Merge tag 'sound-6.10-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "This one became bigger than usual, not in the total size but rather
  containing lots of small changes all over the places.

  The majority of changes are about ASoC, especially SOF / Intel stuff,
  and we see an interesting work for ASoC DAPM graph visualization,
  while there are many other code cleanup and refactoring, too.

  Core:
   - A deadlock fix at device disconnection
   - A new tool dapm-graph for visualising the DAPM state

  ASoC:
   - Large updates throughout the Intel audio drivers
   - Fixes and clarifications for the DAPM documentation
   - Cleanups of accessors for driver data, module labelling, and for
     constification
   - Modernsation and cleanup work in the Mediatek drivers
   - Several fixes and features for the DaVinci I2S driver
   - New drivers for several AMD and Intel platforms, Nuvoton NAU8325,
     Rockchip RK3308 and Texas Instruments PCM6240

  HD-audio:
   - Cleanup for CONFIG_PM dependencies
   - Cirrus HD-audio codec fixes and quirks

  Others:
   - Series of tree-wide fixes in Makefiles to use *-y
   - Additions of missing module descriptions
   - Scarlett2 USB mixer enhancements
   - A series of legacy emu10k1 fixes and improvements"

* tag 'sound-6.10-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (603 commits)
  ALSA: hda/realtek: Drop doubly quirk entry for 103c:8a2e
  ALSA: hda/realtek - fixed headset Mic not show
  ASoC: SOF: amd: Fix build error with built-in config
  ALSA: scarlett2: Increase mixer range to +12dB
  ALSA: scarlett2: Add S/PDIF source selection controls
  ALSA: core: Remove superfluous CONFIG_PM
  ALSA: Fix deadlocks with kctl removals at disconnection
  ASoC: audio-graph-card2: call of_node_get() before of_get_next_child()
  ASoC: SOF: amd: Correct spaces in Makefile
  ASoC: rt715-sdca-sdw: Fix wrong complete waiting in rt715_dev_resume()
  ASoC: Intel: sof_sdw_rt_amp: use dai parameter
  ASoC: Intel: sof_sdw: add dai parameter to rtd_init callback
  ASoC: Intel: sof_sdw: use .controls/.widgets to add controls/widgets
  ASoC: Intel: sof_sdw: add controls and dapm widgets in codec_info
  ASoC: Intel: sof_sdw: use generic name for controls/widgets
  ASoC: Intel: sof_sdw_cs_amp: rename Speakers to Speaker
  ASoC: Intel: maxim-common: change max98373 data to static
  ASoC: Intel: sof_sdw: add max98373 dapm routes
  ASoC: Intel: sof_rt5682: use max_98373_dai_link function
  ASoC: Intel: sof_nau8825: use max_98373_dai_link function
  ...

39 files changed, 2049 insertions, 589 deletions

diff --git a/Documentation/devicetree/bindings/sound/davinci-mcbsp.txt b/Documentation/devicetree/bindings/sound/davinci-mcbsp.txt
deleted file mode 100644
index 3ffc2562fb31..000000000000
--- a/Documentation/devicetree/bindings/sound/davinci-mcbsp.txt
+++ /dev/null
@@ -1,50 +0,0 @@
-Texas Instruments DaVinci McBSP module
-~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
-
-This binding describes the "Multi-channel Buffered Serial Port" (McBSP)
-audio interface found in some TI DaVinci processors like the OMAP-L138 or AM180x.
-
-
-Required properties:
-~~~~~~~~~~~~~~~~~~~~
-- compatible :
-        "ti,da850-mcbsp" : for DA850, AM180x and OPAM-L138 platforms
-
-- reg : physical base address and length of the controller memory mapped
-        region(s).
-- reg-names : Should contain:
-        * "mpu" for the main registers (required).
-        * "dat" for the data FIFO (optional).
-
-- dmas: three element list of DMA controller phandles, DMA request line and
-	TC channel ordered triplets.
-- dma-names: identifier string for each DMA request line in the dmas property.
-	These strings correspond 1:1 with the ordered pairs in dmas. The dma
-	identifiers must be "rx" and "tx".
-
-Optional properties:
-~~~~~~~~~~~~~~~~~~~~
-- interrupts : Interrupt numbers for McBSP
-- interrupt-names : Known interrupt names are "rx" and "tx"
-
-- pinctrl-0: Should specify pin control group used for this controller.
-- pinctrl-names: Should contain only one value - "default", for more details
-        please refer to pinctrl-bindings.txt
-
-Example (AM1808):
-~~~~~~~~~~~~~~~~~
-
-mcbsp0: mcbsp@1d10000 {
-	compatible = "ti,da850-mcbsp";
-	pinctrl-names = "default";
-	pinctrl-0 = <&mcbsp0_pins>;
-
-	reg = 	<0x00110000 0x1000>,
-		<0x00310000 0x1000>;
-	reg-names = "mpu", "dat";
-	interrupts = <97 98>;
-	interrupt-names = "rx", "tx";
-	dmas = <&edma0 3 1
-		&edma0 2 1>;
-	dma-names = "tx", "rx";
-};
diff --git a/Documentation/devicetree/bindings/sound/davinci-mcbsp.yaml b/Documentation/devicetree/bindings/sound/davinci-mcbsp.yaml
new file mode 100644
index 000000000000..4fa677023827
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/davinci-mcbsp.yaml
@@ -0,0 +1,113 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/davinci-mcbsp.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: McBSP Controller for TI SoCs
+
+maintainers:
+  - Bastien Curutchet <bastien.curutchet@bootlin.com>
+
+allOf:
+  - $ref: dai-common.yaml#
+
+properties:
+  compatible:
+    enum:
+      - ti,da850-mcbsp
+
+  reg:
+    minItems: 1
+    items:
+      - description: CFG registers
+      - description: data registers
+
+  reg-names:
+    minItems: 1
+    items:
+      - const: mpu
+      - const: dat
+
+  dmas:
+    items:
+      - description: transmission DMA channel
+      - description: reception DMA channel
+
+  dma-names:
+    items:
+      - const: tx
+      - const: rx
+
+  interrupts:
+    items:
+      - description: RX interrupt
+      - description: TX interrupt
+
+  interrupt-names:
+    items:
+      - const: rx
+      - const: tx
+
+  clocks:
+    minItems: 1
+    items:
+      - description: functional clock
+      - description: external input clock for sample rate generator.
+
+  clock-names:
+    minItems: 1
+    items:
+      - const: fck
+      - const: clks
+
+  power-domains:
+    maxItems: 1
+
+  "#sound-dai-cells":
+    const: 0
+
+  ti,T1-framing-tx:
+    $ref: /schemas/types.yaml#/definitions/flag
+    description:
+      If the property is present, tx data delay is set to 2 bit clock periods.
+      McBSP will insert a blank period (high-impedance period) before the first
+      data bit. This can be used to interface to T1-framing devices.
+
+  ti,T1-framing-rx:
+    $ref: /schemas/types.yaml#/definitions/flag
+    description:
+      If the property is present, rx data delay is set to 2 bit clock periods.
+      McBSP will discard the bit preceding the data stream (called framing bit).
+      This can be used to interface to T1-framing devices.
+
+required:
+  - "#sound-dai-cells"
+  - compatible
+  - reg
+  - reg-names
+  - dmas
+  - dma-names
+  - clocks
+
+unevaluatedProperties: false
+
+examples:
+  - |
+    mcbsp0@1d10000 {
+      #sound-dai-cells = <0>;
+      compatible = "ti,da850-mcbsp";
+      pinctrl-names = "default";
+      pinctrl-0 = <&mcbsp0_pins>;
+
+      reg = <0x111000 0x1000>,
+            <0x311000 0x1000>;
+      reg-names = "mpu", "dat";
+      interrupts = <97>, <98>;
+      interrupt-names = "rx", "tx";
+      dmas = <&edma0 3 1>,
+             <&edma0 2 1>;
+      dma-names = "tx", "rx";
+
+      clocks = <&psc1 14>;
+    };
diff --git a/Documentation/devicetree/bindings/sound/fsl,audmix.txt b/Documentation/devicetree/bindings/sound/fsl,audmix.txt
deleted file mode 100644
index 840b7e0d6a63..000000000000
--- a/Documentation/devicetree/bindings/sound/fsl,audmix.txt
+++ /dev/null
@@ -1,50 +0,0 @@
-NXP Audio Mixer (AUDMIX).
-
-The Audio Mixer is a on-chip functional module that allows mixing of two
-audio streams into a single audio stream. Audio Mixer has two input serial
-audio interfaces. These are driven by two Synchronous Audio interface
-modules (SAI). Each input serial interface carries 8 audio channels in its
-frame in TDM manner. Mixer mixes audio samples of corresponding channels
-from two interfaces into a single sample. Before mixing, audio samples of
-two inputs can be attenuated based on configuration. The output of the
-Audio Mixer is also a serial audio interface. Like input interfaces it has
-the same TDM frame format. This output is used to drive the serial DAC TDM
-interface of audio codec and also sent to the external pins along with the
-receive path of normal audio SAI module for readback by the CPU.
-
-The output of Audio Mixer can be selected from any of the three streams
- - serial audio input 1
- - serial audio input 2
- - mixed audio
-
-Mixing operation is independent of audio sample rate but the two audio
-input streams must have same audio sample rate with same number of channels
-in TDM frame to be eligible for mixing.
-
-Device driver required properties:
-=================================
-  - compatible		: Compatible list, contains "fsl,imx8qm-audmix"
-
-  - reg			: Offset and length of the register set for the device.
-
-  - clocks		: Must contain an entry for each entry in clock-names.
-
-  - clock-names		: Must include the "ipg" for register access.
-
-  - power-domains	: Must contain the phandle to AUDMIX power domain node
-
-  - dais		: Must contain a list of phandles to AUDMIX connected
-			  DAIs. The current implementation requires two phandles
-			  to SAI interfaces to be provided, the first SAI in the
-			  list being used to route the AUDMIX output.
-
-Device driver configuration example:
-======================================
-  audmix: audmix@59840000 {
-    compatible = "fsl,imx8qm-audmix";
-    reg = <0x0 0x59840000 0x0 0x10000>;
-    clocks = <&clk IMX8QXP_AUD_AUDMIX_IPG>;
-    clock-names = "ipg";
-    power-domains = <&pd_audmix>;
-    dais = <&sai4>, <&sai5>;
-  };
diff --git a/Documentation/devicetree/bindings/sound/fsl,audmix.yaml b/Documentation/devicetree/bindings/sound/fsl,audmix.yaml
new file mode 100644
index 000000000000..9413b901cf77
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/fsl,audmix.yaml
@@ -0,0 +1,83 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/fsl,audmix.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: NXP Audio Mixer (AUDMIX).
+
+maintainers:
+  - Shengjiu Wang <shengjiu.wang@nxp.com>
+  - Frank Li <Frank.Li@nxp.com>
+
+description: |
+  The Audio Mixer is a on-chip functional module that allows mixing of two
+  audio streams into a single audio stream. Audio Mixer has two input serial
+  audio interfaces. These are driven by two Synchronous Audio interface
+  modules (SAI). Each input serial interface carries 8 audio channels in its
+  frame in TDM manner. Mixer mixes audio samples of corresponding channels
+  from two interfaces into a single sample. Before mixing, audio samples of
+  two inputs can be attenuated based on configuration. The output of the
+  Audio Mixer is also a serial audio interface. Like input interfaces it has
+  the same TDM frame format. This output is used to drive the serial DAC TDM
+  interface of audio codec and also sent to the external pins along with the
+  receive path of normal audio SAI module for readback by the CPU.
+
+  The output of Audio Mixer can be selected from any of the three streams
+    - serial audio input 1
+    - serial audio input 2
+    - mixed audio
+
+  Mixing operation is independent of audio sample rate but the two audio
+  input streams must have same audio sample rate with same number of channels
+  in TDM frame to be eligible for mixing.
+
+properties:
+  compatible:
+    const: fsl,imx8qm-audmix
+
+  reg:
+    maxItems: 1
+
+  clocks:
+    maxItems: 1
+
+  clock-names:
+    items:
+      - const: ipg
+
+  power-domains:
+    maxItems: 1
+
+  dais:
+    description: contain a list of phandles to AUDMIX connected DAIs.
+    $ref: /schemas/types.yaml#/definitions/phandle-array
+    minItems: 2
+    items:
+      - description: the AUDMIX output
+        maxItems: 1
+      - description: serial audio input 1
+        maxItems: 1
+      - description: serial audio input 2
+        maxItems: 1
+
+required:
+  - compatible
+  - reg
+  - clocks
+  - clock-names
+  - power-domains
+  - dais
+
+unevaluatedProperties: false
+
+examples:
+  - |
+    audmix@59840000 {
+      compatible = "fsl,imx8qm-audmix";
+      reg = <0x59840000 0x10000>;
+      clocks = <&amix_lpcg 0>;
+      clock-names = "ipg";
+      power-domains = <&pd_audmix>;
+      dais = <&sai4>, <&sai5>;
+    };
diff --git a/Documentation/devicetree/bindings/sound/fsl,esai.txt b/Documentation/devicetree/bindings/sound/fsl,esai.txt
deleted file mode 100644
index 90112ca1ff42..000000000000
--- a/Documentation/devicetree/bindings/sound/fsl,esai.txt
+++ /dev/null
@@ -1,68 +0,0 @@
-Freescale Enhanced Serial Audio Interface (ESAI) Controller
-
-The Enhanced Serial Audio Interface (ESAI) provides a full-duplex serial port
-for serial communication with a variety of serial devices, including industry
-standard codecs, Sony/Phillips Digital Interface (S/PDIF) transceivers, and
-other DSPs. It has up to six transmitters and four receivers.
-
-Required properties:
-
-  - compatible		: Compatible list, should contain one of the following
-			  compatibles:
-			  "fsl,imx35-esai",
-			  "fsl,vf610-esai",
-			  "fsl,imx6ull-esai",
-			  "fsl,imx8qm-esai",
-
-  - 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:
-	"core"		  The core clock used to access registers
-	"extal"		  The esai baud clock for esai controller used to
-			  derive HCK, SCK and FS.
-	"fsys"		  The system clock derived from ahb clock used to
-			  derive HCK, SCK and FS.
-	"spba"		  The spba clock is required when ESAI is placed as a
-			  bus slave of the Shared Peripheral Bus and when two
-			  or more bus masters (CPU, DMA or DSP) try to access
-			  it. This property is optional depending on the SoC
-			  design.
-
-  - fsl,fifo-depth	: The number of elements in the transmit and receive
-			  FIFOs. This number is the maximum allowed value for
-			  TFCR[TFWM] or RFCR[RFWM].
-
-  - fsl,esai-synchronous: This is a boolean property. If present, indicating
-			  that ESAI would work in the synchronous mode, which
-			  means all the settings for Receiving would be
-			  duplicated from Transmission related registers.
-
-Optional properties:
-
-  - big-endian		: If this property is absent, the native endian mode
-			  will be in use as default, or the big endian mode
-			  will be in use for all the device registers.
-
-Example:
-
-esai: esai@2024000 {
-	compatible = "fsl,imx35-esai";
-	reg = <0x02024000 0x4000>;
-	interrupts = <0 51 0x04>;
-	clocks = <&clks 208>, <&clks 118>, <&clks 208>;
-	clock-names = "core", "extal", "fsys";
-	dmas = <&sdma 23 21 0>, <&sdma 24 21 0>;
-	dma-names = "rx", "tx";
-	fsl,fifo-depth = <128>;
-	fsl,esai-synchronous;
-	big-endian;
-};
diff --git a/Documentation/devicetree/bindings/sound/fsl,esai.yaml b/Documentation/devicetree/bindings/sound/fsl,esai.yaml
new file mode 100644
index 000000000000..f99ed20fa684
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/fsl,esai.yaml
@@ -0,0 +1,118 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/fsl,esai.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Freescale Enhanced Serial Audio Interface (ESAI) Controller
+
+maintainers:
+  - Shengjiu Wang <shengjiu.wang@nxp.com>
+  - Frank Li <Frank.Li@nxp.com>
+
+description:
+  The Enhanced Serial Audio Interface (ESAI) provides a full-duplex serial port
+  for serial communication with a variety of serial devices, including industry
+  standard codecs, Sony/Phillips Digital Interface (S/PDIF) transceivers, and
+  other DSPs. It has up to six transmitters and four receivers.
+
+properties:
+  compatible:
+    enum:
+      - fsl,imx35-esai
+      - fsl,imx6ull-esai
+      - fsl,imx8qm-esai
+      - fsl,vf610-esai
+
+  reg:
+    maxItems: 1
+
+  interrupts:
+    maxItems: 1
+
+  clocks:
+    minItems: 3
+    items:
+      - description:
+          The core clock used to access registers.
+      - description:
+          The esai baud clock for esai controller used to
+          derive HCK, SCK and FS.
+      - description:
+          The system clock derived from ahb clock used to
+          derive HCK, SCK and FS.
+      - description:
+          The spba clock is required when ESAI is placed as a
+          bus slave of the Shared Peripheral Bus and when two
+          or more bus masters (CPU, DMA or DSP) try to access
+          it. This property is optional depending on the SoC
+          design.
+
+  clock-names:
+    minItems: 3
+    items:
+      - const: core
+      - const: extal
+      - const: fsys
+      - const: spba
+
+  dmas:
+    minItems: 2
+    maxItems: 2
+
+  dma-names:
+    items:
+      - const: rx
+      - const: tx
+
+  fsl,fifo-depth:
+    $ref: /schemas/types.yaml#/definitions/uint32
+    default: 64
+    description:
+      The number of elements in the transmit and receive
+      FIFOs. This number is the maximum allowed value for
+      TFCR[TFWM] or RFCR[RFWM].
+
+  fsl,esai-synchronous:
+    $ref: /schemas/types.yaml#/definitions/flag
+    description:
+      This is a boolean property. If present, indicating
+      that ESAI would work in the synchronous mode, which
+      means all the settings for Receiving would be
+      duplicated from Transmission related registers.
+
+  big-endian:
+    $ref: /schemas/types.yaml#/definitions/flag
+    description:
+      If this property is absent, the native endian mode
+      will be in use as default, or the big endian mode
+      will be in use for all the device registers.
+
+required:
+  - compatible
+  - reg
+  - interrupts
+  - clocks
+  - clock-names
+  - dmas
+  - dma-names
+
+unevaluatedProperties: false
+
+allOf:
+  - $ref: dai-common.yaml#
+
+examples:
+  - |
+    esai@2024000 {
+      compatible = "fsl,imx35-esai";
+      reg = <0x02024000 0x4000>;
+      interrupts = <0 51 0x04>;
+      clocks = <&clks 208>, <&clks 118>, <&clks 208>;
+      clock-names = "core", "extal", "fsys";
+      dmas = <&sdma 23 21 0>, <&sdma 24 21 0>;
+      dma-names = "rx", "tx";
+      fsl,fifo-depth = <128>;
+      fsl,esai-synchronous;
+      big-endian;
+    };
diff --git a/Documentation/devicetree/bindings/sound/fsl,imx-asrc.yaml b/Documentation/devicetree/bindings/sound/fsl,imx-asrc.yaml
index bfef2fcb75b1..76aa1f248488 100644
--- a/Documentation/devicetree/bindings/sound/fsl,imx-asrc.yaml
+++ b/Documentation/devicetree/bindings/sound/fsl,imx-asrc.yaml
@@ -74,6 +74,9 @@ properties:
       - const: asrck_f
       - const: spba
 
+  power-domains:
+    maxItems: 1
+
   fsl,asrc-rate:
     $ref: /schemas/types.yaml#/definitions/uint32
     description: The mutual sample rate used by DPCM Back Ends
@@ -131,6 +134,17 @@ allOf:
       properties:
         fsl,asrc-clk-map: false
 
+  - if:
+      properties:
+        compatible:
+          contains:
+            enum:
+              - fsl,imx8qm-asrc
+              - fsl,imx8qxp-asrc
+    then:
+      required:
+        - power-domains
+
 additionalProperties: false
 
 examples:
diff --git a/Documentation/devicetree/bindings/sound/fsl,imx-audio-spdif.yaml b/Documentation/devicetree/bindings/sound/fsl,imx-audio-spdif.yaml
new file mode 100644
index 000000000000..5fc543d02ecb
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/fsl,imx-audio-spdif.yaml
@@ -0,0 +1,66 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/fsl,imx-audio-spdif.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Freescale i.MX audio complex with S/PDIF transceiver
+
+maintainers:
+  - Shengjiu Wang <shengjiu.wang@nxp.com>
+
+properties:
+  compatible:
+    oneOf:
+      - items:
+          - enum:
+              - fsl,imx-sabreauto-spdif
+              - fsl,imx6sx-sdb-spdif
+          - const: fsl,imx-audio-spdif
+      - enum:
+          - fsl,imx-audio-spdif
+
+  model:
+    $ref: /schemas/types.yaml#/definitions/string
+    description: User specified audio sound card name
+
+  spdif-controller:
+    $ref: /schemas/types.yaml#/definitions/phandle
+    description: The phandle of the i.MX S/PDIF controller
+
+  spdif-out:
+    type: boolean
+    description:
+      If present, the transmitting function of S/PDIF will be enabled,
+      indicating there's a physical S/PDIF out connector or jack on the
+      board or it's connecting to some other IP block, such as an HDMI
+      encoder or display-controller.
+
+  spdif-in:
+    type: boolean
+    description:
+      If present, the receiving function of S/PDIF will be enabled,
+      indicating there is a physical S/PDIF in connector/jack on the board.
+
+required:
+  - compatible
+  - model
+  - spdif-controller
+
+anyOf:
+  - required:
+      - spdif-in
+  - required:
+      - spdif-out
+
+additionalProperties: false
+
+examples:
+  - |
+    sound-spdif {
+        compatible = "fsl,imx-audio-spdif";
+        model = "imx-spdif";
+        spdif-controller = <&spdif>;
+        spdif-out;
+        spdif-in;
+    };
diff --git a/Documentation/devicetree/bindings/sound/fsl,sai.yaml b/Documentation/devicetree/bindings/sound/fsl,sai.yaml
index 2456d958adee..a5d9c246cc47 100644
--- a/Documentation/devicetree/bindings/sound/fsl,sai.yaml
+++ b/Documentation/devicetree/bindings/sound/fsl,sai.yaml
@@ -81,14 +81,12 @@ properties:
 
   dmas:
     minItems: 1
-    items:
-      - description: DMA controller phandle and request line for RX
-      - description: DMA controller phandle and request line for TX
+    maxItems: 2
 
   dma-names:
     minItems: 1
     items:
-      - const: rx
+      - enum: [ rx, tx ]
       - const: tx
 
   interrupts:
diff --git a/Documentation/devicetree/bindings/sound/fsl,spdif.yaml b/Documentation/devicetree