Merge tag 'sound-5.8-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "It was another busy development cycle, and the majority of changes are
  found in ASoC side. Below are Some highlights.

  ASoC core:
   - Lots of core cleanups and refactorings, still on-going work by
     Morimoto-san

  ASoC drivers:
   - Continued work on cleaning up and improving the Intel SOF stuff,
     along with new platform support including SoundWire

   - Fixes to make the Marvell SSPA driver work upstream

   - Support for AMD Renoir ACP, Dialog DA7212, Freescale EASRC and
     i.MX8M, Intel Elkhard Lake, Maxim MAX98390, Nuvoton NAU8812 and
     NAU8814 and Realtek RT1016.

  USB-audio:
   - Improvement for sync and implicit feedback streams with the more
     accurate frame size calculation and full-duplex support

   - Support for RME Babyface Pro and Prioneer DJ DJM

  HD-audio:
   - Fixes for Mic mute LED on HP machines

   - Re-enable support of Intel SST driver for SKL/KBL platforms

  FireWire:
   - Lots of refactoring, add support for RME FireFace and MOTU
     UltraLite-mk3"

* tag 'sound-5.8-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (428 commits)
  ALSA: es1688: Add the missed snd_card_free()
  ALSA: hda: add sienna_cichlid audio asic id for sienna_cichlid up
  ALSA: usb-audio: Add Pioneer DJ DJM-900NXS2 support
  ASoC: qcom: q6asm-dai: kCFI fix
  ASoC: soc-card: add snd_soc_card_remove_dai_link()
  ASoC: soc-card: add snd_soc_card_add_dai_link()
  ASoC: soc-card: add snd_soc_card_set_bias_level_post()
  ASoC: soc-card: add snd_soc_card_set_bias_level()
  ASoC: soc-card: add snd_soc_card_remove()
  ASoC: soc-card: add snd_soc_card_late_probe()
  ASoC: soc-card: add snd_soc_card_probe()
  ASoC: soc-card: add probed bit field to snd_soc_card
  ASoC: soc-card: add snd_soc_card_resume_post()
  ASoC: soc-card: add snd_soc_card_resume_pre()
  ASoC: soc-card: add snd_soc_card_suspend_post()
  ASoC: soc-card: add snd_soc_card_suspend_pre()
  ASoC: soc-card: move snd_soc_card_subclass to soc-card
  ASoC: soc-card: move snd_soc_card_get_codec_dai() to soc-card
  ASoC: soc-card: move snd_soc_card_set/get_drvdata() to soc-card
  ASoC: soc-card: move snd_soc_card_jack_new() to soc-card
  ...

Author: torvalds

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

@@ -17,6 +17,8 @@ properties:
   compatible:
     enum:
       - fsl,imx8qxp-dsp
+      - fsl,imx8qm-dsp
+      - fsl,imx8mp-dsp
 
   reg:
     description: Should contain register location and length

Documentation/devicetree/bindings/sound/da7213.txt

@@ -1,9 +1,9 @@
-Dialog Semiconductor DA7213 Audio Codec bindings
+Dialog Semiconductor DA7212/DA7213 Audio Codec bindings
 
 ======
 
 Required properties:
-- compatible : Should be "dlg,da7213"
+- compatible : Should be "dlg,da7212" or "dlg,da7213"
 - reg: Specifies the I2C slave address
 
 Optional properties:
@@ -21,6 +21,10 @@ Optional properties:
 - dlg,dmic-clkrate : DMIC clock frequency (Hz).
 	[<1500000>, <3000000>]
 
+ - VDDA-supply : Regulator phandle for Analogue power supply
+ - VDDMIC-supply : Regulator phandle for Mic Bias
+ - VDDIO-supply : Regulator phandle for I/O power supply
+
 ======
 
 Example:

Documentation/devicetree/bindings/sound/fsl,asrc.txt

@@ -51,6 +51,10 @@ Optional properties:
 			  will be in use as default. Otherwise, the big endian
 			  mode will be in use for all the device registers.
 
+   - fsl,asrc-format	: Defines a mutual sample format used by DPCM Back
+			  Ends, which can replace the fsl,asrc-width.
+			  The value is 2 (S16_LE), or 6 (S24_LE).
+
 Example:
 
 asrc: asrc@2034000 {

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

@@ -0,0 +1,101 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/fsl,easrc.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: NXP Asynchronous Sample Rate Converter (ASRC) Controller
+
+maintainers:
+  - Shengjiu Wang <[email protected]>
+
+properties:
+  $nodename:
+    pattern: "^easrc@.*"
+
+  compatible:
+    const: fsl,imx8mn-easrc
+
+  reg:
+    maxItems: 1
+
+  interrupts:
+    maxItems: 1
+
+  clocks:
+    items:
+      - description: Peripheral clock
+
+  clock-names:
+    items:
+      - const: mem
+
+  dmas:
+    maxItems: 8
+
+  dma-names:
+    items:
+      - const: ctx0_rx
+      - const: ctx0_tx
+      - const: ctx1_rx
+      - const: ctx1_tx
+      - const: ctx2_rx
+      - const: ctx2_tx
+      - const: ctx3_rx
+      - const: ctx3_tx
+
+  firmware-name:
+    allOf:
+      - $ref: /schemas/types.yaml#/definitions/string
+      - const: imx/easrc/easrc-imx8mn.bin
+    description: The coefficient table for the filters
+
+  fsl,asrc-rate:
+    allOf:
+      - $ref: /schemas/types.yaml#/definitions/uint32
+      - minimum: 8000
+      - maximum: 192000
+    description: Defines a mutual sample rate used by DPCM Back Ends
+
+  fsl,asrc-format:
+    allOf:
+      - $ref: /schemas/types.yaml#/definitions/uint32
+      - enum: [2, 6, 10, 32, 36]
+        default: 2
+    description:
+      Defines a mutual sample format used by DPCM Back Ends
+
+required:
+  - compatible
+  - reg
+  - interrupts
+  - clocks
+  - clock-names
+  - dmas
+  - dma-names
+  - firmware-name
+  - fsl,asrc-rate
+  - fsl,asrc-format
+
+examples:
+  - |
+    #include <dt-bindings/clock/imx8mn-clock.h>
+
+    easrc: easrc@300c0000 {
+           compatible = "fsl,imx8mn-easrc";
+           reg = <0x0 0x300c0000 0x0 0x10000>;
+           interrupts = <0x0 122 0x4>;
+           clocks = <&clk IMX8MN_CLK_ASRC_ROOT>;
+           clock-names = "mem";
+           dmas = <&sdma2 16 23 0> , <&sdma2 17 23 0>,
+                  <&sdma2 18 23 0> , <&sdma2 19 23 0>,
+                  <&sdma2 20 23 0> , <&sdma2 21 23 0>,
+                  <&sdma2 22 23 0> , <&sdma2 23 23 0>;
+           dma-names = "ctx0_rx", "ctx0_tx",
+                       "ctx1_rx", "ctx1_tx",
+                       "ctx2_rx", "ctx2_tx",
+                       "ctx3_rx", "ctx3_tx";
+           firmware-name = "imx/easrc/easrc-imx8mn.bin";
+           fsl,asrc-rate  = <8000>;
+           fsl,asrc-format = <2>;
+    };

Documentation/devicetree/bindings/sound/fsl,esai.txt

@@ -12,6 +12,7 @@ Required properties:
 			  "fsl,imx35-esai",
 			  "fsl,vf610-esai",
 			  "fsl,imx6ull-esai",
+			  "fsl,imx8qm-esai",
 
   - reg			: Offset and length of the register set for the device.
 

Documentation/devicetree/bindings/sound/marvell,mmp-sspa.yaml

@@ -0,0 +1,122 @@
+# SPDX-License-Identifier: (GPL-2.0+ OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/marvell,mmp-sspa.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Marvel SSPA Digital Audio Interface Bindings
+
+maintainers:
+  - Lubomir Rintel <[email protected]>
+
+properties:
+  $nodename:
+    pattern: "^audio-controller(@.*)?$"
+
+  compatible:
+    const: marvell,mmp-sspa
+
+  reg:
+    items:
+      - description: RX block
+      - description: TX block
+
+  interrupts:
+    maxItems: 1
+
+  clocks:
+    items:
+      - description: Clock for the Audio block
+      - description: I2S bit clock
+
+  clock-names:
+    items:
+      - const: audio
+      - const: bitclk
+
+  power-domains:
+    maxItems: 1
+
+  '#sound-dai-cells':
+    const: 0
+
+  dmas:
+    items:
+      - description: TX DMA Channel
+      - description: RX DMA Channel
+
+  dma-names:
+    items:
+      - const: tx
+      - const: rx
+
+  port:
+    type: object
+
+    properties:
+      endpoint:
+        type: object
+
+        properties:
+          remote-endpoint: true
+
+          frame-master:
+            type: boolean
+            description: SoC generates the frame clock
+
+          bitclock-master:
+            type: boolean
+            description: SoC generates the bit clock
+
+          dai-format:
+            $ref: /schemas/types.yaml#/definitions/string
+            description: The digital audio format
+            const: i2s
+
+        required:
+          - remote-endpoint
+
+    required:
+      - endpoint
+
+    additionalProperties: false
+
+required:
+  - "#sound-dai-cells"
+  - compatible
+  - reg
+  - interrupts
+  - clocks
+  - clock-names
+  - dmas
+  - dma-names
+  - port
+
+additionalProperties: false
+
+examples:
+  - |
+    #include <dt-bindings/clock/marvell,mmp2.h>
+
+    audio-controller@d42a0c00 {
+      compatible = "marvell,mmp-sspa";
+      reg = <0xd42a0c00 0x30>,
+            <0xd42a0c80 0x30>;
+      interrupts = <2>;
+      clock-names = "audio", "bitclk";
+      clocks = <&soc_clocks 127>,
+               <&audio_clk 1>;
+      #sound-dai-cells = <0>;
+      dmas = <&adma0 0>, <&adma0 1>;
+      dma-names = "tx", "rx";
+      port {
+        endpoint {
+          remote-endpoint = <&rt5631_0>;
+          frame-master;
+          bitclock-master;
+          dai-format = "i2s";
+        };
+      };
+    };
+
+...

Documentation/devicetree/bindings/sound/nau8810.txt

@@ -1,10 +1,11 @@
-NAU8810 audio CODEC
+NAU8810/NAU8812/NAU8814 audio CODEC
 
 This device supports I2C only.
 
 Required properties:
 
-  - compatible : "nuvoton,nau8810"
+  - compatible : One of "nuvoton,nau8810" or "nuvoton,nau8812" or
+	"nuvoton,nau8814"
 
   - reg : the I2C address of the device.
 

Documentation/devicetree/bindings/sound/nau8825.txt

@@ -101,5 +101,5 @@ Example:
       nuvoton,crosstalk-enable;
 
       clock-names = "mclk";
-      clocks = <&tegra_car TEGRA210_CLK_CLK_OUT_2>;
+      clocks = <&tegra_pmc TEGRA_PMC_CLK_OUT_2>;
   };

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

@@ -29,6 +29,7 @@ Optional properties:
 - nvidia,hp-det-gpios : The GPIO that detect headphones are plugged in
 - nvidia,int-mic-en-gpios : The GPIO that enables the internal microphone
 - nvidia,ext-mic-en-gpios : The GPIO that enables the external microphone
+- nvidia,headset : The Mic Jack represents state of the headset microphone pin
 
 Example:
 

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

@@ -30,13 +30,29 @@ Required properties:
 - reg			: Must contain an address for each entry in reg-names.
 - reg-names		: A list which must include the following entries:
 				* "lpass-lpaif"
+- #address-cells	: Must be 1
+- #size-cells		: Must be 0
 
 
 
 Optional properties:
 
 - qcom,adsp		: Phandle for the audio DSP node
 
+By default, the driver uses up to 4 MI2S SD lines, for a total of 8 channels.
+The SD lines to use can be configured by adding subnodes for each of the DAIs.
+
+Required properties for each DAI (represented by a subnode):
+- reg			: Must be one of the DAI IDs
+			  (usually part of dt-bindings header)
+- qcom,playback-sd-lines: List of serial data lines to use for playback
+			  Each SD line should be represented by a number from 0-3.
+- qcom,capture-sd-lines	: List of serial data lines to use for capture
+			  Each SD line should be represented by a number from 0-3.
+
+Note that adding a subnode changes the default to "no lines configured",
+so both playback and capture lines should be configured when a subnode is added.
+
 Example:
 
 lpass@28100000 {
@@ -51,4 +67,13 @@ lpass@28100000 {
 	reg = <0x28100000 0x10000>;
 	reg-names = "lpass-lpaif";
 	qcom,adsp = <&adsp>;
+
+	#address-cells = <1>;
+	#size-cells = <0>;
+
+	/* Optional to set different MI2S SD lines */
+	dai@3 {
+		reg = <MI2S_QUATERNARY>;
+		qcom,playback-sd-lines = <0 1>;
+	};
 };