == 18) ||	/* DAW1/2 -> Main */
+			    (c == 1 && b == 20) ||	/* DAW3/4 -> Line3/4 */
+			    (c == 2 && b == 22) ||	/* DAW4/5 -> Line5/6 */
+			    (c == 3 && b == 24)) {	/* DAW5/6 -> S/PDIF */
+				/* Left */
+				snd_s1810c_send_ctl_packet(dev, a, b, c, 0, e);
+				snd_s1810c_send_ctl_packet(dev, a, b, c, 1, 0);
+				b++;
+				/* Right */
+				snd_s1810c_send_ctl_packet(dev, a, b, c, 0, 0);
+				snd_s1810c_send_ctl_packet(dev, a, b, c, 1, e);
+			} else {
+				/* Leave the rest disconnected */
+				snd_s1810c_send_ctl_packet(dev, a, b, c, 0, 0);
+				snd_s1810c_send_ctl_packet(dev, a, b, c, 1, 0);
+			}
+		}
+	}
+
+	/* Set initial volume levels for S/PDIF (again) ? */
+	a = 0x64;
+	e = 0xbc;
+	c = 3;
+	for (n = 0; n < 2; n++) {
+		off = n * 18;
+		for (b = off; b < 18 + off; b++) {
+			snd_s1810c_send_ctl_packet(dev, a, b, c, 0, e);
+			snd_s1810c_send_ctl_packet(dev, a, b, c, 1, e);
+		}
+		e = 0xb53bf0;
+	}
+
+	/* Connect S/PDIF outputs (again) ? */
+	a = 0x65;
+	e = 0x01000000;
+	snd_s1810c_send_ctl_packet(dev, a, 3, 0, 0, e);
+	snd_s1810c_send_ctl_packet(dev, a, 3, 0, 1, e);
+
+	/* Again ? */
+	snd_s1810c_send_ctl_packet(dev, a, 3, 0, 0, e);
+	snd_s1810c_send_ctl_packet(dev, a, 3, 0, 1, e);
+
+	return 0;
+}
+
+/*
+ * Sync state with the device and retrieve the requested field,
+ * whose index is specified in (kctl->private_value & 0xFF),
+ * from the received fields array.
+ */
+static int
+snd_s1810c_get_switch_state(struct usb_mixer_interface *mixer,
+			    struct snd_kcontrol *kctl, u32 *state)
+{
+	struct snd_usb_audio *chip = mixer->chip;
+	struct s1810_mixer_state *private = mixer->private_data;
+	u32 field = 0;
+	u32 ctl_idx = (u32) (kctl->private_value & 0xFF);
+	int ret = 0;
+
+	mutex_lock(&private->usb_mutex);
+	ret = snd_sc1810c_get_status_field(chip->dev, &field,
+					   ctl_idx, &private->seqnum);
+	if (ret < 0)
+		goto unlock;
+
+	*state = field;
+ unlock:
+	mutex_unlock(&private->usb_mutex);
+	return ret ? ret : 0;
+}
+
+/*
+ * Send a control packet to the device for the control id
+ * specified in (kctl->private_value >> 8) with value
+ * specified in (kctl->private_value >> 16).
+ */
+static int
+snd_s1810c_set_switch_state(struct usb_mixer_interface *mixer,
+			    struct snd_kcontrol *kctl)
+{
+	struct snd_usb_audio *chip = mixer->chip;
+	struct s1810_mixer_state *private = mixer->private_data;
+	u32 pval = (u32) kctl->private_value;
+	u32 ctl_id = (pval >> 8) & 0xFF;
+	u32 ctl_val = (pval >> 16) & 0x1;
+	int ret = 0;
+
+	mutex_lock(&private->usb_mutex);
+	ret = snd_s1810c_send_ctl_packet(chip->dev, 0, 0, 0, ctl_id, ctl_val);
+	mutex_unlock(&private->usb_mutex);
+	return ret;
+}
+
+/* Generic get/set/init functions for switch controls */
+
+static int
+snd_s1810c_switch_get(struct snd_kcontrol *kctl,
+		      struct snd_ctl_elem_value *ctl_elem)
+{
+	struct usb_mixer_elem_list *list = snd_kcontrol_chip(kctl);
+	struct usb_mixer_interface *mixer = list->mixer;
+	struct s1810_mixer_state *private = mixer->private_data;
+	u32 pval = (u32) kctl->private_value;
+	u32 ctl_idx = pval & 0xFF;
+	u32 state = 0;
+	int ret = 0;
+
+	mutex_lock(&private->data_mutex);
+	ret = snd_s1810c_get_switch_state(mixer, kctl, &state);
+	if (ret < 0)
+		goto unlock;
+
+	switch (ctl_idx) {
+	case SC1810C_STATE_LINE_SW:
+	case SC1810C_STATE_AB_SW:
+		ctl_elem->value.enumerated.item[0] = (int)state;
+		break;
+	default:
+		ctl_elem->value.integer.value[0] = (long)state;
+	}
+
+ unlock:
+	mutex_unlock(&private->data_mutex);
+	return (ret < 0) ? ret : 0;
+}
+
+static int
+snd_s1810c_switch_set(struct snd_kcontrol *kctl,
+		      struct snd_ctl_elem_value *ctl_elem)
+{
+	struct usb_mixer_elem_list *list = snd_kcontrol_chip(kctl);
+	struct usb_mixer_interface *mixer = list->mixer;
+	struct s1810_mixer_state *private = mixer->private_data;
+	u32 pval = (u32) kctl->private_value;
+	u32 ctl_idx = pval & 0xFF;
+	u32 curval = 0;
+	u32 newval = 0;
+	int ret = 0;
+
+	mutex_lock(&private->data_mutex);
+	ret = snd_s1810c_get_switch_state(mixer, kctl, &curval);
+	if (ret < 0)
+		goto unlock;
+
+	switch (ctl_idx) {
+	case SC1810C_STATE_LINE_SW:
+	case SC1810C_STATE_AB_SW:
+		newval = (u32) ctl_elem->value.enumerated.item[0];
+		break;
+	default:
+		newval = (u32) ctl_elem->value.integer.value[0];
+	}
+
+	if (curval == newval)
+		goto unlock;
+
+	kctl->private_value &= ~(0x1 << 16);
+	kctl->private_value |= (unsigned int)(newval & 0x1) << 16;
+	ret = snd_s1810c_set_switch_state(mixer, kctl);
+
+ unlock:
+	mutex_unlock(&private->data_mutex);
+	return (ret < 0) ? 0 : 1;
+}
+
+static int
+snd_s1810c_switch_init(struct usb_mixer_interface *mixer,
+		       const struct snd_kcontrol_new *new_kctl)
+{
+	struct snd_kcontrol *kctl;
+	struct usb_mixer_elem_info *elem;
+
+	elem = kzalloc(sizeof(struct usb_mixer_elem_info), GFP_KERNEL);
+	if (!elem)
+		return -ENOMEM;
+
+	elem->head.mixer = mixer;
+	elem->control = 0;
+	elem->head.id = 0;
+	elem->channels = 1;
+
+	kctl = snd_ctl_new1(new_kctl, elem);
+	if (!kctl) {
+		kfree(elem);
+		return -ENOMEM;
+	}
+	kctl->private_free = snd_usb_mixer_elem_free;
+
+	return snd_usb_mixer_add_control(&elem->head, kctl);
+}
+
+static int
+snd_s1810c_line_sw_info(struct snd_kcontrol *kctl,
+			struct snd_ctl_elem_info *uinfo)
+{
+	static const char *const texts[2] = {
+		"Preamp On (Mic/Inst)",
+		"Preamp Off (Line in)"
+	};
+
+	return snd_ctl_enum_info(uinfo, 1, ARRAY_SIZE(texts), texts);
+}
+
+static const struct snd_kcontrol_new snd_s1810c_line_sw = {
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name = "Line 1/2 Source Type",
+	.info = snd_s1810c_line_sw_info,
+	.get = snd_s1810c_switch_get,
+	.put = snd_s1810c_switch_set,
+	.private_value = (SC1810C_STATE_LINE_SW | SC1810C_CTL_LINE_SW << 8)
+};
+
+static const struct snd_kcontrol_new snd_s1810c_mute_sw = {
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name = "Mute Main Out Switch",
+	.info = snd_ctl_boolean_mono_info,
+	.get = snd_s1810c_switch_get,
+	.put = snd_s1810c_switch_set,
+	.private_value = (SC1810C_STATE_MUTE_SW | SC1810C_CTL_MUTE_SW << 8)
+};
+
+static const struct snd_kcontrol_new snd_s1810c_48v_sw = {
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name = "48V Phantom Power On Mic Inputs Switch",
+	.info = snd_ctl_boolean_mono_info,
+	.get = snd_s1810c_switch_get,
+	.put = snd_s1810c_switch_set,
+	.private_value = (SC1810C_STATE_48V_SW | SC1810C_CTL_48V_SW << 8)
+};
+
+static int
+snd_s1810c_ab_sw_info(struct snd_kcontrol *kctl,
+		      struct snd_ctl_elem_info *uinfo)
+{
+	static const char *const texts[2] = {
+		"1/2",
+		"3/4"
+	};
+
+	return snd_ctl_enum_info(uinfo, 1, ARRAY_SIZE(texts), texts);
+}
+
+static const struct snd_kcontrol_new snd_s1810c_ab_sw = {
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name = "Headphone 1 Source Route",
+	.info = snd_s1810c_ab_sw_info,
+	.get = snd_s1810c_switch_get,
+	.put = snd_s1810c_switch_set,
+	.private_value = (SC1810C_STATE_AB_SW | SC1810C_CTL_AB_SW << 8)
+};
+
+static void snd_sc1810_mixer_state_free(struct usb_mixer_interface *mixer)
+{
+	struct s1810_mixer_state *private = mixer->private_data;
+	kfree(private);
+	mixer->private_data = NULL;
+}
+
+/* Entry point, called from mixer_quirks.c */
+int snd_sc1810_init_mixer(struct usb_mixer_interface *mixer)
+{
+	struct s1810_mixer_state *private = NULL;
+	struct snd_usb_audio *chip = mixer->chip;
+	struct usb_device *dev = chip->dev;
+	int ret = 0;
+
+	/* Run this only once */
+	if (!list_empty(&chip->mixer_list))
+		return 0;
+
+	dev_info(&dev->dev,
+		 "Presonus Studio 1810c, device_setup: %u\n", chip->setup);
+	if (chip->setup == 1)
+		dev_info(&dev->dev, "(8out/18in @ 48KHz)\n");
+	else if (chip->setup == 2)
+		dev_info(&dev->dev, "(6out/8in @ 192KHz)\n");
+	else
+		dev_info(&dev->dev, "(8out/14in @ 96KHz)\n");
+
+	ret = snd_s1810c_init_mixer_maps(chip);
+	if (ret < 0)
+		return ret;
+
+	private = kzalloc(sizeof(struct s1810_mixer_state), GFP_KERNEL);
+	if (!private)
+		return -ENOMEM;
+
+	mutex_init(&private->usb_mutex);
+	mutex_init(&private->data_mutex);
+
+	mixer->private_data = private;
+	mixer->private_free = snd_sc1810_mixer_state_free;
+
+	private->seqnum = 1;
+
+	ret = snd_s1810c_switch_init(mixer, &snd_s1810c_line_sw);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_s1810c_switch_init(mixer, &snd_s1810c_mute_sw);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_s1810c_switch_init(mixer, &snd_s1810c_48v_sw);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_s1810c_switch_init(mixer, &snd_s1810c_ab_sw);
+	if (ret < 0)
+		return ret;
+	return ret;
+}

sound/usb/mixer_s1810c.h

···
+/* SPDX-License-Identifier: GPL-2.0 */
+/*
+ * Presonus Studio 1810c driver for ALSA
+ * Copyright (C) 2019 Nick Kossifidis <[e-mail]>
+ */
+
+int snd_sc1810_init_mixer(struct usb_mixer_interface *mixer);

sound/usb/pcm.c

···
 		ep = 0x81;
 		ifnum = 1;
 		goto add_sync_ep_from_ifnum;
-	case USB_ID(0x07fd, 0x0004): /* MOTU MicroBook II */
+	case USB_ID(0x07fd, 0x0004): /* MOTU MicroBook II/IIc */
+		/* MicroBook IIc */
+		if (altsd->bInterfaceClass == USB_CLASS_AUDIO)
+			return 0;
+
+		/* MicroBook II */
 		ep = 0x84;
 		ifnum = 0;
 		goto add_sync_ep_from_ifnum;

sound/usb/proc.c

···
 		snd_iprintf(buffer, "  Interface %d\n", fp->iface);
 		snd_iprintf(buffer, "    Altset %d\n", fp->altsetting);
 		snd_iprintf(buffer, "    Format:");
-		for (fmt = 0; fmt <= SNDRV_PCM_FORMAT_LAST; ++fmt)
+		pcm_for_each_format(fmt)
 			if (fp->formats & pcm_format_to_bits(fmt))
 				snd_iprintf(buffer, " %s",
 					    snd_pcm_format_name(fmt));

sound/usb/quirks-table.h

···
 },
 /* MOTU Microbook II */
 {
-	USB_DEVICE(0x07fd, 0x0004),
+	USB_DEVICE_VENDOR_SPEC(0x07fd, 0x0004),
 	.driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
 		.vendor_name = "MOTU",
 		.product_name = "MicroBookII",

sound/usb/quirks.c

···
 	return 0; /* keep this altsetting */
 }
 
+static int s1810c_skip_setting_quirk(struct snd_usb_audio *chip,
+					   int iface, int altno)
+{
+	/*
+	 * Altno settings:
+	 *
+	 * Playback (Interface 1):
+	 * 1: 6 Analog + 2 S/PDIF
+	 * 2: 6 Analog + 2 S/PDIF
+	 * 3: 6 Analog
+	 *
+	 * Capture (Interface 2):
+	 * 1: 8 Analog + 2 S/PDIF + 8 ADAT
+	 * 2: 8 Analog + 2 S/PDIF + 4 ADAT
+	 * 3: 8 Analog
+	 */
+
+	/*
+	 * I'll leave 2 as the default one and
+	 * use device_setup to switch to the
+	 * other two.
+	 */
+	if ((chip->setup == 0 || chip->setup > 2) && altno != 2)
+		return 1;
+	else if (chip->setup == 1 && altno != 1)
+		return 1;
+	else if (chip->setup == 2 && altno != 3)
+		return 1;
+
+	return 0;
+}
+
 int snd_usb_apply_interface_quirk(struct snd_usb_audio *chip,
 				  int iface,
 				  int altno)
···
 	/* fasttrackpro usb: skip altsets incompatible with device_setup */
 	if (chip->usb_id == USB_ID(0x0763, 0x2012))
 		return fasttrackpro_skip_setting_quirk(chip, iface, altno);
+	/* presonus studio 1810c: skip altsets incompatible with device_setup */
+	if (chip->usb_id == USB_ID(0x0194f, 0x010c))
+		return s1810c_skip_setting_quirk(chip, iface, altno);
+
 
 	return 0;
 }
···
 	case USB_ID(0x2466, 0x8010): /* Fractal Audio Axe-Fx 3 */
 		return snd_usb_axefx3_boot_quirk(dev);
 	case USB_ID(0x07fd, 0x0004): /* MOTU MicroBook II */
-		return snd_usb_motu_microbookii_boot_quirk(dev);
+		/*
+		 * For some reason interface 3 with vendor-spec class is
+		 * detected on MicroBook IIc.
+		 */
+		if (get_iface_desc(intf->altsetting)->bInterfaceClass ==
+		    USB_CLASS_VENDOR_SPEC &&
+		    get_iface_desc(intf->altsetting)->bInterfaceNumber < 3)
+			return snd_usb_motu_microbookii_boot_quirk(dev);
+		break;
 	}
 
 	return 0;
···
 		else
 			fp->ep_attr |= USB_ENDPOINT_SYNC_SYNC;
 		break;
+	case USB_ID(0x07fd, 0x0004):  /* MOTU MicroBook IIc */
+		/*
+		 * MaxPacketsOnly attribute is erroneously set in endpoint
+		 * descriptors. As a result this card produces noise with
+		 * all sample rates other than 96 KHz.
+		 */
+		fp->attributes &= ~UAC_EP_CS_ATTR_FILL_MAX;
+		break;
 	}
+}
+
+/*
+ * registration quirk:
+ * the registration is skipped if a device matches with the given ID,
+ * unless the interface reaches to the defined one.  This is for delaying
+ * the registration until the last known interface, so that the card and
+ * devices appear at the same time.
+ */
+
+struct registration_quirk {
+	unsigned int usb_id;	/* composed via USB_ID() */
+	unsigned int interface;	/* the interface to trigger register */
+};
+
+#define REG_QUIRK_ENTRY(vendor, product, iface) \
+	{ .usb_id = USB_ID(vendor, product), .interface = (iface) }
+
+static const struct registration_quirk registration_quirks[] = {
+	REG_QUIRK_ENTRY(0x0951, 0x16d8, 2),	/* Kingston HyperX AMP */
+	{ 0 }					/* terminator */
+};
+
+/* return true if skipping registration */
+bool snd_usb_registration_quirk(struct snd_usb_audio *chip, int iface)
+{
+	const struct registration_quirk *q;
+
+	for (q = registration_quirks; q->usb_id; q++)
+		if (chip->usb_id == q->usb_id)
+			return iface != q->interface;
+
+	/* Register as normal */
+	return false;
 }

sound/usb/quirks.h

···
 					  struct audioformat *fp,
 					  int stream);
 
+bool snd_usb_registration_quirk(struct snd_usb_audio *chip, int iface);
+
 #endif /* __USBAUDIO_QUIRKS_H */

sound/usb/stream.c

···
 		subs = &as->substream[stream];
 		if (subs->ep_num)
 			continue;
+		if (snd_device_get_state(chip->card, as->pcm) !=
+		    SNDRV_DEV_BUILD)
+			chip->need_delayed_register = true;
 		err = snd_pcm_new_stream(as->pcm, stream, 1);
 		if (err < 0)
 			return err;

sound/usb/usbaudio.h

···
 	unsigned int txfr_quirk:1; /* Subframe boundaries on transfers */
 	unsigned int tx_length_quirk:1; /* Put length specifier in transfers */
 	unsigned int setup_fmt_after_resume_quirk:1; /* setup the format to interface after resume */
+	unsigned int need_delayed_register:1; /* warn for delayed registration */
 	int num_interfaces;
 	int num_suspended_intf;
 	int sample_rate_read_error;

sound/usb/usx2y/usbusx2yaudio.c

···
  */
 static void usX2Y_audio_stream_free(struct snd_usX2Y_substream **usX2Y_substream)
 {
-	kfree(usX2Y_substream[SNDRV_PCM_STREAM_PLAYBACK]);
-	usX2Y_substream[SNDRV_PCM_STREAM_PLAYBACK] = NULL;
+	int stream;
 
-	kfree(usX2Y_substream[SNDRV_PCM_STREAM_CAPTURE]);
-	usX2Y_substream[SNDRV_PCM_STREAM_CAPTURE] = NULL;
+	for_each_pcm_streams(stream) {
+		kfree(usX2Y_substream[stream]);
+		usX2Y_substream[stream] = NULL;
+	}
 }
 
 static void snd_usX2Y_pcm_private_free(struct snd_pcm *pcm)