Merge branch 'asoc-5.3' into asoc-next

Documentation/devicetree/bindings/media/st,stm32-dcmi.txt

@@ -11,7 +11,7 @@ Required properties:
 - clock-names: must contain "mclk", which is the DCMI peripherial clock
 - pinctrl: the pincontrol settings to configure muxing properly
            for pins that connect to DCMI device.
-           See Documentation/devicetree/bindings/pinctrl/st,stm32-pinctrl.txt.
+           See Documentation/devicetree/bindings/pinctrl/st,stm32-pinctrl.yaml.
 - dmas: phandle to DMA controller node,
         see Documentation/devicetree/bindings/dma/stm32-dma.txt
 - dma-names: must contain "tx", which is the transmit channel from DCMI to DMA

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

+# SPDX-License-Identifier: (GPL-2.0+ OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/allwinner,sun4i-a10-i2s.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Allwinner A10 I2S Controller Device Tree Bindings
+
+maintainers:
+  - Chen-Yu Tsai <wens@csie.org>
+  - Maxime Ripard <maxime.ripard@bootlin.com>
+
+properties:
+  "#sound-dai-cells":
+    const: 0
+
+  compatible:
+    oneOf:
+      - const: allwinner,sun4i-a10-i2s
+      - const: allwinner,sun6i-a31-i2s
+      - const: allwinner,sun8i-a83t-i2s
+      - const: allwinner,sun8i-h3-i2s
+      - const: allwinner,sun50i-a64-codec-i2s
+      - items:
+          - const: allwinner,sun50i-a64-i2s
+          - const: allwinner,sun8i-h3-i2s
+
+  reg:
+    maxItems: 1
+
+  interrupts:
+    maxItems: 1
+
+  clocks:
+    items:
+      - description: Bus Clock
+      - description: Module Clock
+
+  clock-names:
+    items:
+      - const: apb
+      - const: mod
+
+  # Even though it only applies to subschemas under the conditionals,
+  # not listing them here will trigger a warning because of the
+  # additionalsProperties set to false.
+  dmas: true
+  dma-names: true
+  resets:
+    maxItems: 1
+
+allOf:
+  - if:
+      properties:
+        compatible:
+          contains:
+            enum:
+              - allwinner,sun6i-a31-i2s
+              - allwinner,sun8i-a83t-i2s
+              - allwinner,sun8i-h3-i2s
+              - allwinner,sun50i-a64-codec-i2s
+
+    then:
+      required:
+        - resets
+
+  - if:
+      properties:
+        compatible:
+          contains:
+            const: allwinner,sun8i-a83t-i2s
+
+    then:
+      properties:
+        dmas:
+          minItems: 1
+          maxItems: 2
+          items:
+            - description: RX DMA Channel
+            - description: TX DMA Channel
+          description:
+            Some controllers cannot receive but can only transmit
+            data. In such a case, the RX DMA channel is to be omitted.
+
+        dma-names:
+          oneOf:
+            - items:
+                - const: rx
+                - const: tx
+            - const: tx
+          description:
+            Some controllers cannot receive but can only transmit
+            data. In such a case, the RX name is to be omitted.
+
+    else:
+      properties:
+        dmas:
+          items:
+            - description: RX DMA Channel
+            - description: TX DMA Channel
+
+        dma-names:
+          items:
+            - const: rx
+            - const: tx
+
+required:
+  - "#sound-dai-cells"
+  - compatible
+  - reg
+  - interrupts
+  - clocks
+  - clock-names
+  - dmas
+  - dma-names
+
+additionalProperties: false
+
+examples:
+  - |
+    i2s0: i2s@1c22400 {
+        #sound-dai-cells = <0>;
+        compatible = "allwinner,sun4i-a10-i2s";
+        reg = <0x01c22400 0x400>;
+        interrupts = <0 16 4>;
+        clocks = <&apb0_gates 3>, <&i2s0_clk>;
+        clock-names = "apb", "mod";
+        dmas = <&dma 0 3>, <&dma 0 3>;
+        dma-names = "rx", "tx";
+    };
+
+...

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

+# SPDX-License-Identifier: GPL-2.0
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/allwinner,sun4i-a10-spdif.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Allwinner A10 S/PDIF Controller Device Tree Bindings
+
+maintainers:
+  - Chen-Yu Tsai <wens@csie.org>
+  - Liam Girdwood <lgirdwood@gmail.com>
+  - Mark Brown <broonie@kernel.org>
+  - Maxime Ripard <maxime.ripard@bootlin.com>
+
+properties:
+  "#sound-dai-cells":
+    const: 0
+
+  compatible:
+    oneOf:
+      - const: allwinner,sun4i-a10-spdif
+      - const: allwinner,sun6i-a31-spdif
+      - const: allwinner,sun8i-h3-spdif
+      - const: allwinner,sun50i-h6-spdif
+      - items:
+          - const: allwinner,sun8i-a83t-spdif
+          - const: allwinner,sun8i-h3-spdif
+      - items:
+          - const: allwinner,sun50i-a64-spdif
+          - const: allwinner,sun8i-h3-spdif
+
+  reg:
+    maxItems: 1
+
+  interrupts:
+    maxItems: 1
+
+  clocks:
+    items:
+      - description: Bus Clock
+      - description: Module Clock
+
+  clock-names:
+    items:
+      - const: apb
+      - const: spdif
+
+  # Even though it only applies to subschemas under the conditionals,
+  # not listing them here will trigger a warning because of the
+  # additionalsProperties set to false.
+  dmas: true
+  dma-names: true
+  resets:
+    maxItems: 1
+
+allOf:
+  - if:
+      properties:
+        compatible:
+          contains:
+            enum:
+              - allwinner,sun6i-a31-spdif
+              - allwinner,sun8i-h3-spdif
+
+    then:
+      required:
+        - resets
+
+  - if:
+      properties:
+        compatible:
+          contains:
+            const: allwinner,sun8i-h3-spdif
+
+    then:
+      properties:
+        dmas:
+          description: TX DMA Channel
+
+        dma-names:
+          const: tx
+
+    else:
+      properties:
+        dmas:
+          items:
+            - description: RX DMA Channel
+            - description: TX DMA Channel
+
+        dma-names:
+          items:
+            - const: rx
+            - const: tx
+
+required:
+  - "#sound-dai-cells"
+  - compatible
+  - reg
+  - interrupts
+  - clocks
+  - clock-names
+  - dmas
+  - dma-names
+
+additionalProperties: false
+
+examples:
+  - |
+    spdif: spdif@1c21000 {
+        #sound-dai-cells = <0>;
+        compatible = "allwinner,sun4i-a10-spdif";
+        reg = <0x01c21000 0x40>;
+        interrupts = <13>;
+        clocks = <&apb0_gates 1>, <&spdif_clk>;
+        clock-names = "apb", "spdif";
+        dmas = <&dma 0 2>, <&dma 0 2>;
+        dma-names = "rx", "tx";
+    };
+
+...

Documentation/devicetree/bindings/sound/amlogic,axg-tdm-formatters.txt

@@ -15,11 +15,15 @@ Required properties:
   * "lrclk"    : sample clock
   * "lrclk_sel": sample clock input multiplexer
 
-Example of TDMOUT_A on the A113 SoC:
+Optional property:
+- resets: phandle to the dedicated reset line of the tdm formatter.
+
+Example of TDMOUT_A on the S905X2 SoC:
 
 tdmout_a: audio-controller@500 {
 	compatible = "amlogic,axg-tdmout";
 	reg = <0x0 0x500 0x0 0x40>;
+	resets = <&clkc_audio AUD_RESET_TDMOUT_A>;
 	clocks = <&clkc_audio AUD_CLKID_TDMOUT_A>,
 		 <&clkc_audio AUD_CLKID_TDMOUT_A_SCLK>,
 		 <&clkc_audio AUD_CLKID_TDMOUT_A_SCLK_SEL>,

Documentation/devicetree/bindings/sound/amlogic,g12a-tohdmitx.txt

+* Amlogic HDMI Tx control glue
+
+Required properties:
+- compatible: "amlogic,g12a-tohdmitx"
+- reg: physical base address of the controller and length of memory
+       mapped region.
+- #sound-dai-cells: should be 1.
+
+Example on the S905X2 SoC:
+
+tohdmitx: audio-controller@744 {
+	compatible = "amlogic,g12a-tohdmitx";
+	reg = <0x0 0x744 0x0 0x4>;
+	#sound-dai-cells = <1>;
+};
+
+Example of an 'amlogic,axg-sound-card':
+
+sound {
+	compatible = "amlogic,axg-sound-card";
+
+[...]
+
+	dai-link-x {
+		sound-dai = <&tdmif_a>;
+		dai-format = "i2s";
+		dai-tdm-slot-tx-mask-0 = <1 1>;
+
+		codec-0 {
+			sound-dai = <&tohdmitx TOHDMITX_I2S_IN_A>;
+		};
+
+		codec-1 {
+			sound-dai = <&external_dac>;
+		};
+	};
+
+	dai-link-y {
+		sound-dai = <&tdmif_c>;
+		dai-format = "i2s";
+		dai-tdm-slot-tx-mask-0 = <1 1>;
+
+		codec {
+			sound-dai = <&tohdmitx TOHDMITX_I2S_IN_C>;
+		};
+	};
+
+	dai-link-z {
+		sound-dai = <&tohdmitx TOHDMITX_I2S_OUT>;
+
+		codec {
+			sound-dai = <&hdmi_tx>;
+		};
+	};
+};

Documentation/devicetree/bindings/sound/cs42xx8.txt

@@ -14,6 +14,11 @@ Required properties:
   - VA-supply, VD-supply, VLS-supply, VLC-supply: power supplies for the device,
     as covered in Documentation/devicetree/bindings/regulator/regulator.txt
 
+Optional properties:
+
+  - reset-gpios : a GPIO spec to define which pin is connected to the chip's
+    !RESET pin
+
 Example:
 
 cs42888: codec@48 {
@@ -25,4 +30,5 @@ cs42888: codec@48 {
 	VD-supply = <&reg_audio>;
 	VLS-supply = <&reg_audio>;
 	VLC-supply = <&reg_audio>;
+	reset-gpios = <&pca9557_b 1 GPIO_ACTIVE_LOW>;
 };

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

@@ -44,6 +44,9 @@ Optional properties:
   		 please refer to pinctrl-bindings.txt
 - fck_parent : Should contain a valid clock name which will be used as parent
 	       for the McASP fck
+- auxclk-fs-ratio: When McASP is bus master indicates the ratio between AUCLK
+		   and FS rate if applicable:
+		   AUCLK rate = auxclk-fs-ratio * FS rate
 
 Optional GPIO support:
 If any McASP pin need to be used as GPIO then the McASP node must have:

Documentation/devicetree/bindings/sound/madera.txt

+Cirrus Logic Madera class audio codecs
+
+This describes audio configuration bindings for these codecs.
+
+See also the core bindings for the parent MFD driver:
+See Documentation/devicetree/bindings/mfd/madera.txt
+
+and defines for values used in these bindings:
+include/dt-bindings/sound/madera.h
+
+These properties are all contained in the parent MFD node.
+
+Optional properties:
+  - cirrus,dmic-ref : Indicates how the MICBIAS pins have been externally
+    connected to DMICs on each input, one cell per input.
+    <IN1 IN2 IN3 ...>
+    A value of 0 indicates MICVDD and is the default, other values depend on the
+    codec:
+    For CS47L35 one of the CS47L35_DMIC_REF_xxx values
+    For all other codecs one of the MADERA_DMIC_REF_xxx values
+    Also see the datasheet for a description of the INn_DMIC_SUP field.
+
+  - cirrus,inmode : A list of input mode settings for each input. A maximum of
+    16 cells, with four cells per input in the order INnAL, INnAR INnBL INnBR.
+    For non-muxed inputs the first two cells for that input set the mode for
+    the left and right channel and the second two cells must be 0.
+    For muxed inputs the first two cells for that input set the mode of the
+    left and right A inputs and the second two cells set the mode of the left
+    and right B inputs.
+    Valid mode values are one of the MADERA_INMODE_xxx. If the array is shorter
+    than the number of inputs the unspecified inputs default to
+    MADERA_INMODE_DIFF.
+
+  - cirrus,out-mono : Mono bit for each output, maximum of six cells if the
+    array is shorter outputs will be set to stereo.
+
+  - cirrus,max-channels-clocked : Maximum number of channels that I2S clocks
+    will be generated for. Useful when clock master for systems where the I2S
+    bus has multiple data lines.
+    One cell for each AIF, use a value of zero for AIFs that should be handled
+    normally.
+
+  - cirrus,pdm-fmt : PDM speaker data format, must contain 2 cells
+    (OUT5 and OUT6). See the PDM_SPKn_FMT field in the datasheet for a
+    description of this value.
+    The second cell is ignored for codecs that do not have OUT6.
+
+  - cirrus,pdm-mute : PDM mute format, must contain 2 cells
+    (OUT5 and OUT6). See the PDM_SPKn_CTRL_1 register in the datasheet for a
+    description of this value.
+    The second cell is ignored for codecs that do not have OUT6.
+
+Example:
+
+cs47l35@0 {
+	compatible = "cirrus,cs47l35";
+
+	cirrus,dmic-ref = <0 0 CS47L35_DMIC_REF_MICBIAS1B 0>;
+	cirrus,inmode = <
+		MADERA_INMODE_DMIC MADERA_INMODE_DMIC /* IN1A digital */
+		MADERA_INMODE_SE   MADERA_INMODE_SE   /* IN1B single-ended */
+		MADERA_INMODE_DIFF MADERA_INMODE_DIFF /* IN2 differential */
+		0 0 	/* not used on this codec */
+	>;
+	cirrus,out-mono = <0 0 0 0 0 0>;
+	cirrus,max-channels-clocked = <2 0 0>;
+};

Documentation/devicetree/bindings/sound/max98357a.txt

@@ -9,6 +9,10 @@ Optional properties:
 - sdmode-gpios : GPIO specifier for the chip's SD_MODE pin.
         If this option is not specified then driver does not manage
         the pin state (e.g. chip is always on).
+- sdmode-delay : specify delay time for SD_MODE pin.
+        If this option is specified, which means it's required i2s clocks
+        ready before SD_MODE is unmuted in order to avoid the speaker pop noise.
+        It's observed that 5ms is sufficient.
 
 Example:
 

Documentation/devicetree/bindings/sound/rt1011.txt

+RT1011 Mono Class D Audio Amplifier
+
+This device supports I2C only.
+
+Required properties:
+
+- compatible : "realtek,rt1011".
+
+- reg : The I2C address of the device. This I2C address decide by
+        two input pins (ASEL1 and ASEL2).
+        -------------------------------------
+        |   ASEL2   |  ASEL1   |  Address   |
+        -------------------------------------
+        |     0     |    0     |   0x38     |
+        -------------------------------------
+        |     0     |    1     |   0x39     |
+        -------------------------------------
+        |     1     |    0     |   0x3a     |
+        -------------------------------------
+        |     1     |    1     |   0x3b     |
+        -------------------------------------
+
+Pins on the device (for linking into audio routes) for RT1011:
+
+  * SPO
+
+Example:
+
+rt1011: codec@38 {
+	compatible = "realtek,rt1011";
+	reg = <0x38>;
+};

Documentation/devicetree/bindings/sound/rt1308.txt

+RT1308 audio Amplifier
+
+This device supports I2C only.
+
+Required properties:
+
+- compatible : "realtek,rt1308".
+
+- reg : The I2C address of the device.
+
+
+Example:
+
+rt1308: rt1308@10 {
+	compatible = "realtek,rt1308";
+	reg = <0x10>;
+};

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

@@ -18,7 +18,7 @@ Required properties:
     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.txt
+  - pinctrl-0: see Documentation/devicetree/bindings/pinctrl/st,stm32-pinctrl.yaml
 
 Optional properties:
   - resets: Reference to a reset controller asserting the reset controller

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

@@ -41,7 +41,7 @@ SAI subnodes required properties:
 	"tx": if sai sub-block is configured as playback DAI
 	"rx": if sai sub-block is configured as capture DAI
   - pinctrl-names: should contain only value "default"
-  - pinctrl-0: see Documentation/devicetree/bindings/pinctrl/st,stm32-pinctrl.txt
+  - pinctrl-0: see Documentation/devicetree/bindings/pinctrl/st,stm32-pinctrl.yaml
 
 SAI subnodes Optional properties:
   - st,sync: specify synchronization mode.

Documentation/devicetree/bindings/sound/sun4i-i2s.txt

-* Allwinner A10 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:
-   - "allwinner,sun4i-a10-i2s"
-   - "allwinner,sun6i-a31-i2s"
-   - "allwinner,sun8i-a83t-i2s"
-   - "allwinner,sun8i-h3-i2s"
-   - "allwinner,sun50i-a64-codec-i2s"
-- 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:
-   - "apb" : clock for the I2S bus interface
-   - "mod" : module clock for the I2S controller
-- #sound-dai-cells : Must be equal to 0
-
-Required properties for the following compatibles:
-	- "allwinner,sun6i-a31-i2s"
-	- "allwinner,sun8i-a83t-i2s"
-	- "allwinner,sun8i-h3-i2s"
-	- "allwinner,sun50i-a64-codec-i2s"
-- resets: phandle to the reset line for this codec
-
-Example:
-
-i2s0: i2s@1c22400 {
-	#sound-dai-cells = <0>;
-	compatible = "allwinner,sun4i-a10-i2s";
-	reg = <0x01c22400 0x400>;
-	interrupts = <GIC_SPI 16 IRQ_TYPE_LEVEL_HIGH>;
-	clocks = <&apb0_gates 3>, <&i2s0_clk>;
-	clock-names = "apb", "mod";
-	dmas = <&dma SUN4I_DMA_NORMAL 3>,
-	       <&dma SUN4I_DMA_NORMAL 3>;
-	dma-names = "rx", "tx";
-};

Documentation/devicetree/bindings/sound/sunxi,sun4i-spdif.txt

-Allwinner Sony/Philips Digital Interface Format (S/PDIF) Controller
-
-The Allwinner S/PDIF audio block is a transceiver that allows the
-processor to receive and transmit digital audio via an coaxial cable or
-a fibre cable.
-For now only playback is supported.
-
-Required properties:
-
-  - compatible		: should be one of the following:
-    - "allwinner,sun4i-a10-spdif": for the Allwinner A10 SoC
-    - "allwinner,sun6i-a31-spdif": for the Allwinner A31 SoC
-    - "allwinner,sun8i-h3-spdif": for the Allwinner H3 SoC
-
-  - reg			: Offset and length of the register set for the device.
-
-  - interrupts		: Contains the spdif interrupt.
-
-  - dmas		: Generic dma devicetree binding as described in
-			  Documentation/devicetree/bindings/dma/dma.txt.
-
-  - dma-names		: Two dmas have to be defined, "tx" and "rx".
-
-  - clocks		: Contains an entry for each entry in clock-names.
-
-  - clock-names		: Includes the following entries:
-	"apb"		  clock for the spdif bus.
-	"spdif"		  clock for spdif controller.
-
-  - resets		: reset specifier for the ahb reset (A31 and newer only)
-
-Example:
-
-spdif: spdif@1c21000 {
-	compatible = "allwinner,sun4i-a10-spdif";
-	reg = <0x01c21000 0x40>;
-	interrupts = <13>;
-	clocks = <&apb0_gates 1>, <&spdif_clk>;
-	clock-names = "apb", "spdif";
-	dmas = <&dma 0 2>, <&dma 0 2>;
-	dma-names = "rx", "tx";
-};

MAINTAINERS

@@ -1290,7 +1290,7 @@ ARM PRIMECELL SSP PL022 SPI DRIVER
 M:	Linus Walleij <linus.walleij@linaro.org>
 L:	linux-arm-kernel@lists.infradead.org (moderated for non-subscribers)
 S:	Maintained
-F:	Documentation/devicetree/bindings/spi/spi_pl022.txt
+F:	Documentation/devicetree/bindings/spi/spi-pl022.yaml
 F:	drivers/spi/spi-pl022.c
 
 ARM PRIMECELL UART PL010 AND PL011 DRIVERS
@@ -3929,13 +3929,18 @@ W:	https://github.com/CirrusLogic/linux-drivers/wiki
 S:	Supported
 F:	Documentation/devicetree/bindings/mfd/madera.txt
 F:	Documentation/devicetree/bindings/pinctrl/cirrus,madera-pinctrl.txt
+F:	Documentation/devicetree/bindings/sound/madera.txt
+F:	include/dt-bindings/sound/madera*
 F:	include/linux/irqchip/irq-madera*
 F:	include/linux/mfd/madera/*
+F:	include/sound/madera*
 F:	drivers/gpio/gpio-madera*
 F:	drivers/irqchip/irq-madera*
 F:	drivers/mfd/madera*
 F:	drivers/mfd/cs47l*
 F:	drivers/pinctrl/cirrus/*
+F:	sound/soc/codecs/cs47l*
+F:	sound/soc/codecs/madera*
 
 CLANG-FORMAT FILE
 M:	Miguel Ojeda <miguel.ojeda.sandonis@gmail.com>

drivers/gpu/drm/vc4/vc4_hdmi.c

@@ -58,6 +58,9 @@
 struct vc4_hdmi_audio {
 	struct snd_soc_card card;
 	struct snd_soc_dai_link link;
+	struct snd_soc_dai_link_component cpu;
+	struct snd_soc_dai_link_component codec;
+	struct snd_soc_dai_link_component platform;
 	int samplerate;
 	int channels;
 	struct snd_dmaengine_dai_dma_data dma_data;
@@ -1085,12 +1088,20 @@ static int vc4_hdmi_audio_init(struct vc4_hdmi *hdmi)
 		return ret;
 	}
 
+	dai_link->cpus		= &hdmi->audio.cpu;
+	dai_link->codecs	= &hdmi->audio.codec;
+	dai_link->platforms	= &hdmi->audio.platform;
+
+	dai_link->num_cpus	= 1;
+	dai_link->num_codecs	= 1;
+	dai_link->num_platforms	= 1;
+
 	dai_link->name = "MAI";
 	dai_link->stream_name = "MAI PCM";
-	dai_link->codec_dai_name = vc4_hdmi_audio_codec_dai_drv.name;
-	dai_link->cpu_dai_name = dev_name(dev);
-	dai_link->codec_name = dev_name(dev);
-	dai_link->platform_name = dev_name(dev);
+	dai_link->codecs->dai_name = vc4_hdmi_audio_codec_dai_drv.name;
+	dai_link->cpus->dai_name = dev_name(dev);
+	dai_link->codecs->name = dev_name(dev);
+	dai_link->platforms->name = dev_name(dev);
 
 	card->dai_link = dai_link;
 	card->num_links = 1;

include/dt-bindings/sound/madera.h

+/* SPDX-License-Identifier: GPL-2.0-only */
+/*
+ * Device Tree defines for Madera codecs
+ *
+ * Copyright (C) 2016-2017 Cirrus Logic, Inc. and
+ *                         Cirrus Logic International Semiconductor Ltd.
+ */
+
+#ifndef DT_BINDINGS_SOUND_MADERA_H
+#define DT_BINDINGS_SOUND_MADERA_H
+
+#define MADERA_INMODE_DIFF		0
+#define MADERA_INMODE_SE		1
+#define MADERA_INMODE_DMIC		2
+
+#define MADERA_DMIC_REF_MICVDD		0
+#define MADERA_DMIC_REF_MICBIAS1	1
+#define MADERA_DMIC_REF_MICBIAS2	2
+#define MADERA_DMIC_REF_MICBIAS3	3
+
+#define CS47L35_DMIC_REF_MICBIAS1B	1
+#define CS47L35_DMIC_REF_MICBIAS2A	2
+#define CS47L35_DMIC_REF_MICBIAS2B	3
+
+#endif

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

+/* SPDX-License-Identifier: GPL-2.0 */
+#ifndef __DT_MESON_G12A_TOHDMITX_H
+#define __DT_MESON_G12A_TOHDMITX_H
+
+#define TOHDMITX_I2S_IN_A	0
+#define TOHDMITX_I2S_IN_B	1
+#define TOHDMITX_I2S_IN_C	2
+#define TOHDMITX_I2S_OUT	3
+#define TOHDMITX_SPDIF_IN_A	4
+#define TOHDMITX_SPDIF_IN_B	5
+#define TOHDMITX_SPDIF_OUT	6
+
+#endif /* __DT_MESON_G12A_TOHDMITX_H */

include/linux/mfd/madera/pdata.h

@@ -16,6 +16,7 @@
 #include <linux/regulator/arizona-ldo1.h>
 #include <linux/regulator/arizona-micsupp.h>
 #include <linux/regulator/machine.h>
+#include <sound/madera-pdata.h>
 
 #define MADERA_MAX_MICBIAS		4
 #define MADERA_MAX_CHILD_MICBIAS	4
@@ -39,6 +40,7 @@ struct madera_codec_pdata;
  * @gpsw:	    General purpose switch mode setting. Depends on the external
  *		    hardware connected to the switch. (See the SW1_MODE field
  *		    in the datasheet for the available values for your codec)
+ * @codec:	    Substruct of pdata for the ASoC codec driver
  */
 struct madera_pdata {
 	struct gpio_desc *reset;
@@ -53,6 +55,8 @@ struct madera_pdata {
 	int n_gpio_configs;
 
 	u32 gpsw[MADERA_MAX_GPSW];
+
+	struct madera_codec_pdata codec;
 };
 
 #endif

include/sound/hda_codec.h

@@ -18,6 +18,9 @@
 #include <sound/hda_verbs.h>
 #include <sound/hda_regmap.h>
 
+#define IS_BXT(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x5a98)
+#define IS_CFL(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0xa348)
+
 /*
  * Structures
  */

include/sound/madera-pdata.h

+/* SPDX-License-Identifier: GPL-2.0-only */
+/*
+ * Platform data for Madera codec driver
+ *
+ * Copyright (C) 2016-2019 Cirrus Logic, Inc. and
+ *                         Cirrus Logic International Semiconductor Ltd.
+ */
+
+#ifndef MADERA_CODEC_PDATA_H
+#define MADERA_CODEC_PDATA_H
+
+#include <linux/kernel.h>
+
+#define MADERA_MAX_INPUT		6
+#define MADERA_MAX_MUXED_CHANNELS	4
+#define MADERA_MAX_OUTPUT		6
+#define MADERA_MAX_AIF			4
+#define MADERA_MAX_PDM_SPK		2
+#define MADERA_MAX_DSP			7
+
+/**
+ * struct madera_codec_pdata
+ *
+ * @max_channels_clocked: Maximum number of channels that I2S clocks will be
+ *			  generated for. Useful when clock master for systems
+ *			  where the I2S bus has multiple data lines.
+ * @dmic_ref:		  Indicates how the MICBIAS pins have been externally
+ *			  connected to DMICs on each input. A value of 0
+ *			  indicates MICVDD and is the default. Other values are:
+ *			  For CS47L35 one of the CS47L35_DMIC_REF_xxx values
+ *			  For all other codecs one of the MADERA_DMIC_REF_xxx
+ *			  Also see the datasheet for a description of the
+ *			  INn_DMIC_SUP field.
+ * @inmode:		  Mode for the ADC inputs. One of the MADERA_INMODE_xxx
+ *			  values. Two-dimensional array
+ *			  [input_number][channel number], with four slots per
+ *			  input in the order
+ *			  [n][0]=INnAL [n][1]=INnAR [n][2]=INnBL [n][3]=INnBR
+ * @out_mono:		  For each output set the value to TRUE to indicate that
+ *			  the output is mono. [0]=OUT1, [1]=OUT2, ...
+ * @pdm_fmt:		  PDM speaker data format. See the PDM_SPKn_FMT field in
+ *			  the datasheet for a description of this value.
+ * @pdm_mute:		  PDM mute format. See the PDM_SPKn_CTRL_1 register
+ *			  in the datasheet for a description of this value.
+ */
+struct madera_codec_pdata {
+	u32 max_channels_clocked[MADERA_MAX_AIF];
+
+	u32 dmic_ref[MADERA_MAX_INPUT];
+
+	u32 inmode[MADERA_MAX_INPUT][MADERA_MAX_MUXED_CHANNELS];
+
+	bool out_mono[MADERA_MAX_OUTPUT];
+
+	u32 pdm_fmt[MADERA_MAX_PDM_SPK];
+	u32 pdm_mute[MADERA_MAX_PDM_SPK];
+};
+
+#endif

include/sound/simple_card_utils.h

@@ -42,6 +42,7 @@ struct asoc_simple_priv {
 	struct simple_dai_props {
 		struct asoc_simple_dai *cpu_dai;
 		struct asoc_simple_dai *codec_dai;
+		struct snd_soc_dai_link_component cpus;   /* single cpu */
 		struct snd_soc_dai_link_component codecs; /* single codec */
 		struct snd_soc_dai_link_component platforms;
 		struct asoc_simple_data adata;
@@ -80,16 +81,12 @@ int asoc_simple_parse_card_name(struct snd_soc_card *card,
 				char *prefix);
 
 #define asoc_simple_parse_clk_cpu(dev, node, dai_link, simple_dai)		\
-	asoc_simple_parse_clk(dev, node, dai_link->cpu_of_node, simple_dai, \
-				   dai_link->cpu_dai_name, NULL)
+	asoc_simple_parse_clk(dev, node, simple_dai, dai_link->cpus)
 #define asoc_simple_parse_clk_codec(dev, node, dai_link, simple_dai)	\
-	asoc_simple_parse_clk(dev, node, dai_link->codec_of_node, simple_dai,\
-				   dai_link->codec_dai_name, dai_link->codecs)
+	asoc_simple_parse_clk(dev, node, simple_dai, dai_link->codecs)
 int asoc_simple_parse_clk(struct device *dev,
 			  struct device_node *node,
-			  struct device_node *dai_of_node,
 			  struct asoc_simple_dai *simple_dai,
-			  const char *dai_name,
 			  struct snd_soc_dai_link_component *dlc);
 int asoc_simple_startup(struct snd_pcm_substream *substream);
 void asoc_simple_shutdown(struct snd_pcm_substream *substream);
@@ -100,16 +97,11 @@ int asoc_simple_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
 				   struct snd_pcm_hw_params *params);
 
 #define asoc_simple_parse_cpu(node, dai_link, is_single_link)	\
-	asoc_simple_parse_dai(node, NULL,				\
-		&dai_link->cpu_of_node,					\
-		&dai_link->cpu_dai_name, is_single_link)
+	asoc_simple_parse_dai(node, dai_link->cpus, is_single_link)
 #define asoc_simple_parse_codec(node, dai_link)	\
-	asoc_simple_parse_dai(node, dai_link->codecs,			\
-				   &dai_link->codec_of_node,			\
-				   &dai_link->codec_dai_name, NULL)
+	asoc_simple_parse_dai(node, dai_link->codecs, NULL)
 #define asoc_simple_parse_platform(node, dai_link)	\
-	asoc_simple_parse_dai(node, dai_link->platforms,			\
-		&dai_link->platform_of_node, NULL, NULL)
+	asoc_simple_parse_dai(node, dai_link->platforms, NULL)
 
 #define asoc_simple_parse_tdm(np, dai)			\
 	snd_soc_of_parse_tdm_slot(np,	&(dai)->tx_slot_mask,	\

include/sound/soc.h

@@ -900,17 +900,6 @@ struct snd_soc_dai_link {
 	const char *name;			/* Codec name */
 	const char *stream_name;		/* Stream name */
 
-	/*
-	 *	cpu_name
-	 *	cpu_of_node
-	 *	cpu_dai_name
-	 *
-	 * These are legacy style, and will be replaced to
-	 * modern style (= snd_soc_dai_link_component) in the future,
-	 * but, not yet supported so far.
-	 * If modern style was supported for CPU, all driver will switch
-	 * to use it, and, legacy style code will be removed from ALSA SoC.
-	 */
 	/*
 	 * You MAY specify the link's CPU-side device, either by device name,
 	 * or by DT/OF node, but not both. If this information is omitted,
@@ -918,57 +907,27 @@ struct snd_soc_dai_link {
 	 * must be globally unique. These fields are currently typically used
 	 * only for codec to codec links, or systems using device tree.
 	 */
-	const char *cpu_name;
-	struct device_node *cpu_of_node;
 	/*
 	 * You MAY specify the DAI name of the CPU DAI. If this information is
 	 * omitted, the CPU-side DAI is matched using .cpu_name/.cpu_of_node
 	 * only, which only works well when that device exposes a single DAI.
 	 */
-	const char *cpu_dai_name;
+	struct snd_soc_dai_link_component *cpus;
+	unsigned int num_cpus;
 
-	/*
-	 *	codec_name
-	 *	codec_of_node
-	 *	codec_dai_name
-	 *
-	 * These are legacy style, it will be converted to modern style
-	 * (= snd_soc_dai_link_component) automatically in soc-core
-	 * if driver is using legacy style.
-	 * Driver shouldn't use both legacy and modern style in the same time.
-	 * If modern style was supported for CPU, all driver will switch
-	 * to use it, and, legacy style code will be removed from ALSA SoC.
-	 */
 	/*
 	 * You MUST specify the link's codec, either by device name, or by
 	 * DT/OF node, but not both.
 	 */
-	const char *codec_name;
-	struct device_node *codec_of_node;
 	/* You MUST specify the DAI name within the codec */
-	const char *codec_dai_name;
-
 	struct snd_soc_dai_link_component *codecs;
 	unsigned int num_codecs;
 
-	/*
-	 *	platform_name
-	 *	platform_of_node
-	 *
-	 * These are legacy style, it will be converted to modern style
-	 * (= snd_soc_dai_link_component) automatically in soc-core
-	 * if driver is using legacy style.
-	 * Driver shouldn't use both legacy and modern style in the same time.
-	 * If modern style was supported for CPU, all driver will switch
-	 * to use it, and, legacy style code will be removed from ALSA SoC.
-	 */
 	/*
 	 * You MAY specify the link's platform/PCM/DMA driver, either by
 	 * device name, or by DT/OF node, but not both. Some forms of link
-	 * do not need a platform.
+	 * do not need a platform. In such case, platforms are not mandatory.
 	 */
-	const char *platform_name;
-	struct device_node *platform_of_node;
 	struct snd_soc_dai_link_component *platforms;
 	unsigned int num_platforms;
 
@@ -1030,12 +989,6 @@ struct snd_soc_dai_link {
 	/* Do not create a PCM for this DAI link (Backend link) */
 	unsigned int ignore:1;
 
-	/*
-	 * This driver uses legacy platform naming. Set by the core, machine
-	 * drivers should not modify this value.
-	 */
-	unsigned int legacy_platform:1;
-
 	struct list_head list; /* DAI link list of the soc card */
 	struct snd_soc_dobj dobj; /* For topology */
 };
@@ -1044,6 +997,100 @@ struct snd_soc_dai_link {
 	     ((i) < link->num_codecs) && ((codec) = &link->codecs[i]);	\
 	     (i)++)
 
+#define for_each_link_platforms(link, i, platform)			\
+	for ((i) = 0;							\
+	     ((i) < link->num_platforms) &&				\
+	     ((platform) = &link->platforms[i]);			\
+	     (i)++)
+
+/*
+ * Sample 1 : Single CPU/Codec/Platform
+ *
+ * SND_SOC_DAILINK_DEFS(test,
+ *	DAILINK_COMP_ARRAY(COMP_CPU("cpu_dai")),
+ *	DAILINK_COMP_ARRAY(COMP_CODEC("codec", "codec_dai")),
+ *	DAILINK_COMP_ARRAY(COMP_PLATFORM("platform")));
+ *
+ * struct snd_soc_dai_link link = {
+ *	...
+ *	SND_SOC_DAILINK_REG(test),
+ * };
+ *
+ * Sample 2 : Multi CPU/Codec, no Platform
+ *
+ * SND_SOC_DAILINK_DEFS(test,
+ *	DAILINK_COMP_ARRAY(COMP_CPU("cpu_dai1"),
+ *			   COMP_CPU("cpu_dai2")),
+ *	DAILINK_COMP_ARRAY(COMP_CODEC("codec1", "codec_dai1"),
+ *			   COMP_CODEC("codec2", "codec_dai2")));
+ *
+ * struct snd_soc_dai_link link = {
+ *	...
+ *	SND_SOC_DAILINK_REG(test),
+ * };
+ *
+ * Sample 3 : Define each CPU/Codec/Platform manually
+ *
+ * SND_SOC_DAILINK_DEF(test_cpu,
+ *		DAILINK_COMP_ARRAY(COMP_CPU("cpu_dai1"),
+ *				   COMP_CPU("cpu_dai2")));
+ * SND_SOC_DAILINK_DEF(test_codec,
+ *		DAILINK_COMP_ARRAY(COMP_CODEC("codec1", "codec_dai1"),
+ *				   COMP_CODEC("codec2", "codec_dai2")));
+ * SND_SOC_DAILINK_DEF(test_platform,
+ *		DAILINK_COMP_ARRAY(COMP_PLATFORM("platform")));
+ *
+ * struct snd_soc_dai_link link = {
+ *	...
+ *	SND_SOC_DAILINK_REG(test_cpu,
+ *			    test_codec,
+ *			    test_platform),
+ * };
+ *
+ * Sample 4 : Sample3 without platform
+ *
+ * struct snd_soc_dai_link link = {
+ *	...
+ *	SND_SOC_DAILINK_REG(test_cpu,
+ *			    test_codec);
+ * };
+ */
+
+#define SND_SOC_DAILINK_REG1(name)	 SND_SOC_DAILINK_REG3(name##_cpus, name##_codecs, name##_platforms)
+#define SND_SOC_DAILINK_REG2(cpu, codec) SND_SOC_DAILINK_REG3(cpu, codec, null_dailink_component)
+#define SND_SOC_DAILINK_REG3(cpu, codec, platform)	\
+	.cpus		= cpu,				\
+	.num_cpus	= ARRAY_SIZE(cpu),		\
+	.codecs		= codec,			\
+	.num_codecs	= ARRAY_SIZE(codec),		\
+	.platforms	= platform,			\
+	.num_platforms	= ARRAY_SIZE(platform)
+
+#define SND_SOC_DAILINK_REGx(_1, _2, _3, func, ...) func
+#define SND_SOC_DAILINK_REG(...) \
+	SND_SOC_DAILINK_REGx(__VA_ARGS__,		\
+			SND_SOC_DAILINK_REG3,	\
+			SND_SOC_DAILINK_REG2,	\
+			SND_SOC_DAILINK_REG1)(__VA_ARGS__)
+
+#define SND_SOC_DAILINK_DEF(name, def...)		\
+	static struct snd_soc_dai_link_component name[]	= { def }
+
+#define SND_SOC_DAILINK_DEFS(name, cpu, codec, platform...)	\
+	SND_SOC_DAILINK_DEF(name##_cpus, cpu);			\
+	SND_SOC_DAILINK_DEF(name##_codecs, codec);		\
+	SND_SOC_DAILINK_DEF(name##_platforms, platform)
+
+#define DAILINK_COMP_ARRAY(param...)	param
+#define COMP_EMPTY()			{ }
+#define COMP_CPU(_dai)			{ .dai_name = _dai, }
+#define COMP_CODEC(_name, _dai)		{ .name = _name, .dai_name = _dai, }
+#define COMP_PLATFORM(_name)		{ .name = _name }
+#define COMP_DUMMY()			{ .name = "snd-soc-dummy", .dai_name = "snd-soc-dummy-dai", }
+
+extern struct snd_soc_dai_link_component null_dailink_component[0];
+
+
 struct snd_soc_codec_conf {
 	/*
 	 * specify device either by device name, or by
@@ -1189,7 +1236,7 @@ struct snd_soc_card {
 	     (i)++)
 
 #define for_each_card_links(card, link)				\
-	list_for_each_entry(dai_link, &(card)->dai_link_list, list)
+	list_for_each_entry(link, &(card)->dai_link_list, list)
 #define for_each_card_links_safe(card, link, _link)			\
 	list_for_each_entry_safe(link, _link, &(card)->dai_link_list, list)
 
@@ -1214,7 +1261,6 @@ struct snd_soc_pcm_runtime {
 
 	/* Dynamic PCM BE runtime data */
 	struct snd_soc_dpcm_runtime dpcm[2];
-	int fe_compr;
 
 	long pmdown_time;
 
@@ -1239,6 +1285,7 @@ struct snd_soc_pcm_runtime {
 	/* bit field */
 	unsigned int dev_registered:1;
 	unsigned int pop_wait:1;
+	unsigned int fe_compr:1; /* for Dynamic PCM */
 };
 #define for_each_rtd_codec_dai(rtd, i, dai)\
 	for ((i) = 0;						       \
@@ -1607,15 +1654,11 @@ int snd_soc_fixup_dai_links_platform_name(struct snd_soc_card *card,
 		if (!name)
 			return -ENOMEM;
 
-		if (dai_link->platforms)
-			/* only single platform is supported for now */
-			dai_link->platforms->name = name;
-		else
-			/*
-			 * legacy mode, this case will be removed when all
-			 * derivers are switched to modern style dai_link.
-			 */
-			dai_link->platform_name = name;
+		if (!dai_link->platforms)
+			return -EINVAL;
+
+		/* only single platform is supported for now */
+		dai_link->platforms->name = name;
 	}
 
 	return 0;

include/sound/sof/dai-intel.h

@@ -167,9 +167,10 @@ struct sof_ipc_dai_dmic_params {
 
 	uint32_t wake_up_time;      /**< Time from clock start to data (us) */
 	uint32_t min_clock_on_time; /**< Min. time that clk is kept on (us) */
+	uint32_t unmute_ramp_time;  /**< Length of logarithmic gain ramp (ms) */
 
 	/* reserved for future use */
-	uint32_t reserved[6];
+	uint32_t reserved[5];
 
 	/**< variable number of pdm controller config */
 	struct sof_ipc_dai_dmic_pdm_ctrl pdm[0];

include/sound/sof/header.h

@@ -49,6 +49,7 @@
 #define SOF_IPC_GLB_DAI_MSG			SOF_GLB_TYPE(0x8U)
 #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)
 
 /*
  * DSP Command Message Types
@@ -99,9 +100,13 @@
 #define SOF_IPC_STREAM_VORBIS_PARAMS		SOF_CMD_TYPE(0x010)
 #define SOF_IPC_STREAM_VORBIS_FREE		SOF_CMD_TYPE(0x011)
 
-/* trace and debug */
+/* trace */
 #define SOF_IPC_TRACE_DMA_PARAMS		SOF_CMD_TYPE(0x001)
 #define SOF_IPC_TRACE_DMA_POSITION		SOF_CMD_TYPE(0x002)
+#define SOF_IPC_TRACE_DMA_PARAMS_EXT		SOF_CMD_TYPE(0x003)
+
+/* debug */
+#define SOF_IPC_TEST_IPC_FLOOD                  SOF_CMD_TYPE(0x001)
 
 /* Get message component id */
 #define SOF_IPC_MESSAGE_ID(x)			((x) & 0xffff)

include/sound/sof/topology.h

@@ -35,6 +35,7 @@ enum sof_comp_type {
 	SOF_COMP_KEYWORD_DETECT,
 	SOF_COMP_KPB,			/* A key phrase buffer component */
 	SOF_COMP_SELECTOR,		/**< channel selector component */
+	SOF_COMP_DEMUX,
 	/* keep FILEREAD/FILEWRITE as the last ones */
 	SOF_COMP_FILEREAD = 10000,	/**< host test based file IO */
 	SOF_COMP_FILEWRITE = 10001,	/**< host test based file IO */
@@ -83,9 +84,9 @@ struct sof_ipc_buffer {
 struct sof_ipc_comp_config {
 	struct sof_ipc_cmd_hdr hdr;
 	uint32_t periods_sink;	/**< 0 means variable */
-	uint32_t periods_source;	/**< 0 means variable */
+	uint32_t periods_source;/**< 0 means variable */
 	uint32_t reserved1;	/**< reserved */
-	uint32_t frame_fmt;		/**< SOF_IPC_FRAME_ */
+	uint32_t frame_fmt;	/**< SOF_IPC_FRAME_ */
 	uint32_t xrun_action;
 
 	/* reserved for future use */
@@ -175,6 +176,8 @@ enum sof_ipc_process_type {
 	SOF_PROCESS_KEYWORD_DETECT,	/**< Keyword Detection */
 	SOF_PROCESS_KPB,		/**< KeyPhrase Buffer Manager */
 	SOF_PROCESS_CHAN_SELECTOR,	/**< Channel Selector */
+	SOF_PROCESS_MUX,
+	SOF_PROCESS_DEMUX,
 };
 
 /* generic "effect", "codec" or proprietary processing component */

include/sound/sof/trace.h

@@ -19,12 +19,22 @@
 #define SOF_TRACE_FILENAME_SIZE		32
 
 /* DMA for Trace params info - SOF_IPC_DEBUG_DMA_PARAMS */
+/* Deprecated - use sof_ipc_dma_trace_params_ext */
 struct sof_ipc_dma_trace_params {
 	struct sof_ipc_cmd_hdr hdr;
 	struct sof_ipc_host_buffer buffer;
 	uint32_t stream_tag;
 }  __packed;
 
+/* DMA for Trace params info - SOF_IPC_DEBUG_DMA_PARAMS_EXT */
+struct sof_ipc_dma_trace_params_ext {
+	struct sof_ipc_cmd_hdr hdr;
+	struct sof_ipc_host_buffer buffer;
+	uint32_t stream_tag;
+	uint64_t timestamp_ns; /* in nanosecond */
+	uint32_t reserved[8];
+}  __packed;
+
 /* DMA for Trace params info - SOF_IPC_DEBUG_DMA_PARAMS */
 struct sof_ipc_dma_trace_posn {
 	struct sof_ipc_reply rhdr;
@@ -56,7 +66,9 @@ struct sof_ipc_dma_trace_posn {
 #define SOF_IPC_PANIC_WFI			(SOF_IPC_PANIC_MAGIC | 0xa)
 #define SOF_IPC_PANIC_ASSERT			(SOF_IPC_PANIC_MAGIC | 0xb)
 
-/* panic info include filename and line number */
+/* panic info include filename and line number
+ * filename array will not include null terminator if fully filled
+ */
 struct sof_ipc_panic_info {
 	struct sof_ipc_hdr hdr;
 	uint32_t code;			/* SOF_IPC_PANIC_ */

include/uapi/sound/sof/abi.h

@@ -26,7 +26,7 @@
 
 /* SOF ABI version major, minor and patch numbers */
 #define SOF_ABI_MAJOR 3
-#define SOF_ABI_MINOR 6
+#define SOF_ABI_MINOR 8
 #define SOF_ABI_PATCH 0
 
 /* SOF ABI version number. Format within 32bit word is MMmmmppp */

include/uapi/sound/sof/eq.h

-/* SPDX-License-Identifier: ((GPL-2.0 WITH Linux-syscall-note) 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) 2018 Intel Corporation. All rights reserved.
- */
-
-#ifndef __INCLUDE_UAPI_SOUND_SOF_USER_EQ_H__
-#define __INCLUDE_UAPI_SOUND_SOF_USER_EQ_H__
-
-/* FIR EQ type */
-
-#define SOF_EQ_FIR_IDX_SWITCH	0
-
-#define SOF_EQ_FIR_MAX_SIZE 4096 /* Max size allowed for coef data in bytes */
-
-#define SOF_EQ_FIR_MAX_LENGTH 192 /* Max length for individual filter */
-
-#define SOF_EQ_FIR_MAX_RESPONSES 8 /* A blob can define max 8 FIR EQs */
-
-/*
- * eq_fir_configuration data structure contains this information
- *     uint32_t size
- *	   This is the number of bytes need to store the received EQ
- *	   configuration.
- *     uint16_t channels_in_config
- *         This describes the number of channels in this EQ config data. It
- *         can be different from PLATFORM_MAX_CHANNELS.
- *     uint16_t number_of_responses
- *         0=no responses, 1=one response defined, 2=two responses defined, etc.
- *     int16_t data[]
- *         assign_response[channels_in_config]
- *             0 = use first response, 1 = use 2nd response, etc.
- *             E.g. {0, 0, 0, 0, 1, 1, 1, 1} would apply to channels 0-3 the
- *	       same first defined response and for to channels 4-7 the second.
- *         coef_data[]
- *             Repeated data
- *             { filter_length, output_shift, h[] }
- *	       for every EQ response defined where vector h has filter_length
- *             number of coefficients. Coefficients in h[] are in Q1.15 format.
- *             E.g. 16384 (Q1.15) = 0.5. The shifts are number of right shifts.
- *
- * NOTE: The channels_in_config must be even to have coef_data aligned to
- * 32 bit word in RAM. Therefore a mono EQ assign must be duplicated to 2ch
- * even if it would never used. Similarly a 5ch EQ assign must be increased
- * to 6ch. EQ init will return an error if this is not met.
- *
- * NOTE: The filter_length must be multiple of four. Therefore the filter must
- * be padded from the end with zeros have this condition met.
- */
-
-struct sof_eq_fir_config {
-	uint32_t size;
-	uint16_t channels_in_config;
-	uint16_t number_of_responses;
-
-	/* reserved */
-	uint32_t reserved[4];
-
-	int16_t data[];
-} __packed;
-
-struct sof_eq_fir_coef_data {
-	int16_t length; /* Number of FIR taps */
-	int16_t out_shift; /* Amount of right shifts at output */
-
-	/* reserved */
-	uint32_t reserved[4];
-
-	int16_t coef[]; /* FIR coefficients */
-} __packed;
-
-/* In the struct above there's two 16 bit words (length, shift) and four
- * reserved 32 bit words before the actual FIR coefficients. This information
- * is used in parsing of the configuration blob.
- */
-#define SOF_EQ_FIR_COEF_NHEADER \
-	(sizeof(struct sof_eq_fir_coef_data) / sizeof(int16_t))
-
-/* IIR EQ type */
-
-#define SOF_EQ_IIR_IDX_SWITCH   0
-
-#define SOF_EQ_IIR_MAX_SIZE 1024 /* Max size allowed for coef data in bytes */
-
-#define SOF_EQ_IIR_MAX_RESPONSES 8 /* A blob can define max 8 IIR EQs */
-
-/* eq_iir_configuration
- *     uint32_t channels_in_config
- *         This describes the number of channels in this EQ config data. It
- *         can be different from PLATFORM_MAX_CHANNELS.
- *     uint32_t number_of_responses_defined
- *         0=no responses, 1=one response defined, 2=two responses defined, etc.
- *     int32_t data[]
- *         Data consist of two parts. First is the response assign vector that
- *	   has length of channels_in_config. The latter part is coefficient
- *         data.
- *         uint32_t assign_response[channels_in_config]
- *             -1 = not defined, 0 = use first response, 1 = use 2nd, etc.
- *             E.g. {0, 0, 0, 0, -1, -1, -1, -1} would apply to channels 0-3 the
- *             same first defined response and leave channels 4-7 unequalized.
- *         coefficient_data[]
- *             <1st EQ>
- *             uint32_t num_biquads
- *             uint32_t num_biquads_in_series
- *             <1st biquad>
- *             int32_t coef_a2       Q2.30 format
- *             int32_t coef_a1       Q2.30 format
- *             int32_t coef_b2       Q2.30 format
- *             int32_t coef_b1       Q2.30 format
- *             int32_t coef_b0       Q2.30 format
- *             int32_t output_shift  number of shifts right, shift left is negative
- *             int32_t output_gain   Q2.14 format
- *             <2nd biquad>
- *             ...
- *             <2nd EQ>
- *
- *         Note: A flat response biquad can be made with a section set to
- *         b0 = 1.0, gain = 1.0, and other parameters set to 0
- *         {0, 0, 0, 0, 1073741824, 0, 16484}
- */
-
-struct sof_eq_iir_config {
-	uint32_t size;
-	uint32_t channels_in_config;
-	uint32_t number_of_responses;
-
-	/* reserved */
-	uint32_t reserved[4];
-
-	int32_t data[]; /* eq_assign[channels], eq 0, eq 1, ... */
-} __packed;
-
-struct sof_eq_iir_header_df2t {
-	uint32_t num_sections;
-	uint32_t num_sections_in_series;
-
-	/* reserved */
-	uint32_t reserved[4];
-
-	int32_t biquads[]; /* Repeated biquad coefficients */
-} __packed;
-
-struct sof_eq_iir_biquad_df2t {
-	int32_t a2; /* Q2.30 */
-	int32_t a1; /* Q2.30 */
-	int32_t b2; /* Q2.30 */
-	int32_t b1; /* Q2.30 */
-	int32_t b0; /* Q2.30 */
-	int32_t output_shift; /* Number of right shifts */
-	int32_t output_gain;  /* Q2.14 */
-} __packed;
-
-/* A full 22th order equalizer with 11 biquads cover octave bands 1-11 in
- * in the 0 - 20 kHz bandwidth.
- */
-#define SOF_EQ_IIR_DF2T_BIQUADS_MAX 11
-
-/* The number of int32_t words in sof_eq_iir_header_df2t:
- *	num_sections, num_sections_in_series, reserved[4]
- */
-#define SOF_EQ_IIR_NHEADER_DF2T \
-	(sizeof(struct sof_eq_iir_header_df2t) / sizeof(int32_t))
-
-/* The number of int32_t words in sof_eq_iir_biquad_df2t:
- *	a2, a1, b2, b1, b0, output_shift, output_gain
- */
-#define SOF_EQ_IIR_NBIQUAD_DF2T \
-	(sizeof(struct sof_eq_iir_biquad_df2t) / sizeof(int32_t))
-
-#endif

include/uapi/sound/sof/manifest.h

-/* SPDX-License-Identifier: ((GPL-2.0 WITH Linux-syscall-note) 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) 2018 Intel Corporation. All rights reserved.
- */
-
-#ifndef __INCLUDE_UAPI_SOUND_SOF_USER_MANIFEST_H__
-#define __INCLUDE_UAPI_SOUND_SOF_USER_MANIFEST_H__
-
-/* start offset for base FW module */
-#define SOF_MAN_ELF_TEXT_OFFSET		0x2000
-
-/* FW Extended Manifest Header id = $AE1 */
-#define SOF_MAN_EXT_HEADER_MAGIC	0x31454124
-
-/* module type load type */
-#define SOF_MAN_MOD_TYPE_BUILTIN	0
-#define SOF_MAN_MOD_TYPE_MODULE		1
-
-struct sof_man_module_type {
-	uint32_t load_type:4;	/* SOF_MAN_MOD_TYPE_ */
-	uint32_t auto_start:1;
-	uint32_t domain_ll:1;
-	uint32_t domain_dp:1;
-	uint32_t rsvd_:25;
-};
-
-/* segment flags.type */
-#define SOF_MAN_SEGMENT_TEXT		0
-#define SOF_MAN_SEGMENT_RODATA		1
-#define SOF_MAN_SEGMENT_DATA		1
-#define SOF_MAN_SEGMENT_BSS		2
-#define SOF_MAN_SEGMENT_EMPTY		15
-
-union sof_man_segment_flags {
-	uint32_t ul;
-	struct {
-		uint32_t contents:1;
-		uint32_t alloc:1;
-		uint32_t load:1;
-		uint32_t readonly:1;
-		uint32_t code:1;
-		uint32_t data:1;
-		uint32_t _rsvd0:2;
-		uint32_t type:4;	/* MAN_SEGMENT_ */
-		uint32_t _rsvd1:4;
-		uint32_t length:16;	/* of segment in pages */
-	} r;
-} __packed;
-
-/*
- * Module segment descriptor. Used by ROM - Immutable.
- */
-struct sof_man_segment_desc {
-	union sof_man_segment_flags flags;
-	uint32_t v_base_addr;
-	uint32_t file_offset;
-} __packed;
-
-/*
- * The firmware binary can be split into several modules.
- */
-
-#define SOF_MAN_MOD_ID_LEN		4
-#define SOF_MAN_MOD_NAME_LEN		8
-#define SOF_MAN_MOD_SHA256_LEN		32
-#define SOF_MAN_MOD_ID			{'$', 'A', 'M', 'E'}
-
-/*
- * Each module has an entry in the FW header. Used by ROM - Immutable.
- */
-struct sof_man_module {
-	uint8_t struct_id[SOF_MAN_MOD_ID_LEN];	/* SOF_MAN_MOD_ID */
-	uint8_t name[SOF_MAN_MOD_NAME_LEN];
-	uint8_t uuid[16];
-	struct sof_man_module_type type;
-	uint8_t hash[SOF_MAN_MOD_SHA256_LEN];
-	uint32_t entry_point;
-	uint16_t cfg_offset;
-	uint16_t cfg_count;
-	uint32_t affinity_mask;
-	uint16_t instance_max_count;	/* max number of instances */
-	uint16_t instance_bss_size;	/* instance (pages) */
-	struct sof_man_segment_desc segment[3];
-} __packed;
-
-/*
- * Each module has a configuration in the FW header. Used by ROM - Immutable.
- */
-struct sof_man_mod_config {
-	uint32_t par[4];	/* module parameters */
-	uint32_t is_pages;	/* actual size of instance .bss (pages) */
-	uint32_t cps;		/* cycles per second */
-	uint32_t ibs;		/* input buffer size (bytes) */
-	uint32_t obs;		/* output buffer size (bytes) */
-	uint32_t module_flags;	/* flags, reserved for future use */
-	uint32_t cpc;		/* cycles per single run */
-	uint32_t obls;		/* output block size, reserved for future use */
-} __packed;
-
-/*
- * FW Manifest Header
- */
-
-#define SOF_MAN_FW_HDR_FW_NAME_LEN	8
-#define SOF_MAN_FW_HDR_ID		{'$', 'A', 'M', '1'}
-#define SOF_MAN_FW_HDR_NAME		"ADSPFW"
-#define SOF_MAN_FW_HDR_FLAGS		0x0
-#define SOF_MAN_FW_HDR_FEATURES		0xff
-
-/*
- * The firmware has a standard header that is checked by the ROM on firmware
- * loading. preload_page_count is used by DMA code loader and is entire
- * image size on CNL. i.e. CNL: total size of the binary’s .text and .rodata
- * Used by ROM - Immutable.
- */
-struct sof_man_fw_header {
-	uint8_t header_id[4];
-	uint32_t header_len;
-	uint8_t name[SOF_MAN_FW_HDR_FW_NAME_LEN];
-	/* number of pages of preloaded image loaded by driver */
-	uint32_t preload_page_count;
-	uint32_t fw_image_flags;
-	uint32_t feature_mask;
-	uint16_t major_version;
-	uint16_t minor_version;
-	uint16_t hotfix_version;
-	uint16_t build_version;
-	uint32_t num_module_entries;
-	uint32_t hw_buf_base_addr;
-	uint32_t hw_buf_length;
-	/* target address for binary loading as offset in IMR - must be == base offset */
-	uint32_t load_offset;
-} __packed;
-
-/*
- * Firmware manifest descriptor. This can contain N modules and N module
- * configs. Used by ROM - Immutable.
- */
-struct sof_man_fw_desc {
-	struct sof_man_fw_header header;
-
-	/* Warning - hack for module arrays. For some unknown reason the we
-	 * have a variable size array of struct man_module followed by a
-	 * variable size array of struct mod_config. These should have been
-	 * merged into a variable array of a parent structure. We have to hack
-	 * around this in many places....
-	 *
-	 * struct sof_man_module man_module[];
-	 * struct sof_man_mod_config mod_config[];
-	 */
-
-} __packed;
-
-/*
- * Component Descriptor. Used by ROM - Immutable.
- */
-struct sof_man_component_desc {
-	uint32_t reserved[2];	/* all 0 */
-	uint32_t version;
-	uint8_t hash[SOF_MAN_MOD_SHA256_LEN];
-	uint32_t base_offset;
-	uint32_t limit_offset;
-	uint32_t attributes[4];
-} __packed;
-
-/*
- * Audio DSP extended metadata. Used by ROM - Immutable.
- */
-struct sof_man_adsp_meta_file_ext {
-	uint32_t ext_type;	/* always 17 for ADSP extension */
-	uint32_t ext_len;
-	uint32_t imr_type;
-	uint8_t reserved[16];	/* all 0 */
-	struct sof_man_component_desc comp_desc[1];
-} __packed;
-
-/*
- * Module Manifest for rimage module metadata. Not used by ROM.
- */
-struct sof_man_module_manifest {
-	struct sof_man_module module;
-	uint32_t text_size;
-} __packed;
-
-#endif

include/uapi/sound/sof/tokens.h

@@ -85,6 +85,7 @@
 #define SOF_TKN_INTEL_DMIC_NUM_PDM_ACTIVE	605
 #define SOF_TKN_INTEL_DMIC_SAMPLE_RATE		608
 #define SOF_TKN_INTEL_DMIC_FIFO_WORD_LENGTH	609
+#define SOF_TKN_INTEL_DMIC_UNMUTE_RAMP_TIME_MS  610
 
 /* DMIC PDM */
 #define SOF_TKN_INTEL_DMIC_PDM_CTRL_ID		700

include/uapi/sound/sof/tone.h

-/* SPDX-License-Identifier: ((GPL-2.0 WITH Linux-syscall-note) 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) 2018 Intel Corporation. All rights reserved.
-*/
-
-#ifndef __INCLUDE_UAPI_SOUND_SOF_USER_TONE_H__
-#define __INCLUDE_UAPI_SOUND_SOF_USER_TONE_H__
-
-#define SOF_TONE_IDX_FREQUENCY		0
-#define SOF_TONE_IDX_AMPLITUDE		1
-#define SOF_TONE_IDX_FREQ_MULT		2
-#define SOF_TONE_IDX_AMPL_MULT		3
-#define SOF_TONE_IDX_LENGTH		4
-#define SOF_TONE_IDX_PERIOD		5
-#define SOF_TONE_IDX_REPEATS		6
-#define SOF_TONE_IDX_LIN_RAMP_STEP	7
-
-#endif

include/uapi/sound/sof/trace.h

-/* SPDX-License-Identifier: ((GPL-2.0 WITH Linux-syscall-note) 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) 2018 Intel Corporation. All rights reserved.
- */
-
-#ifndef __INCLUDE_UAPI_SOUND_SOF_USER_TRACE_H__
-#define __INCLUDE_UAPI_SOUND_SOF_USER_TRACE_H__
-
-/*
- * Host system time.
- *
- * This property is used by the driver to pass down information about
- * current system time. It is expressed in us.
- * FW translates timestamps (in log entries, probe pockets) to this time
- * domain.
- *
- * (cavs: SystemTime).
- */
-struct system_time {
-	uint32_t val_l;  /* Lower dword of current host time value */
-	uint32_t val_u;  /* Upper dword of current host time value */
-} __packed;
-
-#define LOG_ENABLE		1  /* Enable logging */
-#define LOG_DISABLE		0  /* Disable logging */
-
-#define LOG_LEVEL_CRITICAL	1  /* (FDK fatal) */
-#define LOG_LEVEL_VERBOSE	2
-
-/*
- * Layout of a log fifo.
- */
-struct log_buffer_layout {
-	uint32_t read_ptr;  /*read pointer */
-	uint32_t write_ptr; /* write pointer */
-	uint32_t buffer[0]; /* buffer */
-} __packed;
-
-/*
- * Log buffer status reported by FW.
- */
-struct log_buffer_status {
-	uint32_t core_id;  /* ID of core that logged to other half */
-} __packed;
-
-#define TRACE_ID_LENGTH 12
-
-/*
- *  Log entry header.
- *
- * The header is followed by an array of arguments (uint32_t[]).
- * Number of arguments is specified by the params_num field of log_entry
- */
-struct log_entry_header {
-	uint32_t id_0 : TRACE_ID_LENGTH;	/* e.g. Pipeline ID */
-	uint32_t id_1 : TRACE_ID_LENGTH;	/* e.g. Component ID */
-	uint32_t core_id : 8;		/* Reporting core's id */
-
-	uint64_t timestamp;		/* Timestamp (in dsp ticks) */
-	uint32_t log_entry_address;	/* Address of log entry in ELF */
-} __packed;
-
-#endif

sound/hda/ext/hdac_ext_bus.c

@@ -85,7 +85,6 @@ int snd_hdac_ext_bus_init(struct hdac_bus *bus, struct device *dev,
 			const struct hdac_ext_bus_ops *ext_ops)
 {
 	int ret;
-	static int idx;
 
 	/* check if io ops are provided, if not load the defaults */
 	if (io_ops == NULL)
@@ -96,7 +95,12 @@ int snd_hdac_ext_bus_init(struct hdac_bus *bus, struct device *dev,
 		return ret;
 
 	bus->ext_ops = ext_ops;
-	bus->idx = idx++;
+	/* FIXME:
+	 * Currently only one bus is supported, if there is device with more
+	 * buses, bus->idx should be greater than 0, but there needs to be a
+	 * reliable way to always assign same number.
+	 */
+	bus->idx = 0;
 	bus->cmd_dma_state = true;
 
 	return 0;

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

@@ -298,69 +298,71 @@ static const struct snd_soc_ops cz_dmic1_cap_ops = {
 	.hw_params = cz_da7219_params,
 };
 
+SND_SOC_DAILINK_DEF(designware1,
+	DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.1.auto")));
+SND_SOC_DAILINK_DEF(designware2,
+	DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.2.auto")));
+SND_SOC_DAILINK_DEF(designware3,
+	DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.3.auto")));
+
+SND_SOC_DAILINK_DEF(dlgs,
+	DAILINK_COMP_ARRAY(COMP_CODEC("i2c-DLGS7219:00", "da7219-hifi")));
+SND_SOC_DAILINK_DEF(mx,
+	DAILINK_COMP_ARRAY(COMP_CODEC("MX98357A:00", "HiFi")));
+SND_SOC_DAILINK_DEF(adau,
+	DAILINK_COMP_ARRAY(COMP_CODEC("ADAU7002:00", "adau7002-hifi")));
+
+SND_SOC_DAILINK_DEF(platform,
+	DAILINK_COMP_ARRAY(COMP_PLATFORM("acp_audio_dma.0.auto")));
+
 static struct snd_soc_dai_link cz_dai_7219_98357[] = {
 	{
 		.name = "amd-da7219-play",
 		.stream_name = "Playback",
-		.platform_name = "acp_audio_dma.0.auto",
-		.cpu_dai_name = "designware-i2s.1.auto",
-		.codec_dai_name = "da7219-hifi",
-		.codec_name = "i2c-DLGS7219:00",
 		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
 				| SND_SOC_DAIFMT_CBM_CFM,
 		.init = cz_da7219_init,
 		.dpcm_playback = 1,
 		.ops = &cz_da7219_play_ops,
+		SND_SOC_DAILINK_REG(designware1, dlgs, platform),
 	},
 	{
 		.name = "amd-da7219-cap",
 		.stream_name = "Capture",
-		.platform_name = "acp_audio_dma.0.auto",
-		.cpu_dai_name = "designware-i2s.2.auto",
-		.codec_dai_name = "da7219-hifi",
-		.codec_name = "i2c-DLGS7219:00",
 		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
 				| SND_SOC_DAIFMT_CBM_CFM,
 		.dpcm_capture = 1,
 		.ops = &cz_da7219_cap_ops,
+		SND_SOC_DAILINK_REG(designware2, dlgs, platform),
 	},
 	{
 		.name = "amd-max98357-play",
 		.stream_name = "HiFi Playback",
-		.platform_name = "acp_audio_dma.0.auto",
-		.cpu_dai_name = "designware-i2s.3.auto",
-		.codec_dai_name = "HiFi",
-		.codec_name = "MX98357A:00",
 		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
 				| SND_SOC_DAIFMT_CBM_CFM,
 		.dpcm_playback = 1,
 		.ops = &cz_max_play_ops,
+		SND_SOC_DAILINK_REG(designware3, mx, platform),
 	},
 	{
 		/* C panel DMIC */
 		.name = "dmic0",
 		.stream_name = "DMIC0 Capture",
-		.platform_name = "acp_audio_dma.0.auto",
-		.cpu_dai_name = "designware-i2s.3.auto",
-		.codec_dai_name = "adau7002-hifi",
-		.codec_name = "ADAU7002:00",
 		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
 				| SND_SOC_DAIFMT_CBM_CFM,
 		.dpcm_capture = 1,
 		.ops = &cz_dmic0_cap_ops,
+		SND_SOC_DAILINK_REG(designware3, adau, platform),
 	},
 	{
 		/* A/B panel DMIC */
 		.name = "dmic1",
 		.stream_name = "DMIC1 Capture",
-		.platform_name = "acp_audio_dma.0.auto",
-		.cpu_dai_name = "designware-i2s.2.auto",
-		.codec_dai_name = "adau7002-hifi",
-		.codec_name = "ADAU7002:00",
 		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
 				| SND_SOC_DAIFMT_CBM_CFM,
 		.dpcm_capture = 1,
 		.ops = &cz_dmic1_cap_ops,
+		SND_SOC_DAILINK_REG(designware2, adau, platform),
 	},
 };
 

sound/soc/amd/acp-rt5645.c

@@ -95,29 +95,34 @@ static struct snd_soc_ops cz_aif1_ops = {
 	.hw_params = cz_aif1_hw_params,
 };
 
+SND_SOC_DAILINK_DEF(designware1,
+	DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.1.auto")));
+SND_SOC_DAILINK_DEF(designware2,
+	DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.2.auto")));
+
+SND_SOC_DAILINK_DEF(codec,
+	DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10EC5650:00", "rt5645-aif1")));
+
+SND_SOC_DAILINK_DEF(platform,
+	DAILINK_COMP_ARRAY(COMP_PLATFORM("acp_audio_dma.0.auto")));
+
 static struct snd_soc_dai_link cz_dai_rt5650[] = {
 	{
 		.name = "amd-rt5645-play",
 		.stream_name = "RT5645_AIF1",
-		.platform_name = "acp_audio_dma.0.auto",
-		.cpu_dai_name = "designware-i2s.1.auto",
-		.codec_dai_name = "rt5645-aif1",
-		.codec_name = "i2c-10EC5650:00",
 		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
 				| SND_SOC_DAIFMT_CBM_CFM,
 		.init = cz_init,
 		.ops = &cz_aif1_ops,
+		SND_SOC_DAILINK_REG(designware1, codec, platform),
 	},
 	{
 		.name = "amd-rt5645-cap",
 		.stream_name = "RT5645_AIF1",
-		.platform_name = "acp_audio_dma.0.auto",
-		.cpu_dai_name = "designware-i2s.2.auto",
-		.codec_dai_name = "rt5645-aif1",
-		.codec_name = "i2c-10EC5650:00",
 		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
 				| SND_SOC_DAIFMT_CBM_CFM,
 		.ops = &cz_aif1_ops,
+		SND_SOC_DAILINK_REG(designware2, codec, platform),
 	},
 };
 

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

@@ -32,6 +32,7 @@ struct i2s_stream_instance {
 	u16 channels;
 	u32 xfer_resolution;
 	struct page *pg;
+	u64 bytescount;
 	void __iomem *acp3x_base;
 };
 
@@ -317,6 +318,24 @@ static int acp3x_dma_open(struct snd_pcm_substream *substream)
 	return 0;
 }
 
+static u64 acp_get_byte_count(struct i2s_stream_instance *rtd, int direction)
+{
+	u64 byte_count;
+
+	if (direction == SNDRV_PCM_STREAM_PLAYBACK) {
+		byte_count = rv_readl(rtd->acp3x_base +
+				      mmACP_BT_TX_LINEARPOSITIONCNTR_HIGH);
+		byte_count |= rv_readl(rtd->acp3x_base +
+				       mmACP_BT_TX_LINEARPOSITIONCNTR_LOW);
+	} else {
+		byte_count = rv_readl(rtd->acp3x_base +
+				      mmACP_BT_RX_LINEARPOSITIONCNTR_HIGH);
+		byte_count |= rv_readl(rtd->acp3x_base +
+				       mmACP_BT_RX_LINEARPOSITIONCNTR_LOW);
+	}
+	return byte_count;
+}
+
 static int acp3x_dma_hw_params(struct snd_pcm_substream *substream,
 			       struct snd_pcm_hw_params *params)
 {
@@ -350,18 +369,17 @@ static int acp3x_dma_hw_params(struct snd_pcm_substream *substream,
 static snd_pcm_uframes_t acp3x_dma_pointer(struct snd_pcm_substream *substream)
 {
 	u32 pos = 0;
-	struct i2s_stream_instance *rtd = substream->runtime->private_data;
-
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		pos = rv_readl(rtd->acp3x_base +
-			       mmACP_BT_TX_LINKPOSITIONCNTR);
-	else
-		pos = rv_readl(rtd->acp3x_base +
-			       mmACP_BT_RX_LINKPOSITIONCNTR);
-
-	if (pos >= MAX_BUFFER)
-		pos = 0;
-
+	u32 buffersize = 0;
+	u64 bytescount = 0;
+	struct i2s_stream_instance *rtd =
+		substream->runtime->private_data;
+
+	buffersize = frames_to_bytes(substream->runtime,
+				     substream->runtime->buffer_size);
+	bytescount = acp_get_byte_count(rtd, substream->stream);
+	if (bytescount > rtd->bytescount)
+		bytescount -= rtd->bytescount;
+	pos = do_div(bytescount, buffersize);
 	return bytes_to_frames(substream->runtime, pos);
 }
 
@@ -521,6 +539,7 @@ static int acp3x_dai_i2s_trigger(struct snd_pcm_substream *substream,
 	case SNDRV_PCM_TRIGGER_START:
 	case SNDRV_PCM_TRIGGER_RESUME:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		rtd->bytescount = acp_get_byte_count(rtd, substream->stream);
 		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 			rv_writel(period_bytes, rtd->acp3x_base +
 				  mmACP_BT_TX_INTR_WATERMARK_SIZE);

sound/soc/atmel/atmel-classd.c

@@ -497,17 +497,30 @@ static int atmel_classd_asoc_card_init(struct device *dev,
 {
 	struct snd_soc_dai_link *dai_link;
 	struct atmel_classd *dd = snd_soc_card_get_drvdata(card);
+	struct snd_soc_dai_link_component *comp;
 
 	dai_link = devm_kzalloc(dev, sizeof(*dai_link), GFP_KERNEL);
 	if (!dai_link)
 		return -ENOMEM;
 
+	comp = devm_kzalloc(dev, 3 * sizeof(*comp), GFP_KERNEL);
+	if (!comp)
+		return -ENOMEM;
+
+	dai_link->cpus		= &comp[0];
+	dai_link->codecs	= &comp[1];
+	dai_link->platforms	= &comp[2];
+
+	dai_link->num_cpus	= 1;
+	dai_link->num_codecs	= 1;
+	dai_link->num_platforms	= 1;
+
 	dai_link->name			= "CLASSD";
 	dai_link->stream_name		= "CLASSD PCM";
-	dai_link->codec_dai_name	= ATMEL_CLASSD_CODEC_DAI_NAME;
-	dai_link->cpu_dai_name		= dev_name(dev);
-	dai_link->codec_name		= dev_name(dev);
-	dai_link->platform_name		= dev_name(dev);
+	dai_link->codecs->dai_name	= ATMEL_CLASSD_CODEC_DAI_NAME;
+	dai_link->cpus->dai_name	= dev_name(dev);
+	dai_link->codecs->name		= dev_name(dev);
+	dai_link->platforms->name	= dev_name(dev);
 
 	card->dai_link	= dai_link;
 	card->num_links	= 1;

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

@@ -111,16 +111,11 @@ static const struct snd_dmaengine_pcm_config atmel_dmaengine_pcm_config = {
 
 int atmel_pcm_dma_platform_register(struct device *dev)
 {
-	return snd_dmaengine_pcm_register(dev, &atmel_dmaengine_pcm_config, 0);
+	return devm_snd_dmaengine_pcm_register(dev,
+					&atmel_dmaengine_pcm_config, 0);
 }
 EXPORT_SYMBOL(atmel_pcm_dma_platform_register);
 
-void atmel_pcm_dma_platform_unregister(struct device *dev)
-{
-	snd_dmaengine_pcm_unregister(dev);
-}
-EXPORT_SYMBOL(atmel_pcm_dma_platform_unregister);
-
 MODULE_AUTHOR("Bo Shen <voice.shen@atmel.com>");
 MODULE_DESCRIPTION("Atmel DMA based PCM module");
 MODULE_LICENSE("GPL");

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

@@ -393,11 +393,6 @@ int atmel_pcm_pdc_platform_register(struct device *dev)
 }
 EXPORT_SYMBOL(atmel_pcm_pdc_platform_register);
 
-void atmel_pcm_pdc_platform_unregister(struct device *dev)
-{
-}
-EXPORT_SYMBOL(atmel_pcm_pdc_platform_unregister);
-
 MODULE_AUTHOR("Sedji Gaouaou <sedji.gaouaou@atmel.com>");
 MODULE_DESCRIPTION("Atmel PCM module");
 MODULE_LICENSE("GPL");

sound/soc/atmel/atmel-pcm.h

@@ -72,28 +72,20 @@ struct atmel_pcm_dma_params {
 
 #if IS_ENABLED(CONFIG_SND_ATMEL_SOC_PDC)
 int atmel_pcm_pdc_platform_register(struct device *dev);
-void atmel_pcm_pdc_platform_unregister(struct device *dev);
 #else
 static inline int atmel_pcm_pdc_platform_register(struct device *dev)
 {
 	return 0;
 }
-static inline void atmel_pcm_pdc_platform_unregister(struct device *dev)
-{
-}
 #endif
 
 #if IS_ENABLED(CONFIG_SND_ATMEL_SOC_DMA)
 int atmel_pcm_dma_platform_register(struct device *dev);
-void atmel_pcm_dma_platform_unregister(struct device *dev);
 #else
 static inline int atmel_pcm_dma_platform_register(struct device *dev)
 {
 	return 0;
 }
-static inline void atmel_pcm_dma_platform_unregister(struct device *dev)
-{
-}
 #endif
 
 #endif /* _ATMEL_PCM_H */

sound/soc/atmel/atmel-pdmic.c

@@ -508,17 +508,30 @@ static int atmel_pdmic_asoc_card_init(struct device *dev,
 {
 	struct snd_soc_dai_link *dai_link;
 	struct atmel_pdmic *dd = snd_soc_card_get_drvdata(card);
+	struct snd_soc_dai_link_component *comp;
 
 	dai_link = devm_kzalloc(dev, sizeof(*dai_link), GFP_KERNEL);
 	if (!dai_link)
 		return -ENOMEM;
 
+	comp = devm_kzalloc(dev, 3 * sizeof(*comp), GFP_KERNEL);
+	if (!comp)
+		return -ENOMEM;
+
+	dai_link->cpus		= &comp[0];
+	dai_link->codecs	= &comp[1];
+	dai_link->platforms	= &comp[2];
+
+	dai_link->num_cpus	= 1;
+	dai_link->num_codecs	= 1;
+	dai_link->num_platforms	= 1;
+
 	dai_link->name			= "PDMIC";
 	dai_link->stream_name		= "PDMIC PCM";
-	dai_link->codec_dai_name	= ATMEL_PDMIC_CODEC_DAI_NAME;
-	dai_link->cpu_dai_name		= dev_name(dev);
-	dai_link->codec_name		= dev_name(dev);
-	dai_link->platform_name		= dev_name(dev);
+	dai_link->codecs->dai_name	= ATMEL_PDMIC_CODEC_DAI_NAME;
+	dai_link->cpus->dai_name	= dev_name(dev);
+	dai_link->codecs->name		= dev_name(dev);
+	dai_link->platforms->name	= dev_name(dev);
 
 	card->dai_link	= dai_link;
 	card->num_links	= 1;

sound/soc/atmel/atmel_ssc_dai.c

@@ -1012,16 +1012,6 @@ static int asoc_ssc_init(struct device *dev)
 	return 0;
 }
 
-static void asoc_ssc_exit(struct device *dev)
-{
-	struct ssc_device *ssc = dev_get_drvdata(dev);
-
-	if (ssc->pdata->use_dma)
-		atmel_pcm_dma_platform_unregister(dev);
-	else
-		atmel_pcm_pdc_platform_unregister(dev);
-}
-
 /**
  * atmel_ssc_set_audio - Allocate the specified SSC for audio use.
  */
@@ -1050,7 +1040,6 @@ void atmel_ssc_put_audio(int ssc_id)
 {
 	struct ssc_device *ssc = ssc_info[ssc_id].ssc;
 
-	asoc_ssc_exit(&ssc->pdev->dev);
 	ssc_free(ssc);
 }
 EXPORT_SYMBOL_GPL(atmel_ssc_put_audio);

sound/soc/atmel/atmel_wm8904.c

@@ -56,14 +56,19 @@ static const struct snd_soc_ops atmel_asoc_wm8904_ops = {
 	.hw_params = atmel_asoc_wm8904_hw_params,
 };
 
+SND_SOC_DAILINK_DEFS(pcm,
+	DAILINK_COMP_ARRAY(COMP_EMPTY()),
+	DAILINK_COMP_ARRAY(COMP_CODEC(NULL, "wm8904-hifi")),
+	DAILINK_COMP_ARRAY(COMP_EMPTY()));
+
 static struct snd_soc_dai_link atmel_asoc_wm8904_dailink = {
 	.name = "WM8904",
 	.stream_name = "WM8904 PCM",
-	.codec_dai_name = "wm8904-hifi",
 	.dai_fmt = SND_SOC_DAIFMT_I2S
 		| SND_SOC_DAIFMT_NB_NF
 		| SND_SOC_DAIFMT_CBM_CFM,
 	.ops = &atmel_asoc_wm8904_ops,
+	SND_SOC_DAILINK_REG(pcm),
 };
 
 static struct snd_soc_card atmel_asoc_wm8904_card = {
@@ -107,8 +112,8 @@ static int atmel_asoc_wm8904_dt_init(struct platform_device *pdev)
 		ret = -EINVAL;
 		return ret;
 	}
-	dailink->cpu_of_node = cpu_np;
-	dailink->platform_of_node = cpu_np;
+	dailink->cpus->of_node = cpu_np;
+	dailink->platforms->of_node = cpu_np;
 	of_node_put(cpu_np);
 
 	codec_np = of_parse_phandle(np, "atmel,audio-codec", 0);
@@ -117,7 +122,7 @@ static int atmel_asoc_wm8904_dt_init(struct platform_device *pdev)
 		ret = -EINVAL;
 		return ret;
 	}
-	dailink->codec_of_node = codec_np;
+	dailink->codecs->of_node = codec_np;
 	of_node_put(codec_np);
 
 	return 0;
@@ -136,7 +141,7 @@ static int atmel_asoc_wm8904_probe(struct platform_device *pdev)
 		return ret;
 	}
 
-	id = of_alias_get_id((struct device_node *)dailink->cpu_of_node, "ssc");
+	id = of_alias_get_id((struct device_node *)dailink->cpus->of_node, "ssc");
 	ret = atmel_ssc_set_audio(id);
 	if (ret != 0) {
 		dev_err(&pdev->dev, "failed to set SSC %d for audio\n", id);
@@ -162,7 +167,7 @@ static int atmel_asoc_wm8904_remove(struct platform_device *pdev)
 	struct snd_soc_dai_link *dailink = &atmel_asoc_wm8904_dailink;
 	int id;
 
-	id = of_alias_get_id((struct device_node *)dailink->cpu_of_node, "ssc");
+	id = of_alias_get_id((struct device_node *)dailink->cpus->of_node, "ssc");
 
 	snd_soc_unregister_card(card);
 	atmel_ssc_put_audio(id);

sound/soc/atmel/mikroe-proto.c

@@ -63,6 +63,7 @@ static struct snd_soc_card snd_proto = {
 static int snd_proto_probe(struct platform_device *pdev)
 {
 	struct snd_soc_dai_link *dai;
+	struct snd_soc_dai_link_component *comp;
 	struct device_node *np = pdev->dev.of_node;
 	struct device_node *codec_np, *cpu_np;
 	struct device_node *bitclkmaster = NULL;
@@ -84,12 +85,24 @@ static int snd_proto_probe(struct platform_device *pdev)
 	if (!dai)
 		return -ENOMEM;
 
+	/* for cpus/codecs/platforms */
+	comp = devm_kzalloc(&pdev->dev, 3 * sizeof(*comp), GFP_KERNEL);
+	if (!comp)
+		return -ENOMEM;
+
 	snd_proto.dai_link = dai;
 	snd_proto.num_links = 1;
 
+	dai->cpus = &comp[0];
+	dai->num_cpus = 1;
+	dai->codecs = &comp[1];
+	dai->num_codecs = 1;
+	dai->platforms = &comp[2];
+	dai->num_platforms = 1;
+
 	dai->name = "WM8731";
 	dai->stream_name = "WM8731 HiFi";
-	dai->codec_dai_name = "wm8731-hifi";
+	dai->codecs->dai_name = "wm8731-hifi";
 	dai->init = &snd_proto_init;
 
 	codec_np = of_parse_phandle(np, "audio-codec", 0);
@@ -97,15 +110,15 @@ static int snd_proto_probe(struct platform_device *pdev)
 		dev_err(&pdev->dev, "audio-codec node missing\n");
 		return -EINVAL;
 	}
-	dai->codec_of_node = codec_np;
+	dai->codecs->of_node = codec_np;
 
 	cpu_np = of_parse_phandle(np, "i2s-controller", 0);
 	if (!cpu_np) {
 		dev_err(&pdev->dev, "i2s-controller missing\n");
 		return -EINVAL;
 	}
-	dai->cpu_of_node = cpu_np;
-	dai->platform_of_node = cpu_np;
+	dai->cpus->of_node = cpu_np;
+	dai->platforms->of_node = cpu_np;
 
 	dai_fmt = snd_soc_of_parse_daifmt(np, NULL,
 					  &bitclkmaster, &framemaster);

sound/soc/atmel/sam9g20_wm8731.c

@@ -116,16 +116,18 @@ static int at91sam9g20ek_wm8731_init(struct snd_soc_pcm_runtime *rtd)
 	return 0;
 }
 
+SND_SOC_DAILINK_DEFS(pcm,
+	DAILINK_COMP_ARRAY(COMP_CPU("at91rm9200_ssc.0")),
+	DAILINK_COMP_ARRAY(COMP_CODEC("wm8731.0-001b", "wm8731-hifi")),
+	DAILINK_COMP_ARRAY(COMP_PLATFORM("at91rm9200_ssc.0")));
+
 static struct snd_soc_dai_link at91sam9g20ek_dai = {
 	.name = "WM8731",
 	.stream_name = "WM8731 PCM",
-	.cpu_dai_name = "at91rm9200_ssc.0",
-	.codec_dai_name = "wm8731-hifi",
 	.init = at91sam9g20ek_wm8731_init,
-	.platform_name = "at91rm9200_ssc.0",
-	.codec_name = "wm8731.0-001b",
 	.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
 		   SND_SOC_DAIFMT_CBM_CFM,
+	SND_SOC_DAILINK_REG(pcm),
 };
 
 static struct snd_soc_card snd_soc_at91sam9g20ek = {
@@ -198,24 +200,24 @@ static int at91sam9g20ek_audio_probe(struct platform_device *pdev)
 		goto err;
 
 	/* Parse codec info */
-	at91sam9g20ek_dai.codec_name = NULL;
+	at91sam9g20ek_dai.codecs->name = NULL;
 	codec_np = of_parse_phandle(np, "atmel,audio-codec", 0);
 	if (!codec_np) {
 		dev_err(&pdev->dev, "codec info missing\n");
 		return -EINVAL;
 	}
-	at91sam9g20ek_dai.codec_of_node = codec_np;
+	at91sam9g20ek_dai.codecs->of_node = codec_np;
 
 	/* Parse dai and platform info */
-	at91sam9g20ek_dai.cpu_dai_name = NULL;
-	at91sam9g20ek_dai.platform_name = NULL;
+	at91sam9g20ek_dai.cpus->dai_name = NULL;
+	at91sam9g20ek_dai.platforms->name = NULL;
 	cpu_np = of_parse_phandle(np, "atmel,ssc-controller", 0);
 	if (!cpu_np) {
 		dev_err(&pdev->dev, "dai and pcm info missing\n");
 		return -EINVAL;
 	}
-	at91sam9g20ek_dai.cpu_of_node = cpu_np;
-	at91sam9g20ek_dai.platform_of_node = cpu_np;
+	at91sam9g20ek_dai.cpus->of_node = cpu_np;
+	at91sam9g20ek_dai.platforms->of_node = cpu_np;
 
 	of_node_put(codec_np);
 	of_node_put(cpu_np);

sound/soc/atmel/sam9x5_wm8731.c

@@ -77,6 +77,7 @@ static int sam9x5_wm8731_driver_probe(struct platform_device *pdev)
 	struct snd_soc_card *card;
 	struct snd_soc_dai_link *dai;
 	struct sam9x5_drvdata *priv;
+	struct snd_soc_dai_link_component *comp;
 	int ret;
 
 	if (!np) {
@@ -87,7 +88,8 @@ static int sam9x5_wm8731_driver_probe(struct platform_device *pdev)
 	card = devm_kzalloc(&pdev->dev, sizeof(*card), GFP_KERNEL);
 	priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL);
 	dai = devm_kzalloc(&pdev->dev, sizeof(*dai), GFP_KERNEL);
-	if (!dai || !card || !priv) {
+	comp = devm_kzalloc(&pdev->dev, 3 * sizeof(*comp), GFP_KERNEL);
+	if (!dai || !card || !priv || !comp) {
 		ret = -ENOMEM;
 		goto out;
 	}
@@ -100,9 +102,17 @@ static int sam9x5_wm8731_driver_probe(struct platform_device *pdev)
 	card->num_links = 1;
 	card->dapm_widgets = sam9x5_dapm_widgets;
 	card->num_dapm_widgets = ARRAY_SIZE(sam9x5_dapm_widgets);
+
+	dai->cpus = &comp[0];
+	dai->num_cpus = 1;
+	dai->codecs = &comp[1];
+	dai->num_codecs = 1;
+	dai->platforms = &comp[2];
+	dai->num_platforms = 1;
+
 	dai->name = "WM8731";
 	dai->stream_name = "WM8731 PCM";
-	dai->codec_dai_name = "wm8731-hifi";
+	dai->codecs->dai_name = "wm8731-hifi";
 	dai->init = sam9x5_wm8731_init;
 	dai->dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF
 		| SND_SOC_DAIFMT_CBM_CFM;
@@ -126,7 +136,7 @@ static int sam9x5_wm8731_driver_probe(struct platform_device *pdev)
 		goto out;
 	}
 
-	dai->codec_of_node = codec_np;
+	dai->codecs->of_node = codec_np;
 
 	cpu_np = of_parse_phandle(np, "atmel,ssc-controller", 0);
 	if (!cpu_np) {
@@ -134,8 +144,8 @@ static int sam9x5_wm8731_driver_probe(struct platform_device *pdev)
 		ret = -EINVAL;
 		goto out;
 	}
-	dai->cpu_of_node = cpu_np;
-	dai->platform_of_node = cpu_np;
+	dai->cpus->of_node = cpu_np;
+	dai->platforms->of_node = cpu_np;
 
 	priv->ssc_id = of_alias_get_id(cpu_np, "ssc");
 

sound/soc/atmel/tse850-pcm5142.c

@@ -294,13 +294,18 @@ static const struct snd_soc_dapm_route tse850_intercon[] = {
 	{ "DAC", NULL, "OUTL" },
 };
 
+SND_SOC_DAILINK_DEFS(pcm,
+	DAILINK_COMP_ARRAY(COMP_EMPTY()),
+	DAILINK_COMP_ARRAY(COMP_CODEC(NULL, "pcm512x-hifi")),
+	DAILINK_COMP_ARRAY(COMP_EMPTY()));
+
 static struct snd_soc_dai_link tse850_dailink = {
 	.name = "TSE-850",
 	.stream_name = "TSE-850-PCM",
-	.codec_dai_name = "pcm512x-hifi",
 	.dai_fmt = SND_SOC_DAIFMT_I2S
 		 | SND_SOC_DAIFMT_NB_NF
 		 | SND_SOC_DAIFMT_CBM_CFS,
+	SND_SOC_DAILINK_REG(pcm),
 };
 
 static struct snd_soc_card tse850_card = {
@@ -331,8 +336,8 @@ static int tse850_dt_init(struct platform_device *pdev)
 		dev_err(&pdev->dev, "failed to get cpu dai\n");
 		return -EINVAL;
 	}
-	dailink->cpu_of_node = cpu_np;
-	dailink->platform_of_node = cpu_np;
+	dailink->cpus->of_node = cpu_np;
+	dailink->platforms->of_node = cpu_np;
 	of_node_put(cpu_np);
 
 	codec_np = of_parse_phandle(np, "axentia,audio-codec", 0);
@@ -340,7 +345,7 @@ static int tse850_dt_init(struct platform_device *pdev)
 		dev_err(&pdev->dev, "failed to get codec info\n");
 		return -EINVAL;
 	}
-	dailink->codec_of_node = codec_np;
+	dailink->codecs->of_node = codec_np;
 	of_node_put(codec_np);
 
 	return 0;

sound/soc/au1x/db1000.c

@@ -19,13 +19,15 @@
 
 #include "psc.h"
 
+SND_SOC_DAILINK_DEFS(hifi,
+	DAILINK_COMP_ARRAY(COMP_CPU("alchemy-ac97c")),
+	DAILINK_COMP_ARRAY(COMP_CODEC("ac97-codec", "ac97-hifi")),
+	DAILINK_COMP_ARRAY(COMP_PLATFORM("alchemy-pcm-dma.0")));
+
 static struct snd_soc_dai_link db1000_ac97_dai = {
 	.name		= "AC97",
 	.stream_name	= "AC97 HiFi",
-	.codec_dai_name	= "ac97-hifi",
-	.cpu_dai_name	= "alchemy-ac97c",
-	.platform_name	= "alchemy-pcm-dma.0",
-	.codec_name	= "ac97-codec",
+	SND_SOC_DAILINK_REG(hifi),
 };
 
 static struct snd_soc_card db1000_ac97 = {

sound/soc/au1x/db1200.c

@@ -47,13 +47,15 @@ static const struct platform_device_id db1200_pids[] = {
 
 /*-------------------------  AC97 PART  ---------------------------*/
 
+SND_SOC_DAILINK_DEFS(db1200_ac97,
+	DAILINK_COMP_ARRAY(COMP_CPU("au1xpsc_ac97.1")),
+	DAILINK_COMP_ARRAY(COMP_CODEC("ac97-codec.1", "ac97-hifi")),
+	DAILINK_COMP_ARRAY(COMP_PLATFORM("au1xpsc-pcm.1")));
+
 static struct snd_soc_dai_link db1200_ac97_dai = {
 	.name		= "AC97",
 	.stream_name	= "AC97 HiFi",
-	.codec_dai_name	= "ac97-hifi",
-	.cpu_dai_name	= "au1xpsc_ac97.1",
-	.platform_name	= "au1xpsc-pcm.1",
-	.codec_name	= "ac97-codec.1",
+	SND_SOC_DAILINK_REG(db1200_ac97),
 };
 
 static struct snd_soc_card db1200_ac97_machine = {
@@ -63,13 +65,15 @@ static struct snd_soc_card db1200_ac97_machine = {
 	.num_links	= 1,
 };
 
+SND_SOC_DAILINK_DEFS(db1300_ac97,
+	DAILINK_COMP_ARRAY(COMP_CPU("au1xpsc_ac97.1")),
+	DAILINK_COMP_ARRAY(COMP_CODEC("wm9712-codec.1", "wm9712-hifi")),
+	DAILINK_COMP_ARRAY(COMP_PLATFORM("au1xpsc-pcm.1")));
+
 static struct snd_soc_dai_link db1300_ac97_dai = {
 	.name		= "AC97",
 	.stream_name	= "AC97 HiFi",
-	.codec_dai_name	= "wm9712-hifi",
-	.cpu_dai_name	= "au1xpsc_ac97.1",
-	.platform_name	= "au1xpsc-pcm.1",
-	.codec_name	= "wm9712-codec.1",
+	SND_SOC_DAILINK_REG(db1300_ac97),
 };
 
 static struct snd_soc_card db1300_ac97_machine = {
@@ -104,16 +108,18 @@ static const struct snd_soc_ops db1200_i2s_wm8731_ops = {
 	.startup	= db1200_i2s_startup,
 };
 
+SND_SOC_DAILINK_DEFS(db1200_i2s,
+	DAILINK_COMP_ARRAY(COMP_CPU("au1xpsc_i2s.1")),
+	DAILINK_COMP_ARRAY(COMP_CODEC("wm8731.0-001b", "wm8731-hifi")),
+	DAILINK_COMP_ARRAY(COMP_PLATFORM("au1xpsc-pcm.1")));
+
 static struct snd_soc_dai_link db1200_i2s_dai = {
 	.name		= "WM8731",
 	.stream_name	= "WM8731 PCM",
-	.codec_dai_name	= "wm8731-hifi",
-	.cpu_dai_name	= "au1xpsc_i2s.1",
-	.platform_name	= "au1xpsc-pcm.1",
-	.codec_name	= "wm8731.0-001b",
 	.dai_fmt	= SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF |
 			  SND_SOC_DAIFMT_CBM_CFM,
 	.ops		= &db1200_i2s_wm8731_ops,
+	SND_SOC_DAILINK_REG(db1200_i2s),
 };
 
 static struct snd_soc_card db1200_i2s_machine = {
@@ -123,16 +129,18 @@ static struct snd_soc_card db1200_i2s_machine = {
 	.num_links	= 1,
 };
 
+SND_SOC_DAILINK_DEFS(db1300_i2s,
+	DAILINK_COMP_ARRAY(COMP_CPU("au1xpsc_i2s.2")),
+	DAILINK_COMP_ARRAY(COMP_CODEC("wm8731.0-001b", "wm8731-hifi")),
+	DAILINK_COMP_ARRAY(COMP_PLATFORM("au1xpsc-pcm.2")));
+
 static struct snd_soc_dai_link db1300_i2s_dai = {
 	.name		= "WM8731",
 	.stream_name	= "WM8731 PCM",
-	.codec_dai_name	= "wm8731-hifi",
-	.cpu_dai_name	= "au1xpsc_i2s.2",
-	.platform_name	= "au1xpsc-pcm.2",
-	.codec_name	= "wm8731.0-001b",
 	.dai_fmt	= SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF |
 			  SND_SOC_DAIFMT_CBM_CFM,
 	.ops		= &db1200_i2s_wm8731_ops,
+	SND_SOC_DAILINK_REG(db1300_i2s),
 };
 
 static struct snd_soc_card db1300_i2s_machine = {
@@ -142,16 +150,18 @@ static struct snd_soc_card db1300_i2s_machine = {
 	.num_links	= 1,
 };
 
+SND_SOC_DAILINK_DEFS(db1550_i2s,
+	DAILINK_COMP_ARRAY(COMP_CPU("au1xpsc_i2s.3")),
+	DAILINK_COMP_ARRAY(COMP_CODEC("wm8731.0-001b", "wm8731-hifi")),
+	DAILINK_COMP_ARRAY(COMP_PLATFORM("au1xpsc-pcm.3")));
+
 static struct snd_soc_dai_link db1550_i2s_dai = {
 	.name		= "WM8731",
 	.stream_name	= "WM8731 PCM",
-	.codec_dai_name	= "wm8731-hifi",
-	.cpu_dai_name	= "au1xpsc_i2s.3",
-	.platform_name	= "au1xpsc-pcm.3",
-	.codec_name	= "wm8731.0-001b",
 	.dai_fmt	= SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF |
 			  SND_SOC_DAIFMT_CBM_CFM,
 	.ops		= &db1200_i2s_wm8731_ops,
+	SND_SOC_DAILINK_REG(db1550_i2s),
 };
 
 static struct snd_soc_card db1550_i2s_machine = {

sound/soc/au1x/psc-i2s.c

@@ -340,16 +340,14 @@ static int au1xpsc_i2s_drvprobe(struct platform_device *pdev)
 
 	platform_set_drvdata(pdev, wd);
 
-	return snd_soc_register_component(&pdev->dev, &au1xpsc_i2s_component,
-					  &wd->dai_drv, 1);
+	return devm_snd_soc_register_component(&pdev->dev,
+				&au1xpsc_i2s_component, &wd->dai_drv, 1);
 }
 
 static int au1xpsc_i2s_drvremove(struct platform_device *pdev)
 {
 	struct au1xpsc_audio_data *wd = platform_get_drvdata(pdev);
 
-	snd_soc_unregister_component(&pdev->dev);
-
 	__raw_writel(0, I2S_CFG(wd));
 	wmb(); /* drain writebuffer */
 	__raw_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd));

sound/soc/cirrus/edb93xx.c

@@ -51,16 +51,18 @@ static const struct snd_soc_ops edb93xx_ops = {
 	.hw_params	= edb93xx_hw_params,
 };
 
+SND_SOC_DAILINK_DEFS(hifi,
+	DAILINK_COMP_ARRAY(COMP_CPU("ep93xx-i2s")),
+	DAILINK_COMP_ARRAY(COMP_CODEC("spi0.0", "cs4271-hifi")),
+	DAILINK_COMP_ARRAY(COMP_PLATFORM("ep93xx-i2s")));
+
 static struct snd_soc_dai_link edb93xx_dai = {
 	.name		= "CS4271",
 	.stream_name	= "CS4271 HiFi",
-	.platform_name	= "ep93xx-i2s",
-	.cpu_dai_name	= "ep93xx-i2s",
-	.codec_name	= "spi0.0",
-	.codec_dai_name	= "cs4271-hifi",
 	.dai_fmt	= SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
 			  SND_SOC_DAIFMT_CBS_CFS,
 	.ops		= &edb93xx_ops,
+	SND_SOC_DAILINK_REG(hifi),
 };
 
 static struct snd_soc_card snd_soc_edb93xx = {

sound/soc/cirrus/ep93xx-i2s.c

@@ -473,19 +473,17 @@ static int ep93xx_i2s_probe(struct platform_device *pdev)
 
 	dev_set_drvdata(&pdev->dev, info);
 
-	err = snd_soc_register_component(&pdev->dev, &ep93xx_i2s_component,
+	err = devm_snd_soc_register_component(&pdev->dev, &ep93xx_i2s_component,
 					 &ep93xx_i2s_dai, 1);
 	if (err)
 		goto fail_put_lrclk;
 
 	err = devm_ep93xx_pcm_platform_register(&pdev->dev);
 	if (err)
-		goto fail_unregister;
+		goto fail_put_lrclk;
 
 	return 0;
 
-fail_unregister:
-	snd_soc_unregister_component(&pdev->dev);
 fail_put_lrclk:
 	clk_put(info->lrclk);
 fail_put_sclk:
@@ -500,7 +498,6 @@ static int ep93xx_i2s_remove(struct platform_device *pdev)
 {
 	struct ep93xx_i2s_info *info = dev_get_drvdata(&pdev->dev);
 
-	snd_soc_unregister_component(&pdev->dev);
 	clk_put(info->lrclk);
 	clk_put(info->sclk);
 	clk_put(info->mclk);

sound/soc/cirrus/simone.c

@@ -18,13 +18,15 @@
 
 #include <asm/mach-types.h>
 
+SND_SOC_DAILINK_DEFS(hifi,
+	DAILINK_COMP_ARRAY(COMP_CPU("ep93xx-ac97")),
+	DAILINK_COMP_ARRAY(COMP_CODEC("ac97-codec", "ac97-hifi")),
+	DAILINK_COMP_ARRAY(COMP_PLATFORM("ep93xx-ac97")));
+
 static struct snd_soc_dai_link simone_dai = {
 	.name		= "AC97",
 	.stream_name	= "AC97 HiFi",
-	.cpu_dai_name	= "ep93xx-ac97",
-	.codec_dai_name	= "ac97-hifi",
-	.codec_name	= "ac97-codec",
-	.platform_name	= "ep93xx-ac97",
+	SND_SOC_DAILINK_REG(hifi),
 };
 
 static struct snd_soc_card snd_soc_simone = {

sound/soc/cirrus/snappercl15.c

@@ -60,16 +60,19 @@ static const struct snd_soc_dapm_route audio_map[] = {
 	{"MICIN", NULL, "Mic Jack"},
 };
 
+SND_SOC_DAILINK_DEFS(aic23,
+	DAILINK_COMP_ARRAY(COMP_CPU("ep93xx-i2s")),
+	DAILINK_COMP_ARRAY(COMP_CODEC("tlv320aic23-codec.0-001a",
+				      "tlv320aic23-hifi")),
+	DAILINK_COMP_ARRAY(COMP_PLATFORM("ep93xx-i2s")));
+
 static struct snd_soc_dai_link snappercl15_dai = {
 	.name		= "tlv320aic23",
 	.stream_name	= "AIC23",
-	.cpu_dai_name	= "ep93xx-i2s",
-	.codec_dai_name	= "tlv320aic23-hifi",
-	.codec_name	= "tlv320aic23-codec.0-001a",
-	.platform_name	= "ep93xx-i2s",
 	.dai_fmt	= SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
 			  SND_SOC_DAIFMT_CBS_CFS,
 	.ops		= &snappercl15_ops,
+	SND_SOC_DAILINK_REG(aic23),
 };
 
 static struct snd_soc_card snd_soc_snappercl15 = {

sound/soc/codecs/Kconfig

@@ -71,8 +71,12 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_CS4341 if SND_SOC_I2C_AND_SPI
 	select SND_SOC_CS4349 if I2C
 	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_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
@@ -140,7 +144,9 @@ config SND_SOC_ALL_CODECS
 	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_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
@@ -283,10 +289,12 @@ config SND_SOC_WM_HUBS
 config SND_SOC_WM_ADSP
 	tristate
 	select SND_SOC_COMPRESS
+	default y if SND_SOC_MADERA=y
 	default y if SND_SOC_CS47L24=y
 	default y if SND_SOC_WM5102=y
 	default y if SND_SOC_WM5110=y
 	default y if SND_SOC_WM2200=y
+	default m if SND_SOC_MADERA=m
 	default m if SND_SOC_CS47L24=m
 	default m if SND_SOC_WM5102=m
 	default m if SND_SOC_WM5110=m
@@ -576,6 +584,15 @@ config SND_SOC_CS4349
 config SND_SOC_CS47L24
 	tristate
 
+config SND_SOC_CS47L35
+	tristate
+
+config SND_SOC_CS47L85
+	tristate
+
+config SND_SOC_CS47L90
+	tristate
+
 # Cirrus Logic Quad-Channel ADC
 config SND_SOC_CS53L30
 	tristate "Cirrus Logic CS53L30 CODEC"
@@ -585,6 +602,12 @@ config SND_SOC_CX20442
 	tristate
 	depends on TTY
 
+config SND_SOC_CX2072X
+	tristate "Conexant CX2072X CODEC"
+	depends on I2C
+	help
+	  Enable support for Conexant CX20721 and CX20723 codec chips.
+
 config SND_SOC_JZ4740_CODEC
 	depends on MIPS || COMPILE_TEST
 	select REGMAP_MMIO
@@ -697,6 +720,15 @@ config SND_SOC_LOCHNAGAR_SC
 	  This driver support the sound card functionality of the Cirrus
 	  Logic Lochnagar audio development board.
 
+config SND_SOC_MADERA
+	tristate
+	default y if SND_SOC_CS47L35=y
+	default y if SND_SOC_CS47L85=y
+	default y if SND_SOC_CS47L90=y
+	default m if SND_SOC_CS47L35=m
+	default m if SND_SOC_CS47L85=m
+	default m if SND_SOC_CS47L90=m
+
 config SND_SOC_MAX98088
 	tristate "Maxim MAX98088/9 Low-Power, Stereo Audio Codec"
 	depends on I2C
@@ -708,7 +740,8 @@ config SND_SOC_MAX98095
        tristate
 
 config SND_SOC_MAX98357A
-       tristate
+	tristate "Maxim MAX98357A CODEC"
+	depends on GPIOLIB
 
 config SND_SOC_MAX98371
        tristate
@@ -870,7 +903,9 @@ config SND_SOC_RL6231
 	default y if SND_SOC_RT5670=y
 	default y if SND_SOC_RT5677=y
 	default y if SND_SOC_RT5682=y
+	default y if SND_SOC_RT1011=y
 	default y if SND_SOC_RT1305=y
+	default y if SND_SOC_RT1308=y
 	default m if SND_SOC_RT5514=m
 	default m if SND_SOC_RT5616=m
 	default m if SND_SOC_RT5640=m
@@ -884,7 +919,9 @@ config SND_SOC_RL6231
 	default m if SND_SOC_RT5670=m
 	default m if SND_SOC_RT5677=m
 	default m if SND_SOC_RT5682=m
+	default m if SND_SOC_RT1011=m
 	default m if SND_SOC_RT1305=m
+	default m if SND_SOC_RT1308=m
 
 config SND_SOC_RL6347A
 	tristate
@@ -907,9 +944,15 @@ config SND_SOC_RT298
 	tristate
 	depends on I2C
 
+config SND_SOC_RT1011
+	tristate
+
 config SND_SOC_RT1305
 	tristate
 
+config SND_SOC_RT1308
+	tristate
+
 config SND_SOC_RT5514
 	tristate
 

sound/soc/codecs/Makefile

@@ -65,8 +65,12 @@ snd-soc-cs43130-objs := cs43130.o
 snd-soc-cs4341-objs := cs4341.o
 snd-soc-cs4349-objs := cs4349.o
 snd-soc-cs47l24-objs := cs47l24.o
+snd-soc-cs47l35-objs := cs47l35.o
+snd-soc-cs47l85-objs := cs47l85.o
+snd-soc-cs47l90-objs := cs47l90.o
 snd-soc-cs53l30-objs := cs53l30.o
 snd-soc-cx20442-objs := cx20442.o
+snd-soc-cx2072x-objs := cx2072x.o
 snd-soc-da7210-objs := da7210.o
 snd-soc-da7213-objs := da7213.o
 snd-soc-da7218-objs := da7218.o
@@ -92,6 +96,7 @@ snd-soc-l3-objs := l3.o
 snd-soc-lm4857-objs := lm4857.o
 snd-soc-lm49453-objs := lm49453.o
 snd-soc-lochnagar-sc-objs := lochnagar-sc.o
+snd-soc-madera-objs := madera.o
 snd-soc-max9759-objs := max9759.o
 snd-soc-max9768-objs := max9768.o
 snd-soc-max98088-objs := max98088.o
@@ -141,7 +146,9 @@ snd-soc-pcm512x-spi-objs := pcm512x-spi.o
 snd-soc-rk3328-objs := rk3328_codec.o
 snd-soc-rl6231-objs := rl6231.o
 snd-soc-rl6347a-objs := rl6347a.o
+snd-soc-rt1011-objs := rt1011.o
 snd-soc-rt1305-objs := rt1305.o
+snd-soc-rt1308-objs := rt1308.o
 snd-soc-rt274-objs := rt274.o
 snd-soc-rt286-objs := rt286.o
 snd-soc-rt298-objs := rt298.o
@@ -339,8 +346,12 @@ obj-$(CONFIG_SND_SOC_CS43130)   += snd-soc-cs43130.o
 obj-$(CONFIG_SND_SOC_CS4341)	+= snd-soc-cs4341.o
 obj-$(CONFIG_SND_SOC_CS4349)	+= snd-soc-cs4349.o
 obj-$(CONFIG_SND_SOC_CS47L24)	+= snd-soc-cs47l24.o
+obj-$(CONFIG_SND_SOC_CS47L35)	+= snd-soc-cs47l35.o
+obj-$(CONFIG_SND_SOC_CS47L85)	+= snd-soc-cs47l85.o
+obj-$(CONFIG_SND_SOC_CS47L90)	+= snd-soc-cs47l90.o
 obj-$(CONFIG_SND_SOC_CS53L30)	+= snd-soc-cs53l30.o
 obj-$(CONFIG_SND_SOC_CX20442)	+= snd-soc-cx20442.o
+obj-$(CONFIG_SND_SOC_CX2072X)	+= snd-soc-cx2072x.o
 obj-$(CONFIG_SND_SOC_DA7210)	+= snd-soc-da7210.o
 obj-$(CONFIG_SND_SOC_DA7213)	+= snd-soc-da7213.o
 obj-$(CONFIG_SND_SOC_DA7218)	+= snd-soc-da7218.o
@@ -366,6 +377,7 @@ obj-$(CONFIG_SND_SOC_L3)	+= snd-soc-l3.o
 obj-$(CONFIG_SND_SOC_LM4857)	+= snd-soc-lm4857.o
 obj-$(CONFIG_SND_SOC_LM49453)   += snd-soc-lm49453.o
 obj-$(CONFIG_SND_SOC_LOCHNAGAR_SC)	+= snd-soc-lochnagar-sc.o
+obj-$(CONFIG_SND_SOC_MADERA)	+= snd-soc-madera.o
 obj-$(CONFIG_SND_SOC_MAX9759)	+= snd-soc-max9759.o
 obj-$(CONFIG_SND_SOC_MAX9768)	+= snd-soc-max9768.o
 obj-$(CONFIG_SND_SOC_MAX98088)	+= snd-soc-max98088.o
@@ -415,7 +427,9 @@ obj-$(CONFIG_SND_SOC_PCM512x_SPI)	+= snd-soc-pcm512x-spi.o
 obj-$(CONFIG_SND_SOC_RK3328)	+= snd-soc-rk3328.o
 obj-$(CONFIG_SND_SOC_RL6231)	+= snd-soc-rl6231.o
 obj-$(CONFIG_SND_SOC_RL6347A)	+= snd-soc-rl6347a.o
+obj-$(CONFIG_SND_SOC_RT1011)	+= snd-soc-rt1011.o
 obj-$(CONFIG_SND_SOC_RT1305)	+= snd-soc-rt1305.o
+obj-$(CONFIG_SND_SOC_RT1308)	+= snd-soc-rt1308.o
 obj-$(CONFIG_SND_SOC_RT274)	+= snd-soc-rt274.o
 obj-$(CONFIG_SND_SOC_RT286)	+= snd-soc-rt286.o
 obj-$(CONFIG_SND_SOC_RT298)	+= snd-soc-rt298.o

sound/soc/codecs/ad193x.c

@@ -413,6 +413,48 @@ static struct snd_soc_dai_driver ad193x_no_adc_dai = {
 	.ops = &ad193x_dai_ops,
 };
 
+struct ad193x_reg_default {
+	unsigned int reg;
+	unsigned int val;
+};
+
+/* codec register values to set after reset */
+static void ad193x_reg_default_init(struct ad193x_priv *ad193x)
+{
+	const struct ad193x_reg_default reg_init[] = {
+		{  0, 0x99 },	/* PLL_CLK_CTRL0: pll input: mclki/xi 12.288Mhz */
+		{  1, 0x04 },	/* PLL_CLK_CTRL1: no on-chip Vref */
+		{  2, 0x40 },	/* DAC_CTRL0: TDM mode */
+		{  3, 0x00 },	/* DAC_CTRL1: reset */
+		{  4, 0x1A },	/* DAC_CTRL2: 48kHz de-emphasis, unmute dac */
+		{  5, 0x00 },	/* DAC_CHNL_MUTE: unmute DAC channels */
+		{  6, 0x00 },	/* DAC_L1_VOL: no attenuation */
+		{  7, 0x00 },	/* DAC_R1_VOL: no attenuation */
+		{  8, 0x00 },	/* DAC_L2_VOL: no attenuation */
+		{  9, 0x00 },	/* DAC_R2_VOL: no attenuation */
+		{ 10, 0x00 },	/* DAC_L3_VOL: no attenuation */
+		{ 11, 0x00 },	/* DAC_R3_VOL: no attenuation */
+		{ 12, 0x00 },	/* DAC_L4_VOL: no attenuation */
+		{ 13, 0x00 },	/* DAC_R4_VOL: no attenuation */
+	};
+	const struct ad193x_reg_default reg_adc_init[] = {
+		{ 14, 0x03 },	/* ADC_CTRL0: high-pass filter enable */
+		{ 15, 0x43 },	/* ADC_CTRL1: sata delay=1, adc aux mode */
+		{ 16, 0x00 },	/* ADC_CTRL2: reset */
+	};
+	int i;
+
+	for (i = 0; i < ARRAY_SIZE(reg_init); i++)
+		regmap_write(ad193x->regmap, reg_init[i].reg, reg_init[i].val);
+
+	if (ad193x_has_adc(ad193x)) {
+		for (i = 0; i < ARRAY_SIZE(reg_adc_init); i++) {
+			regmap_write(ad193x->regmap, reg_adc_init[i].reg,
+				     reg_adc_init[i].val);
+		}
+	}
+}
+
 static int ad193x_component_probe(struct snd_soc_component *component)
 {
 	struct ad193x_priv *ad193x = snd_soc_component_get_drvdata(component);
@@ -420,25 +462,7 @@ static int ad193x_component_probe(struct snd_soc_component *component)
 	int num, ret;
 
 	/* default setting for ad193x */
-
-	/* unmute dac channels */
-	regmap_write(ad193x->regmap, AD193X_DAC_CHNL_MUTE, 0x0);
-	/* de-emphasis: 48kHz, powedown dac */
-	regmap_write(ad193x->regmap, AD193X_DAC_CTRL2, 0x1A);
-	/* dac in tdm mode */
-	regmap_write(ad193x->regmap, AD193X_DAC_CTRL0, 0x40);
-
-	/* adc only */
-	if (ad193x_has_adc(ad193x)) {
-		/* high-pass filter enable */
-		regmap_write(ad193x->regmap, AD193X_ADC_CTRL0, 0x3);
-		/* sata delay=1, adc aux mode */
-		regmap_write(ad193x->regmap, AD193X_ADC_CTRL1, 0x43);
-	}
-
-	/* pll input: mclki/xi */
-	regmap_write(ad193x->regmap, AD193X_PLL_CLK_CTRL0, 0x99); /* mclk=24.576Mhz: 0x9D; mclk=12.288Mhz: 0x99 */
-	regmap_write(ad193x->regmap, AD193X_PLL_CLK_CTRL1, 0x04);
+	ad193x_reg_default_init(ad193x);
 
 	/* adc only */
 	if (ad193x_has_adc(ad193x)) {

sound/soc/codecs/ak4118.c

@@ -400,14 +400,8 @@ static int ak4118_i2c_probe(struct i2c_client *i2c,
 		return ret;
 	}
 
-	return snd_soc_register_component(&i2c->dev, &soc_component_drv_ak4118,
-					  &ak4118_dai, 1);
-}
-
-static int ak4118_i2c_remove(struct i2c_client *i2c)
-{
-	snd_soc_unregister_component(&i2c->dev);
-	return 0;
+	return devm_snd_soc_register_component(&i2c->dev,
+				&soc_component_drv_ak4118, &ak4118_dai, 1);
 }
 
 static const struct of_device_id ak4118_of_match[] = {
@@ -429,7 +423,6 @@ static struct i2c_driver ak4118_i2c_driver = {
 	},
 	.id_table = ak4118_id_table,
 	.probe  = ak4118_i2c_probe,
-	.remove = ak4118_i2c_remove,
 };
 
 module_i2c_driver(ak4118_i2c_driver);

sound/soc/codecs/cros_ec_codec.c

@@ -413,7 +413,7 @@ static int cros_ec_codec_platform_probe(struct platform_device *pd)
 
 	platform_set_drvdata(pd, codec_data);
 
-	return snd_soc_register_component(dev, &cros_ec_component_driver,
+	return devm_snd_soc_register_component(dev, &cros_ec_component_driver,
 					  cros_ec_dai, ARRAY_SIZE(cros_ec_dai));
 }
 

sound/soc/codecs/cs42xx8.c

@@ -14,6 +14,7 @@
 #include <linux/delay.h>
 #include <linux/module.h>
 #include <linux/of_device.h>
+#include <linux/gpio/consumer.h>
 #include <linux/pm_runtime.h>
 #include <linux/regulator/consumer.h>
 #include <sound/pcm_params.h>
@@ -45,6 +46,7 @@ struct cs42xx8_priv {
 	bool slave_mode;
 	unsigned long sysclk;
 	u32 tx_channels;
+	struct gpio_desc *gpiod_reset;
 };
 
 /* -127.5dB to 0dB with step of 0.5dB */
@@ -467,6 +469,13 @@ int cs42xx8_probe(struct device *dev, struct regmap *regmap)
 		return -EINVAL;
 	}
 
+	cs42xx8->gpiod_reset = devm_gpiod_get_optional(dev, "reset",
+							GPIOD_OUT_HIGH);
+	if (IS_ERR(cs42xx8->gpiod_reset))
+		return PTR_ERR(cs42xx8->gpiod_reset);
+
+	gpiod_set_value_cansleep(cs42xx8->gpiod_reset, 0);
+
 	cs42xx8->clk = devm_clk_get(dev, "mclk");
 	if (IS_ERR(cs42xx8->clk)) {
 		dev_err(dev, "failed to get the clock: %ld\n",
@@ -547,6 +556,8 @@ static int cs42xx8_runtime_resume(struct device *dev)
 		return ret;
 	}
 
+	gpiod_set_value_cansleep(cs42xx8->gpiod_reset, 0);
+
 	ret = regulator_bulk_enable(ARRAY_SIZE(cs42xx8->supplies),
 				    cs42xx8->supplies);
 	if (ret) {
@@ -586,6 +597,8 @@ static int cs42xx8_runtime_suspend(struct device *dev)
 	regulator_bulk_disable(ARRAY_SIZE(cs42xx8->supplies),
 			       cs42xx8->supplies);
 
+	gpiod_set_value_cansleep(cs42xx8->gpiod_reset, 1);
+
 	clk_disable_unprepare(cs42xx8->clk);
 
 	return 0;

sound/soc/codecs/cx2072x.h

+/* SPDX-License-Identifier: GPL-2.0 */
+/*
+ * ALSA SoC CX20721/CX20723 codec driver
+ *
+ * Copyright:	(C) 2017 Conexant Systems, Inc.
+ * Author:	Simon Ho, <Simon.ho@conexant.com>
+ */
+
+#ifndef __CX2072X_H__
+#define __CX2072X_H__
+
+#define CX2072X_MCLK_PLL		1
+#define CX2072X_MCLK_EXTERNAL_PLL	1
+#define CX2072X_MCLK_INTERNAL_OSC	2
+
+/*#define CX2072X_RATES		SNDRV_PCM_RATE_8000_192000*/
+#define CX2072X_RATES_DSP	SNDRV_PCM_RATE_48000
+
+#define CX2072X_REG_MAX					0x8a3c
+
+#define CX2072X_VENDOR_ID				0x0200
+#define CX2072X_REVISION_ID				0x0208
+#define CX2072X_CURRENT_BCLK_FREQUENCY			0x00dc
+#define CX2072X_AFG_POWER_STATE				0x0414
+#define CX2072X_UM_RESPONSE				0x0420
+#define CX2072X_GPIO_DATA				0x0454
+#define CX2072X_GPIO_ENABLE				0x0458
+#define CX2072X_GPIO_DIRECTION				0x045c
+#define CX2072X_GPIO_WAKE				0x0460
+#define CX2072X_GPIO_UM_ENABLE				0x0464
+#define CX2072X_GPIO_STICKY_MASK			0x0468
+#define CX2072X_AFG_FUNCTION_RESET			0x07fc
+#define CX2072X_DAC1_CONVERTER_FORMAT			0x43c8
+#define CX2072X_DAC1_AMP_GAIN_RIGHT			0x41c0
+#define CX2072X_DAC1_AMP_GAIN_LEFT			0x41e0
+#define CX2072X_DAC1_POWER_STATE			0x4014
+#define CX2072X_DAC1_CONVERTER_STREAM_CHANNEL		0x4018
+#define CX2072X_DAC1_EAPD_ENABLE			0x4030
+#define CX2072X_DAC2_CONVERTER_FORMAT			0x47c8
+#define CX2072X_DAC2_AMP_GAIN_RIGHT			0x45c0
+#define CX2072X_DAC2_AMP_GAIN_LEFT			0x45e0
+#define CX2072X_DAC2_POWER_STATE			0x4414
+#define CX2072X_DAC2_CONVERTER_STREAM_CHANNEL		0x4418
+#define CX2072X_ADC1_CONVERTER_FORMAT			0x4fc8
+#define CX2072X_ADC1_AMP_GAIN_RIGHT_0			0x4d80
+#define CX2072X_ADC1_AMP_GAIN_LEFT_0			0x4da0
+#define CX2072X_ADC1_AMP_GAIN_RIGHT_1			0x4d84
+#define CX2072X_ADC1_AMP_GAIN_LEFT_1			0x4da4
+#define CX2072X_ADC1_AMP_GAIN_RIGHT_2			0x4d88
+#define CX2072X_ADC1_AMP_GAIN_LEFT_2			0x4da8
+#define CX2072X_ADC1_AMP_GAIN_RIGHT_3			0x4d8c
+#define CX2072X_ADC1_AMP_GAIN_LEFT_3			0x4dac
+#define CX2072X_ADC1_AMP_GAIN_RIGHT_4			0x4d90
+#define CX2072X_ADC1_AMP_GAIN_LEFT_4			0x4db0
+#define CX2072X_ADC1_AMP_GAIN_RIGHT_5			0x4d94
+#define CX2072X_ADC1_AMP_GAIN_LEFT_5			0x4db4
+#define CX2072X_ADC1_AMP_GAIN_RIGHT_6			0x4d98
+#define CX2072X_ADC1_AMP_GAIN_LEFT_6			0x4db8
+#define CX2072X_ADC1_CONNECTION_SELECT_CONTROL		0x4c04
+#define CX2072X_ADC1_POWER_STATE			0x4c14
+#define CX2072X_ADC1_CONVERTER_STREAM_CHANNEL		0x4c18
+#define CX2072X_ADC2_CONVERTER_FORMAT			0x53c8
+#define CX2072X_ADC2_AMP_GAIN_RIGHT_0			0x5180
+#define CX2072X_ADC2_AMP_GAIN_LEFT_0			0x51a0
+#define CX2072X_ADC2_AMP_GAIN_RIGHT_1			0x5184
+#define CX2072X_ADC2_AMP_GAIN_LEFT_1			0x51a4
+#define CX2072X_ADC2_AMP_GAIN_RIGHT_2			0x5188
+#define CX2072X_ADC2_AMP_GAIN_LEFT_2			0x51a8
+#define CX2072X_ADC2_CONNECTION_SELECT_CONTROL		0x5004
+#define CX2072X_ADC2_POWER_STATE			0x5014
+#define CX2072X_ADC2_CONVERTER_STREAM_CHANNEL		0x5018
+#define CX2072X_PORTA_CONNECTION_SELECT_CTRL		0x5804
+#define CX2072X_PORTA_POWER_STATE			0x5814
+#define CX2072X_PORTA_PIN_CTRL				0x581c
+#define CX2072X_PORTA_UNSOLICITED_RESPONSE		0x5820
+#define CX2072X_PORTA_PIN_SENSE				0x5824
+#define CX2072X_PORTA_EAPD_BTL				0x5830
+#define CX2072X_PORTB_POWER_STATE			0x6014
+#define CX2072X_PORTB_PIN_CTRL				0x601c
+#define CX2072X_PORTB_UNSOLICITED_RESPONSE		0x6020
+#define CX2072X_PORTB_PIN_SENSE				0x6024
+#define CX2072X_PORTB_EAPD_BTL				0x6030
+#define CX2072X_PORTB_GAIN_RIGHT			0x6180
+#define CX2072X_PORTB_GAIN_LEFT				0x61a0
+#define CX2072X_PORTC_POWER_STATE			0x6814
+#define CX2072X_PORTC_PIN_CTRL				0x681c
+#define CX2072X_PORTC_GAIN_RIGHT			0x6980
+#define CX2072X_PORTC_GAIN_LEFT				0x69a0
+#define CX2072X_PORTD_POWER_STATE			0x6414
+#define CX2072X_PORTD_PIN_CTRL				0x641c
+#define CX2072X_PORTD_UNSOLICITED_RESPONSE		0x6420
+#define CX2072X_PORTD_PIN_SENSE				0x6424
+#define CX2072X_PORTD_GAIN_RIGHT			0x6580
+#define CX2072X_PORTD_GAIN_LEFT				0x65a0
+#define CX2072X_PORTE_CONNECTION_SELECT_CTRL		0x7404
+#define CX2072X_PORTE_POWER_STATE			0x7414
+#define CX2072X_PORTE_PIN_CTRL				0x741c
+#define CX2072X_PORTE_UNSOLICITED_RESPONSE		0x7420
+#define CX2072X_PORTE_PIN_SENSE				0x7424
+#define CX2072X_PORTE_EAPD_BTL				0x7430
+#define CX2072X_PORTE_GAIN_RIGHT			0x7580
+#define CX2072X_PORTE_GAIN_LEFT				0x75a0
+#define CX2072X_PORTF_POWER_STATE			0x7814
+#define CX2072X_PORTF_PIN_CTRL				0x781c
+#define CX2072X_PORTF_UNSOLICITED_RESPONSE		0x7820
+#define CX2072X_PORTF_PIN_SENSE				0x7824
+#define CX2072X_PORTF_GAIN_RIGHT			0x7980
+#define CX2072X_PORTF_GAIN_LEFT				0x79a0
+#define CX2072X_PORTG_POWER_STATE			0x5c14
+#define CX2072X_PORTG_PIN_CTRL				0x5c1c
+#define CX2072X_PORTG_CONNECTION_SELECT_CTRL		0x5c04
+#define CX2072X_PORTG_EAPD_BTL				0x5c30
+#define CX2072X_PORTM_POWER_STATE			0x8814
+#define CX2072X_PORTM_PIN_CTRL				0x881c
+#define CX2072X_PORTM_CONNECTION_SELECT_CTRL		0x8804
+#define CX2072X_PORTM_EAPD_BTL				0x8830
+#define CX2072X_MIXER_POWER_STATE			0x5414
+#define CX2072X_MIXER_GAIN_RIGHT_0			0x5580
+#define CX2072X_MIXER_GAIN_LEFT_0			0x55a0
+#define CX2072X_MIXER_GAIN_RIGHT_1			0x5584
+#define CX2072X_MIXER_GAIN_LEFT_1			0x55a4
+#define CX2072X_EQ_ENABLE_BYPASS			0x6d00
+#define CX2072X_EQ_B0_COEFF				0x6d02
+#define CX2072X_EQ_B1_COEFF				0x6d04
+#define CX2072X_EQ_B2_COEFF				0x6d06
+#define CX2072X_EQ_A1_COEFF				0x6d08
+#define CX2072X_EQ_A2_COEFF				0x6d0a
+#define CX2072X_EQ_G_COEFF				0x6d0c
+#define CX2072X_EQ_BAND					0x6d0d
+#define CX2072X_SPKR_DRC_ENABLE_STEP			0x6d10
+#define CX2072X_SPKR_DRC_CONTROL			0x6d14
+#define CX2072X_SPKR_DRC_TEST				0x6d18
+#define CX2072X_DIGITAL_BIOS_TEST0			0x6d80
+#define CX2072X_DIGITAL_BIOS_TEST2			0x6d84
+#define CX2072X_I2SPCM_CONTROL1				0x6e00
+#define CX2072X_I2SPCM_CONTROL2				0x6e04
+#define CX2072X_I2SPCM_CONTROL3				0x6e08
+#define CX2072X_I2SPCM_CONTROL4				0x6e0c
+#define CX2072X_I2SPCM_CONTROL5				0x6e10
+#define CX2072X_I2SPCM_CONTROL6				0x6e18
+#define CX2072X_UM_INTERRUPT_CRTL_E			0x6e14
+#define CX2072X_CODEC_TEST2				0x7108
+#define CX2072X_CODEC_TEST9				0x7124
+#define CX2072X_CODEC_TESTXX				0x7290
+#define CX2072X_CODEC_TEST20				0x7310
+#define CX2072X_CODEC_TEST24				0x731c
+#define CX2072X_CODEC_TEST26				0x7328
+#define CX2072X_ANALOG_TEST3				0x718c
+#define CX2072X_ANALOG_TEST4				0x7190
+#define CX2072X_ANALOG_TEST5				0x7194
+#define CX2072X_ANALOG_TEST6				0x7198
+#define CX2072X_ANALOG_TEST7				0x719c
+#define CX2072X_ANALOG_TEST8				0x71a0
+#define CX2072X_ANALOG_TEST9				0x71a4
+#define CX2072X_ANALOG_TEST10				0x71a8
+#define CX2072X_ANALOG_TEST11				0x71ac
+#define CX2072X_ANALOG_TEST12				0x71b0
+#define CX2072X_ANALOG_TEST13				0x71b4
+#define CX2072X_DIGITAL_TEST0				0x7200
+#define CX2072X_DIGITAL_TEST1				0x7204
+#define CX2072X_DIGITAL_TEST11				0x722c
+#define CX2072X_DIGITAL_TEST12				0x7230
+#define CX2072X_DIGITAL_TEST15				0x723c
+#define CX2072X_DIGITAL_TEST16				0x7080
+#define CX2072X_DIGITAL_TEST17				0x7084
+#define CX2072X_DIGITAL_TEST18				0x7088
+#define CX2072X_DIGITAL_TEST19				0x708c
+#define CX2072X_DIGITAL_TEST20				0x7090
+
+/* not used in the current code, for future extensions (if any) */
+#define CX2072X_MAX_EQ_BAND		7
+#define CX2072X_MAX_EQ_COEFF		11
+#define CX2072X_MAX_DRC_REGS		9
+#define CX2072X_MIC_EQ_COEFF		10
+#define CX2072X_PLBK_EQ_BAND_NUM	7
+#define CX2072X_PLBK_EQ_COEF_LEN	11
+#define CX2072X_PLBK_DRC_PARM_LEN	9
+#define CX2072X_CLASSD_AMP_LEN		6
+
+/* DAI interfae type */
+#define CX2072X_DAI_HIFI	1
+#define CX2072X_DAI_DSP		2
+#define CX2072X_DAI_DSP_PWM	3 /* 4 ch, including mic and AEC */
+
+enum cx2072x_reg_sample_size {
+	CX2072X_SAMPLE_SIZE_8_BITS = 0,
+	CX2072X_SAMPLE_SIZE_16_BITS = 1,
+	CX2072X_SAMPLE_SIZE_24_BITS = 2,
+	CX2072X_SAMPLE_SIZE_RESERVED = 3,
+};
+
+union cx2072x_reg_i2spcm_ctrl_reg1 {
+	struct {
+		u32 rx_data_one_line:1;
+		u32 rx_ws_pol:1;
+		u32 rx_ws_wid:7;
+		u32 rx_frm_len:5;
+		u32 rx_sa_size:2;
+		u32 tx_data_one_line:1;
+		u32 tx_ws_pol:1;
+		u32 tx_ws_wid:7;
+		u32 tx_frm_len:5;
+		u32 tx_sa_size:2;
+	} r;
+	u32 ulval;
+};
+
+union cx2072x_reg_i2spcm_ctrl_reg2 {
+	struct {
+		u32 tx_en_ch1:1;
+		u32 tx_en_ch2:1;
+		u32 tx_en_ch3:1;
+		u32 tx_en_ch4:1;
+		u32 tx_en_ch5:1;
+		u32 tx_en_ch6:1;
+		u32 tx_slot_1:5;
+		u32 tx_slot_2:5;
+		u32 tx_slot_3:5;
+		u32 tx_slot_4:5;
+		u32 res:1;
+		u32 tx_data_neg_bclk:1;
+		u32 tx_master:1;
+		u32 tx_tri_n:1;
+		u32 tx_endian_sel:1;
+		u32 tx_dstart_dly:1;
+	} r;
+	u32 ulval;
+};
+
+union cx2072x_reg_i2spcm_ctrl_reg3 {
+	struct {
+		u32 rx_en_ch1:1;
+		u32 rx_en_ch2:1;
+		u32 rx_en_ch3:1;
+		u32 rx_en_ch4:1;
+		u32 rx_en_ch5:1;
+		u32 rx_en_ch6:1;
+		u32 rx_slot_1:5;
+		u32 rx_slot_2:5;
+		u32 rx_slot_3:5;
+		u32 rx_slot_4:5;
+		u32 res:1;
+		u32 rx_data_neg_bclk:1;
+		u32 rx_master:1;
+		u32 rx_tri_n:1;
+		u32 rx_endian_sel:1;
+		u32 rx_dstart_dly:1;
+	} r;
+	u32 ulval;
+};
+
+union cx2072x_reg_i2spcm_ctrl_reg4 {
+	struct {
+		u32 rx_mute:1;
+		u32 tx_mute:1;
+		u32 reserved:1;
+		u32 dac_34_independent:1;
+		u32 dac_bclk_lrck_share:1;
+		u32 bclk_lrck_share_en:1;
+		u32 reserved2:2;
+		u32 rx_last_dac_ch_en:1;
+		u32 rx_last_dac_ch:3;
+		u32 tx_last_adc_ch_en:1;
+		u32 tx_last_adc_ch:3;
+		u32 rx_slot_5:5;
+		u32 rx_slot_6:5;
+		u32 reserved3:6;
+	} r;
+	u32 ulval;
+};
+
+union cx2072x_reg_i2spcm_ctrl_reg5 {
+	struct {
+		u32 tx_slot_5:5;
+		u32 reserved:3;
+		u32 tx_slot_6:5;
+		u32 reserved2:3;
+		u32 reserved3:8;
+		u32 i2s_pcm_clk_div:7;
+		u32 i2s_pcm_clk_div_chan_en:1;
+	} r;
+	u32 ulval;
+};
+
+union cx2072x_reg_i2spcm_ctrl_reg6 {
+	struct {
+		u32 reserved:5;
+		u32 rx_pause_cycles:3;
+		u32 rx_pause_start_pos:8;
+		u32 reserved2:5;
+		u32 tx_pause_cycles:3;
+		u32 tx_pause_start_pos:8;
+	} r;
+	u32 ulval;
+};
+
+union cx2072x_reg_digital_bios_test2 {
+	struct {
+		u32 pull_down_eapd:2;
+		u32 input_en_eapd_pad:1;
+		u32 push_pull_mode:1;
+		u32 eapd_pad_output_driver:2;
+		u32 pll_source:1;
+		u32 i2s_bclk_en:1;
+		u32 i2s_bclk_invert:1;
+		u32 pll_ref_clock:1;
+		u32 class_d_shield_clk:1;
+		u32 audio_pll_bypass_mode:1;
+		u32 reserved:4;
+	} r;
+	u32 ulval;
+};
+
+#endif /* __CX2072X_H__ */

sound/soc/codecs/hdac_hdmi.c

@@ -1890,6 +1890,12 @@ static void hdmi_codec_remove(struct snd_soc_component *component)
 {
 	struct hdac_hdmi_priv *hdmi = snd_soc_component_get_drvdata(component);
 	struct hdac_device *hdev = hdmi->hdev;
+	int ret;
+
+	ret = snd_hdac_acomp_register_notifier(hdev->bus, NULL);
+	if (ret < 0)
+		dev_err(&hdev->dev, "notifier unregister failed: err: %d\n",
+				ret);
 
 	pm_runtime_disable(&hdev->dev);
 }
@@ -2113,6 +2119,7 @@ static int hdac_hdmi_runtime_suspend(struct device *dev)
 		return -EIO;
 	}
 
+	snd_hdac_codec_link_down(hdev);
 	snd_hdac_ext_bus_link_put(bus, hlink);
 
 	snd_hdac_display_power(bus, hdev->addr, false);
@@ -2139,6 +2146,7 @@ static int hdac_hdmi_runtime_resume(struct device *dev)
 	}
 
 	snd_hdac_ext_bus_link_get(bus, hlink);
+	snd_hdac_codec_link_up(hdev);
 
 	snd_hdac_display_power(bus, hdev->addr, true);
 

sound/soc/codecs/nau8822.c

@@ -828,6 +828,24 @@ static int nau8822_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_component *component = dai->component;
 	struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component);
 	int val_len = 0, val_rate = 0;
+	unsigned int ctrl_val, bclk_fs, bclk_div;
+
+	/* make BCLK and LRC divide configuration if the codec as master. */
+	snd_soc_component_read(component, NAU8822_REG_CLOCKING, &ctrl_val);
+	if (ctrl_val & NAU8822_CLK_MASTER) {
+		/* get the bclk and fs ratio */
+		bclk_fs = snd_soc_params_to_bclk(params) / params_rate(params);
+		if (bclk_fs <= 32)
+			bclk_div = NAU8822_BCLKDIV_8;
+		else if (bclk_fs <= 64)
+			bclk_div = NAU8822_BCLKDIV_4;
+		else if (bclk_fs <= 128)
+			bclk_div = NAU8822_BCLKDIV_2;
+		else
+			return -EINVAL;
+		snd_soc_component_update_bits(component, NAU8822_REG_CLOCKING,
+				NAU8822_BCLKSEL_MASK, bclk_div);
+	}
 
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S16_LE:

sound/soc/codecs/nau8822.h

@@ -107,10 +107,17 @@
 
 /* NAU8822_REG_CLOCKING (0x6) */
 #define NAU8822_CLKIOEN_MASK			0x1
+#define NAU8822_CLK_MASTER			0x1
+#define NAU8822_CLK_SLAVE			0x0
 #define NAU8822_MCLKSEL_SFT			5
 #define NAU8822_MCLKSEL_MASK			(0x7 << 5)
 #define NAU8822_BCLKSEL_SFT			2
 #define NAU8822_BCLKSEL_MASK			(0x7 << 2)
+#define NAU8822_BCLKDIV_1			(0x0 << 2)
+#define NAU8822_BCLKDIV_2			(0x1 << 2)
+#define NAU8822_BCLKDIV_4			(0x2 << 2)
+#define NAU8822_BCLKDIV_8			(0x3 << 2)
+#define NAU8822_BCLKDIV_16			(0x4 << 2)
 #define NAU8822_CLKM_MASK			(0x1 << 8)
 #define NAU8822_CLKM_MCLK			(0x0 << 8)
 #define NAU8822_CLKM_PLL			(0x1 << 8)

sound/soc/codecs/rt5514-spi.c

@@ -470,9 +470,7 @@ static int __maybe_unused rt5514_suspend(struct device *dev)
 
 static int __maybe_unused rt5514_resume(struct device *dev)
 {
-	struct snd_soc_component *component = snd_soc_lookup_component(dev, DRV_NAME);
-	struct rt5514_dsp *rt5514_dsp =
-		snd_soc_component_get_drvdata(component);
+	struct rt5514_dsp *rt5514_dsp = dev_get_drvdata(dev);
 	int irq = to_spi_device(dev)->irq;
 	u8 buf[8];
 

sound/soc/codecs/rt5665.c

@@ -1478,7 +1478,7 @@ static int set_dmic_clk(struct snd_soc_dapm_widget *w,
 {
 	struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm);
 	struct rt5665_priv *rt5665 = snd_soc_component_get_drvdata(component);
-	int pd, idx = -EINVAL;
+	int pd, idx;
 
 	pd = rl6231_get_pre_div(rt5665->regmap,
 		RT5665_ADDA_CLK_1, RT5665_I2S_PD1_SFT);

sound/soc/codecs/rt5677-spi.c

@@ -26,6 +26,8 @@
 
 #include "rt5677-spi.h"
 
+#define DRV_NAME "rt5677spi"
+
 #define RT5677_SPI_BURST_LEN	240
 #define RT5677_SPI_HEADER	5
 #define RT5677_SPI_FREQ		6000000
@@ -230,7 +232,7 @@ MODULE_DEVICE_TABLE(acpi, rt5677_spi_acpi_id);
 
 static struct spi_driver rt5677_spi_driver = {
 	.driver = {
-		.name = "rt5677",
+		.name = DRV_NAME,
 		.acpi_match_table = ACPI_PTR(rt5677_spi_acpi_id),
 	},
 	.probe = rt5677_spi_probe,

sound/soc/codecs/rt5682.c

@@ -2662,15 +2662,9 @@ static int rt5682_i2c_probe(struct i2c_client *i2c,
 
 	}
 
-	return snd_soc_register_component(&i2c->dev, &soc_component_dev_rt5682,
-			rt5682_dai, ARRAY_SIZE(rt5682_dai));
-}
-
-static int rt5682_i2c_remove(struct i2c_client *i2c)
-{
-	snd_soc_unregister_component(&i2c->dev);
-
-	return 0;
+	return devm_snd_soc_register_component(&i2c->dev,
+					&soc_component_dev_rt5682,
+					rt5682_dai, ARRAY_SIZE(rt5682_dai));
 }
 
 static void rt5682_i2c_shutdown(struct i2c_client *client)
@@ -2703,7 +2697,6 @@ static struct i2c_driver rt5682_i2c_driver = {
 		.acpi_match_table = ACPI_PTR(rt5682_acpi_match),
 	},
 	.probe = rt5682_i2c_probe,
-	.remove = rt5682_i2c_remove,
 	.shutdown = rt5682_i2c_shutdown,
 	.id_table = rt5682_i2c_id,
 };

sound/soc/codecs/tas571x.c

@@ -721,8 +721,8 @@ static const struct regmap_config tas5721_regmap_config = {
 static const struct tas571x_chip tas5721_chip = {
 	.supply_names			= tas5721_supply_names,
 	.num_supply_names		= ARRAY_SIZE(tas5721_supply_names),
-	.controls			= tas5711_controls,
-	.num_controls			= ARRAY_SIZE(tas5711_controls),
+	.controls			= tas5721_controls,
+	.num_controls			= ARRAY_SIZE(tas5721_controls),
 	.regmap_config			= &tas5721_regmap_config,
 	.vol_reg_size			= 1,
 };