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
  ...

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

-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

+# 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 <shengjiu.wang@nxp.com>
+
+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

+# 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 <lkundrak@v3.sk>
+
+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

-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

@@ -30,6 +30,8 @@ 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
 
 
 
@@ -37,6 +39,20 @@ 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>;
+	};
 };

Documentation/devicetree/bindings/sound/qcom,q6adm.txt

@@ -29,7 +29,7 @@ used by the apr service device.
 	Definition: Must be 0
 
 = EXAMPLE
-q6adm@8 {
+apr-service@8 {
 	compatible = "qcom,q6adm";
 	reg = <APR_SVC_ADM>;
 	q6routing: routing {

Documentation/devicetree/bindings/sound/qcom,q6afe.txt

@@ -100,7 +100,7 @@ configuration of each dai. Must contain the following properties.
 
 = EXAMPLE
 
-q6afe@4 {
+apr-service@4 {
 	compatible = "qcom,q6afe";
 	reg = <APR_SVC_AFE>;
 
@@ -110,12 +110,12 @@ q6afe@4 {
 		#address-cells = <1>;
 		#size-cells = <0>;
 
-		hdmi@1 {
-			reg = <1>;
+		dai@1 {
+			reg = <HDMI_RX>;
 		};
 
-		tdm@24 {
-			reg = <24>;
+		dai@24 {
+			reg = <PRIMARY_TDM_RX_0>;
 			qcom,tdm-sync-mode = <1>:
 			qcom,tdm-sync-src = <1>;
 			qcom,tdm-data-out = <0>;
@@ -125,8 +125,8 @@ q6afe@4 {
 
 		};
 
-		tdm@25 {
-			reg = <25>;
+		dai@25 {
+			reg = <PRIMARY_TDM_TX_0>;
 			qcom,tdm-sync-mode = <1>:
 			qcom,tdm-sync-src = <1>;
 			qcom,tdm-data-out = <0>;
@@ -135,43 +135,43 @@ q6afe@4 {
 			qcom,tdm-data-align = <0>;
 		};
 
-		prim-mi2s-rx@16 {
-			reg = <16>;
+		dai@16 {
+			reg = <PRIMARY_MI2S_RX>;
 			qcom,sd-lines = <0 2>;
 		};
 
-		prim-mi2s-tx@17 {
-			reg = <17>;
+		dai@17 {
+			reg = <PRIMARY_MI2S_TX>;
 			qcom,sd-lines = <1>;
 		};
 
-		sec-mi2s-rx@18 {
-			reg = <18>;
+		dai@18 {
+			reg = <SECONDARY_MI2S_RX>;
 			qcom,sd-lines = <0 3>;
 		};
 
-		sec-mi2s-tx@19 {
-			reg = <19>;
+		dai@19 {
+			reg = <SECONDARY_MI2S_TX>;
 			qcom,sd-lines = <1>;
 		};
 
-		tert-mi2s-rx@20 {
-			reg = <20>;
+		dai@20 {
+			reg = <TERTIARY_MI2S_RX>;
 			qcom,sd-lines = <1 3>;
 		};
 
-		tert-mi2s-tx@21 {
-			reg = <21>;
+		dai@21 {
+			reg = <TERTIARY_MI2S_TX>;
 			qcom,sd-lines = <0>;
 		};
 
-		quat-mi2s-rx@22 {
-			reg = <22>;
+		dai@22 {
+			reg = <QUATERNARY_MI2S_RX>;
 			qcom,sd-lines = <0>;
 		};
 
-		quat-mi2s-tx@23 {
-			reg = <23>;
+		dai@23 {
+			reg = <QUATERNARY_MI2S_TX>;
 			qcom,sd-lines = <1>;
 		};
 	};

Documentation/devicetree/bindings/sound/qcom,q6asm.txt

@@ -51,13 +51,16 @@ configuration of each dai. Must contain the following properties.
 
 = EXAMPLE
 
-q6asm@7 {
+apr-service@7 {
 	compatible = "qcom,q6asm";
 	reg = <APR_SVC_ASM>;
 	q6asmdai: dais {
 		compatible = "qcom,q6asm-dais";
+		#address-cells = <1>;
+		#size-cells = <0>;
 		#sound-dai-cells = <1>;
-		mm@0 {
+
+		dai@0 {
 			reg = <0>;
 			direction = <2>;
 			is-compress-dai;

Documentation/devicetree/bindings/sound/qcom,q6core.txt

@@ -15,7 +15,7 @@ used by the apr service device.
 		   example "qcom,q6core-v2.0"
 
 = EXAMPLE
-q6core@3 {
+apr-service@3 {
 	compatible = "qcom,q6core";
 	reg = <APR_SVC_ADSP_CORE>;
 };

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

@@ -263,6 +263,7 @@ Required properties:
 				  "renesas,rcar_sound-gen2" if generation2 (or RZ/G1)
 				  "renesas,rcar_sound-gen3" if generation3 (or RZ/G2)
 				  Examples with soctypes are:
+				    - "renesas,rcar_sound-r8a7742" (RZ/G1H)
 				    - "renesas,rcar_sound-r8a7743" (RZ/G1M)
 				    - "renesas,rcar_sound-r8a7744" (RZ/G1N)
 				    - "renesas,rcar_sound-r8a7745" (RZ/G1E)

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

@@ -24,6 +24,7 @@ properties:
             - rockchip,rk3188-i2s
             - rockchip,rk3228-i2s
             - rockchip,rk3288-i2s
+            - rockchip,rk3308-i2s
             - rockchip,rk3328-i2s
             - rockchip,rk3366-i2s
             - rockchip,rk3368-i2s
@@ -47,14 +48,15 @@ properties:
       - const: i2s_hclk
 
   dmas:
-    items:
-      - description: TX DMA Channel
-      - description: RX DMA Channel
+    minItems: 1
+    maxItems: 2
 
   dma-names:
-    items:
-      - const: tx
+    oneOf:
       - const: rx
+      - items:
+        - const: tx
+        - const: rx
 
   power-domains:
     maxItems: 1

Documentation/devicetree/bindings/sound/rt1016.txt

+RT1016 Stereo Class D Audio Amplifier
+
+This device supports I2C only.
+
+Required properties:
+
+- compatible : "realtek,rt1016".
+
+- reg : The I2C address of the device.
+
+
+Example:
+
+rt1016: codec@1a {
+	compatible = "realtek,rt1016";
+	reg = <0x1a>;
+};

Documentation/devicetree/bindings/sound/simple-card.txt

-Simple-Card:
-
-Simple-Card specifies audio DAI connections of SoC <-> codec.
-
-Required properties:
-
-- compatible				: "simple-audio-card"
-
-Optional properties:
-
-- simple-audio-card,name		: User specified audio sound card name, one string
-					  property.
-- simple-audio-card,widgets		: Please refer to widgets.txt.
-- simple-audio-card,routing		: A list of the connections between audio components.
-					  Each entry is a pair of strings, the first being the
-					  connection's sink, the second being the connection's
-					  source.
-- simple-audio-card,mclk-fs             : Multiplication factor between stream rate and codec
-					  mclk. When defined, mclk-fs property defined in
-					  dai-link sub nodes are ignored.
-- simple-audio-card,hp-det-gpio		: Reference to GPIO that signals when
-					  headphones are attached.
-- simple-audio-card,mic-det-gpio	: Reference to GPIO that signals when
-					  a microphone is attached.
-- simple-audio-card,aux-devs		: List of phandles pointing to auxiliary devices, such
-					  as amplifiers, to be added to the sound card.
-- simple-audio-card,pin-switches	: List of strings containing the widget names for
-					  which pin switches must be created.
-
-Optional subnodes:
-
-- simple-audio-card,dai-link		: Container for dai-link level
-					  properties and the CPU and CODEC
-					  sub-nodes. This container may be
-					  omitted when the card has only one
-					  DAI link. See the examples and the
-					  section below.
-
-Dai-link subnode properties and subnodes:
-
-If dai-link subnode is omitted and the subnode properties are directly
-under "sound"-node the subnode property and subnode names have to be
-prefixed with "simple-audio-card,"-prefix.
-
-Required dai-link subnodes:
-
-- cpu					: CPU   sub-node
-- codec					: CODEC sub-node
-
-Optional dai-link subnode properties:
-
-- format				: CPU/CODEC common audio format.
-					  "i2s", "right_j", "left_j" , "dsp_a"
-					  "dsp_b", "ac97", "pdm", "msb", "lsb"
-- frame-master				: Indicates dai-link frame master.
-					  phandle to a cpu or codec subnode.
-- bitclock-master			: Indicates dai-link bit clock master.
-					  phandle to a cpu or codec subnode.
-- bitclock-inversion			: bool property. Add this if the
-					  dai-link uses bit clock inversion.
-- frame-inversion			: bool property. Add this if the
-					  dai-link uses frame clock inversion.
-- mclk-fs             			: Multiplication factor between stream
-					  rate and codec mclk, applied only for
-					  the dai-link.
-
-For backward compatibility the frame-master and bitclock-master
-properties can be used as booleans in codec subnode to indicate if the
-codec is the dai-link frame or bit clock master. In this case there
-should be no dai-link node, the same properties should not be present
-at sound-node level, and the bitclock-inversion and frame-inversion
-properties should also be placed in the codec node if needed.
-
-Required CPU/CODEC subnodes properties:
-
-- sound-dai				: phandle and port of CPU/CODEC
-
-Optional CPU/CODEC subnodes properties:
-
-- dai-tdm-slot-num			: Please refer to tdm-slot.txt.
-- dai-tdm-slot-width			: Please refer to tdm-slot.txt.
-- clocks / system-clock-frequency	: specify subnode's clock if needed.
-					  it can be specified via "clocks" if system has
-					  clock node (= common clock), or "system-clock-frequency"
-					  (if system doens't support common clock)
-					  If a clock is specified, it is
-					  enabled with clk_prepare_enable()
-					  in dai startup() and disabled with
-					  clk_disable_unprepare() in dai
-					  shutdown().
-					  If a clock is specified and a
-					  multiplication factor is given with
-					  mclk-fs, the clock will be set to the
-					  calculated mclk frequency when the
-					  stream starts.
-- system-clock-direction-out		: specifies clock direction as 'out' on
-					  initialization. It is useful for some aCPUs with
-					  fixed clocks.
-
--------------------------------------------
-Example 1 - single DAI link:
--------------------------------------------
-
-sound {
-	compatible = "simple-audio-card";
-	simple-audio-card,name = "VF610-Tower-Sound-Card";
-	simple-audio-card,format = "left_j";
-	simple-audio-card,bitclock-master = <&dailink0_master>;
-	simple-audio-card,frame-master = <&dailink0_master>;
-	simple-audio-card,widgets =
-		"Microphone", "Microphone Jack",
-		"Headphone", "Headphone Jack",
-		"Speaker", "External Speaker";
-	simple-audio-card,routing =
-		"MIC_IN", "Microphone Jack",
-		"Headphone Jack", "HP_OUT",
-		"External Speaker", "LINE_OUT";
-
-	simple-audio-card,cpu {
-		sound-dai = <&sh_fsi2 0>;
-	};
-
-	dailink0_master: simple-audio-card,codec {
-		sound-dai = <&ak4648>;
-		clocks = <&osc>;
-	};
-};
-
-&i2c0 {
-	ak4648: ak4648@12 {
-		#sound-dai-cells = <0>;
-		compatible = "asahi-kasei,ak4648";
-		reg = <0x12>;
-	};
-};
-
-sh_fsi2: sh_fsi2@ec230000 {
-	#sound-dai-cells = <1>;
-	compatible = "renesas,sh_fsi2";
-	reg = <0xec230000 0x400>;
-	interrupt-parent = <&gic>;
-	interrupts = <0 146 0x4>;
-};
-
--------------------------------------------
-Example 2 - many DAI links:
--------------------------------------------
-
-sound {
-	compatible = "simple-audio-card";
-	simple-audio-card,name = "Cubox Audio";
-
-	simple-audio-card,dai-link@0 {		/* I2S - HDMI */
-		reg = <0>;
-		format = "i2s";
-		cpu {
-			sound-dai = <&audio1 0>;
-		};
-		codec {
-			sound-dai = <&tda998x 0>;
-		};
-	};
-
-	simple-audio-card,dai-link@1 {		/* S/PDIF - HDMI */
-		reg = <1>;
-		cpu {
-			sound-dai = <&audio1 1>;
-		};
-		codec {
-			sound-dai = <&tda998x 1>;
-		};
-	};
-
-	simple-audio-card,dai-link@2 {		/* S/PDIF - S/PDIF */
-		reg = <2>;
-		cpu {
-			sound-dai = <&audio1 1>;
-		};
-		codec {
-			sound-dai = <&spdif_codec>;
-		};
-	};
-};
-
--------------------------------------------
-Example 3 - route audio from IMX6 SSI2 through TLV320DAC3100 codec
-through TPA6130A2 amplifier to headphones:
--------------------------------------------
-
-&i2c0 {
-	codec: tlv320dac3100@18 {
-		compatible = "ti,tlv320dac3100";
-		...
-	}
-
-	amp: tpa6130a2@60 {
-		compatible = "ti,tpa6130a2";
-		...
-	}
-}
-
-sound {
-	compatible = "simple-audio-card";
-	...
-	simple-audio-card,widgets =
-		"Headphone", "Headphone Jack";
-	simple-audio-card,routing =
-		"Headphone Jack", "HPLEFT",
-		"Headphone Jack", "HPRIGHT",
-		"LEFTIN", "HPL",
-		"RIGHTIN", "HPR";
-	simple-audio-card,aux-devs = <&amp>;
-	simple-audio-card,cpu {
-		sound-dai = <&ssi2>;
-	};
-	simple-audio-card,codec {
-		sound-dai = <&codec>;
-		clocks = ...
-	};
-};
-
--------------------------------------------
-Example 4. Sampling Rate Conversion
--------------------------------------------
-
-sound {
-	compatible = "simple-audio-card";
-
-	simple-audio-card,name = "rsnd-ak4643";
-	simple-audio-card,format = "left_j";
-	simple-audio-card,bitclock-master = <&sndcodec>;
-	simple-audio-card,frame-master = <&sndcodec>;
-
-	simple-audio-card,convert-rate = <48000>;
-
-	simple-audio-card,prefix = "ak4642";
-	simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback",
-			"DAI0 Capture", "ak4642 Capture";
-
-	sndcpu: simple-audio-card,cpu {
-		sound-dai = <&rcar_sound>;
-	};
-
-	sndcodec: simple-audio-card,codec {
-		sound-dai = <&ak4643>;
-		system-clock-frequency = <11289600>;
-	};
-};
-
--------------------------------------------
-Example 5. 2 CPU 1 Codec (Mixing)
--------------------------------------------
-sound {
-	compatible = "simple-audio-card";
-
-	simple-audio-card,name = "rsnd-ak4643";
-	simple-audio-card,format = "left_j";
-	simple-audio-card,bitclock-master = <&dpcmcpu>;
-	simple-audio-card,frame-master = <&dpcmcpu>;
-
-	simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback",
-			"ak4642 Playback", "DAI1 Playback";
-
-	dpcmcpu: cpu@0 {
-		sound-dai = <&rcar_sound 0>;
-	};
-
-	cpu@1 {
-		sound-dai = <&rcar_sound 1>;
-	};
-
-	codec {
-		prefix = "ak4642";
-		sound-dai = <&ak4643>;
-		clocks = <&audio_clock>;
-	};
-};
-
--------------------------------------------
-Example 6 - many DAI links with DPCM:
--------------------------------------------
-
-CPU0 ------ ak4613
-CPU1 ------ PCM3168A-p  /* DPCM 1ch/2ch */
-CPU2 --/                /* DPCM 3ch/4ch */
-CPU3 --/                /* DPCM 5ch/6ch */
-CPU4 --/                /* DPCM 7ch/8ch */
-CPU5 ------ PCM3168A-c
-
-sound {
-	compatible = "simple-audio-card";
-
-	simple-audio-card,routing =
-		  "pcm3168a Playback", "DAI1 Playback",
-		  "pcm3168a Playback", "DAI2 Playback",
-		  "pcm3168a Playback", "DAI3 Playback",
-		  "pcm3168a Playback", "DAI4 Playback";
-
-	simple-audio-card,dai-link@0 {
-		format = "left_j";
-		bitclock-master = <&sndcpu0>;
-		frame-master = <&sndcpu0>;
-
-		sndcpu0: cpu {
-			sound-dai = <&rcar_sound 0>;
-		};
-		codec {
-			sound-dai = <&ak4613>;
-		};
-	};
-	simple-audio-card,dai-link@1 {
-		format = "i2s";
-		bitclock-master = <&sndcpu1>;
-		frame-master = <&sndcpu1>;
-
-		convert-channels = <8>; /* TDM Split */
-
-		sndcpu1: cpu@0 {
-			sound-dai = <&rcar_sound 1>;
-		};
-		cpu@1 {
-			sound-dai = <&rcar_sound 2>;
-		};
-		cpu@2 {
-			sound-dai = <&rcar_sound 3>;
-		};
-		cpu@3 {
-			sound-dai = <&rcar_sound 4>;
-		};
-		codec {
-			mclk-fs = <512>;
-			prefix = "pcm3168a";
-			dai-tdm-slot-num = <8>;
-			sound-dai = <&pcm3168a 0>;
-		};
-	};
-	simple-audio-card,dai-link@2 {
-		format = "i2s";
-		bitclock-master = <&sndcpu2>;
-		frame-master = <&sndcpu2>;
-
-		sndcpu2: cpu {
-			sound-dai = <&rcar_sound 5>;
-		};
-		codec {
-			mclk-fs = <512>;
-			prefix = "pcm3168a";
-			sound-dai = <&pcm3168a 1>;
-		};
-	};
-};

Documentation/devicetree/bindings/sound/tlv320adcx140.yaml

@@ -63,6 +63,55 @@ properties:
       - $ref: /schemas/types.yaml#/definitions/uint32
       - enum: [0, 1, 2]
 
+  ti,pdm-edge-select:
+    description: |
+       Defines the PDMCLK sampling edge configuration for the PDM inputs.  This
+       array is defined as <PDMIN1 PDMIN2 PDMIN3 PDMIN4>.
+
+       0 - (default) Odd channel is latched on the negative edge and even
+       channel is latched on the the positive edge.
+       1 - Odd channel is latched on the positive edge and even channel is
+       latched on the the negative edge.
+
+       PDMIN1 - PDMCLK latching edge used for channel 1 and 2 data
+       PDMIN2 - PDMCLK latching edge used for channel 3 and 4 data
+       PDMIN3 - PDMCLK latching edge used for channel 5 and 6 data
+       PDMIN4 - PDMCLK latching edge used for channel 7 and 8 data
+
+    allOf:
+      - $ref: /schemas/types.yaml#/definitions/uint32-array
+      - minItems: 1
+        maxItems: 4
+        items:
+          maximum: 1
+        default: [0, 0, 0, 0]
+
+  ti,gpi-config:
+    description: |
+       Defines the configuration for the general purpose input pins (GPI).
+       The array is defined as <GPI1 GPI2 GPI3 GPI4>.
+
+       0 - (default) disabled
+       1 - GPIX is configured as a general-purpose input (GPI)
+       2 - GPIX is configured as a master clock input (MCLK)
+       3 - GPIX is configured as an ASI input for daisy-chain (SDIN)
+       4 - GPIX is configured as a PDM data input for channel 1 and channel
+            (PDMDIN1)
+       5 - GPIX is configured as a PDM data input for channel 3 and channel
+            (PDMDIN2)
+       6 - GPIX is configured as a PDM data input for channel 5 and channel
+            (PDMDIN3)
+       7 - GPIX is configured as a PDM data input for channel 7 and channel
+            (PDMDIN4)
+
+    allOf:
+      - $ref: /schemas/types.yaml#/definitions/uint32-array
+      - minItems: 1
+        maxItems: 4
+        items:
+          maximum: 7
+        default: [0, 0, 0, 0]
+
 required:
   - compatible
   - reg
@@ -77,6 +126,8 @@ examples:
         compatible = "ti,tlv320adc5140";
         reg = <0x4c>;
         ti,mic-bias-source = <6>;
+        ti,pdm-edge-select = <0 1 0 1>;
+        ti,gpi-config = <4 5 6 7>;
         reset-gpios = <&gpio0 14 GPIO_ACTIVE_HIGH>;
       };
     };

Documentation/devicetree/bindings/sound/wm8994.txt

@@ -14,9 +14,15 @@ Required properties:
   - #gpio-cells : Must be 2. The first cell is the pin number and the
     second cell is used to specify optional parameters (currently unused).
 
-  - AVDD2-supply, DBVDD1-supply, DBVDD2-supply, DBVDD3-supply, CPVDD-supply,
-    SPKVDD1-supply, SPKVDD2-supply : power supplies for the device, as covered
-    in Documentation/devicetree/bindings/regulator/regulator.txt
+  - power supplies for the device, as covered in
+    Documentation/devicetree/bindings/regulator/regulator.txt, depending
+    on compatible:
+    - for wlf,wm1811 and wlf,wm8958:
+      AVDD1-supply, AVDD2-supply, DBVDD1-supply, DBVDD2-supply, DBVDD3-supply,
+      DCVDD-supply, CPVDD-supply, SPKVDD1-supply, SPKVDD2-supply
+    - for wlf,wm8994:
+      AVDD1-supply, AVDD2-supply, DBVDD-supply, DCVDD-supply, CPVDD-supply,
+      SPKVDD1-supply, SPKVDD2-supply
 
 Optional properties:
 
@@ -73,11 +79,11 @@ wm8994: codec@1a {
 
 	lineout1-se;
 
+	AVDD1-supply = <&regulator>;
 	AVDD2-supply = <&regulator>;
 	CPVDD-supply = <&regulator>;
-	DBVDD1-supply = <&regulator>;
-	DBVDD2-supply = <&regulator>;
-	DBVDD3-supply = <&regulator>;
+	DBVDD-supply = <&regulator>;
+	DCVDD-supply = <&regulator>;
 	SPKVDD1-supply = <&regulator>;
 	SPKVDD2-supply = <&regulator>;
 };

Documentation/devicetree/bindings/sound/zl38060.yaml

+# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/zl38060.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: ZL38060 Connected Home Audio Processor from Microsemi.
+
+description: |
+  The ZL38060 is a "Connected Home Audio Processor" from Microsemi,
+  which consists of a Digital Signal Processor (DSP), several Digital
+  Audio Interfaces (DAIs), analog outputs, and a block of 14 GPIOs.
+
+maintainers:
+  - Jaroslav Kysela <perex@perex.cz>
+  - Takashi Iwai <tiwai@suse.com>
+
+properties:
+  compatible:
+    const: mscc,zl38060
+
+  reg:
+    description:
+      SPI device address.
+    maxItems: 1
+
+  spi-max-frequency:
+    maximum: 24000000
+
+  reset-gpios:
+    description:
+      A GPIO line handling reset of the chip. As the line is active low,
+      it should be marked GPIO_ACTIVE_LOW (see ../gpio/gpio.txt)
+    maxItems: 1
+
+  '#gpio-cells':
+    const: 2
+
+  gpio-controller: true
+
+  '#sound-dai-cells':
+    const: 0
+
+required:
+  - compatible
+  - reg
+  - '#gpio-cells'
+  - gpio-controller
+  - '#sound-dai-cells'
+
+additionalProperties: false
+
+examples:
+  - |
+    #include <dt-bindings/gpio/gpio.h>
+    spi0 {
+        #address-cells = <1>;
+        #size-cells = <0>;
+
+        codec: zl38060@0 {
+            gpio-controller;
+            #gpio-cells = <2>;
+            #sound-dai-cells = <0>;
+            compatible = "mscc,zl38060";
+            reg = <0>;
+            spi-max-frequency = <12000000>;
+            reset-gpios = <&gpio1 0 GPIO_ACTIVE_LOW>;
+        };
+    };

drivers/soundwire/intel.c

@@ -669,11 +669,11 @@ static int sdw_stream_setup(struct snd_pcm_substream *substream,
 
 	/* Set stream pointer on all CODEC DAIs */
 	for (i = 0; i < rtd->num_codecs; i++) {
-		ret = snd_soc_dai_set_sdw_stream(rtd->codec_dais[i], sdw_stream,
+		ret = snd_soc_dai_set_sdw_stream(asoc_rtd_to_codec(rtd, i), sdw_stream,
 						 substream->stream);
 		if (ret < 0) {
 			dev_err(dai->dev, "failed to set stream pointer on codec dai %s",
-				rtd->codec_dais[i]->name);
+				asoc_rtd_to_codec(rtd, i)->name);
 			goto release_stream;
 		}
 	}

include/sound/control.h

@@ -75,7 +75,7 @@ struct snd_kcontrol {
 	unsigned long private_value;
 	void *private_data;
 	void (*private_free)(struct snd_kcontrol *kcontrol);
-	struct snd_kcontrol_volatile vd[0];	/* volatile data */
+	struct snd_kcontrol_volatile vd[];	/* volatile data */
 };
 
 #define snd_kcontrol(n) list_entry(n, struct snd_kcontrol, list)

include/sound/hda_codec.h

@@ -288,6 +288,10 @@ struct hda_codec {
 #define dev_to_hda_codec(_dev)	container_of(_dev, struct hda_codec, core.dev)
 #define hda_codec_dev(_dev)	(&(_dev)->core.dev)
 
+#define hdac_to_hda_priv(_hdac) \
+			container_of(_hdac, struct hdac_hda_priv, codec.core)
+#define hdac_to_hda_codec(_hdac) container_of(_hdac, struct hda_codec, core)
+
 #define list_for_each_codec(c, bus) \
 	list_for_each_entry(c, &(bus)->core.codec_list, core.list)
 #define list_for_each_codec_safe(c, n, bus)				\
@@ -362,13 +366,6 @@ struct hda_verb {
 void snd_hda_sequence_write(struct hda_codec *codec,
 			    const struct hda_verb *seq);
 
-/* unsolicited event */
-static inline void
-snd_hda_queue_unsol_event(struct hda_bus *bus, u32 res, u32 res_ex)
-{
-	snd_hdac_bus_queue_event(&bus->core, res, res_ex);
-}
-
 /* cached write */
 static inline int
 snd_hda_codec_write_cache(struct hda_codec *codec, hda_nid_t nid,

include/sound/hdaudio.h

@@ -207,8 +207,8 @@ static inline int snd_hdac_power_down_pm(struct hdac_device *codec) { return 0;
 static inline int snd_hdac_keep_power_up(struct hdac_device *codec) { return 0; }
 static inline void snd_hdac_enter_pm(struct hdac_device *codec) {}
 static inline void snd_hdac_leave_pm(struct hdac_device *codec) {}
-static inline bool snd_hdac_is_in_pm(struct hdac_device *codec) { return 0; }
-static inline bool snd_hdac_is_power_on(struct hdac_device *codec) { return 1; }
+static inline bool snd_hdac_is_in_pm(struct hdac_device *codec) { return false; }
+static inline bool snd_hdac_is_power_on(struct hdac_device *codec) { return true; }
 #endif
 
 /*
@@ -364,16 +364,16 @@ struct hdac_bus {
 	/* link management */
 	struct list_head hlink_list;
 	bool cmd_dma_state;
+
+	/* factor used to derive STRIPE control value */
+	unsigned int sdo_limit;
 };
 
 int snd_hdac_bus_init(struct hdac_bus *bus, struct device *dev,
 		      const struct hdac_bus_ops *ops);
 void snd_hdac_bus_exit(struct hdac_bus *bus);
-int snd_hdac_bus_exec_verb(struct hdac_bus *bus, unsigned int addr,
-			   unsigned int cmd, unsigned int *res);
 int snd_hdac_bus_exec_verb_unlocked(struct hdac_bus *bus, unsigned int addr,
 				    unsigned int cmd, unsigned int *res);
-void snd_hdac_bus_queue_event(struct hdac_bus *bus, u32 res, u32 res_ex);
 
 static inline void snd_hdac_codec_link_up(struct hdac_device *codec)
 {

include/sound/intel-nhlt.h

@@ -50,7 +50,7 @@ enum nhlt_device_type {
 
 struct nhlt_specific_cfg {
 	u32 size;
-	u8 caps[0];
+	u8 caps[];
 } __packed;
 
 struct nhlt_fmt_cfg {
@@ -60,7 +60,7 @@ struct nhlt_fmt_cfg {
 
 struct nhlt_fmt {
 	u8 fmt_count;
-	struct nhlt_fmt_cfg fmt_config[0];
+	struct nhlt_fmt_cfg fmt_config[];
 } __packed;
 
 struct nhlt_endpoint {
@@ -80,7 +80,7 @@ struct nhlt_endpoint {
 struct nhlt_acpi_table {
 	struct acpi_table_header header;
 	u8 endpoint_count;
-	struct nhlt_endpoint desc[0];
+	struct nhlt_endpoint desc[];
 } __packed;
 
 struct nhlt_resource_desc  {

include/sound/soc-acpi.h

-/* SPDX-License-Identifier: GPL-2.0
+/* SPDX-License-Identifier: GPL-2.0-only
  *
  * Copyright (C) 2013-15, Intel Corporation. All rights reserved.
  */

include/sound/soc-card.h

+/* SPDX-License-Identifier: GPL-2.0
+ *
+ * soc-card.h
+ *
+ * Copyright (C) 2019 Renesas Electronics Corp.
+ * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+ */
+#ifndef __SOC_CARD_H
+#define __SOC_CARD_H
+
+enum snd_soc_card_subclass {
+	SND_SOC_CARD_CLASS_INIT		= 0,
+	SND_SOC_CARD_CLASS_RUNTIME	= 1,
+};
+
+struct snd_kcontrol *snd_soc_card_get_kcontrol(struct snd_soc_card *soc_card,
+					       const char *name);
+int snd_soc_card_jack_new(struct snd_soc_card *card, const char *id, int type,
+			  struct snd_soc_jack *jack,
+			  struct snd_soc_jack_pin *pins, unsigned int num_pins);
+
+int snd_soc_card_suspend_pre(struct snd_soc_card *card);
+int snd_soc_card_suspend_post(struct snd_soc_card *card);
+int snd_soc_card_resume_pre(struct snd_soc_card *card);
+int snd_soc_card_resume_post(struct snd_soc_card *card);
+
+int snd_soc_card_probe(struct snd_soc_card *card);
+int snd_soc_card_late_probe(struct snd_soc_card *card);
+int snd_soc_card_remove(struct snd_soc_card *card);
+
+int snd_soc_card_set_bias_level(struct snd_soc_card *card,
+				struct snd_soc_dapm_context *dapm,
+				enum snd_soc_bias_level level);
+int snd_soc_card_set_bias_level_post(struct snd_soc_card *card,
+				     struct snd_soc_dapm_context *dapm,
+				     enum snd_soc_bias_level level);
+
+int snd_soc_card_add_dai_link(struct snd_soc_card *card,
+			      struct snd_soc_dai_link *dai_link);
+void snd_soc_card_remove_dai_link(struct snd_soc_card *card,
+				  struct snd_soc_dai_link *dai_link);
+
+/* device driver data */
+static inline void snd_soc_card_set_drvdata(struct snd_soc_card *card,
+					    void *data)
+{
+	card->drvdata = data;
+}
+
+static inline void *snd_soc_card_get_drvdata(struct snd_soc_card *card)
+{
+	return card->drvdata;
+}
+
+static inline
+struct snd_soc_dai *snd_soc_card_get_codec_dai(struct snd_soc_card *card,
+					       const char *dai_name)
+{
+	struct snd_soc_pcm_runtime *rtd;
+
+	for_each_card_rtds(card, rtd) {
+		if (!strcmp(asoc_rtd_to_codec(rtd, 0)->name, dai_name))
+			return asoc_rtd_to_codec(rtd, 0);
+	}
+
+	return NULL;
+}
+
+#endif /* __SOC_CARD_H */

include/sound/soc-component.h

@@ -25,6 +25,44 @@
 	     order++)
 
 /* component interface */
+struct snd_compress_ops {
+	int (*open)(struct snd_soc_component *component,
+		    struct snd_compr_stream *stream);
+	int (*free)(struct snd_soc_component *component,
+		    struct snd_compr_stream *stream);
+	int (*set_params)(struct snd_soc_component *component,
+			  struct snd_compr_stream *stream,
+			  struct snd_compr_params *params);
+	int (*get_params)(struct snd_soc_component *component,
+			  struct snd_compr_stream *stream,
+			  struct snd_codec *params);
+	int (*set_metadata)(struct snd_soc_component *component,
+			    struct snd_compr_stream *stream,
+			    struct snd_compr_metadata *metadata);
+	int (*get_metadata)(struct snd_soc_component *component,
+			    struct snd_compr_stream *stream,
+			    struct snd_compr_metadata *metadata);
+	int (*trigger)(struct snd_soc_component *component,
+		       struct snd_compr_stream *stream, int cmd);
+	int (*pointer)(struct snd_soc_component *component,
+		       struct snd_compr_stream *stream,
+		       struct snd_compr_tstamp *tstamp);
+	int (*copy)(struct snd_soc_component *component,
+		    struct snd_compr_stream *stream, char __user *buf,
+		    size_t count);
+	int (*mmap)(struct snd_soc_component *component,
+		    struct snd_compr_stream *stream,
+		    struct vm_area_struct *vma);
+	int (*ack)(struct snd_soc_component *component,
+		   struct snd_compr_stream *stream, size_t bytes);
+	int (*get_caps)(struct snd_soc_component *component,
+			struct snd_compr_stream *stream,
+			struct snd_compr_caps *caps);
+	int (*get_codec_caps)(struct snd_soc_component *component,
+			      struct snd_compr_stream *stream,
+			      struct snd_compr_codec_caps *codec);
+};
+
 struct snd_soc_component_driver {
 	const char *name;
 
@@ -108,7 +146,7 @@ struct snd_soc_component_driver {
 		    struct snd_pcm_substream *substream,
 		    struct vm_area_struct *vma);
 
-	const struct snd_compr_ops *compr_ops;
+	const struct snd_compress_ops *compress_ops;
 
 	/* probe ordering - for components with runtime dependencies */
 	int probe_order;
@@ -351,10 +389,10 @@ static inline void *snd_soc_component_get_drvdata(struct snd_soc_component *c)
 	return dev_get_drvdata(c->dev);
 }
 
-static inline bool snd_soc_component_is_active(
-	struct snd_soc_component *component)
+static inline unsigned int
+snd_soc_component_active(struct snd_soc_component *component)
 {
-	return component->active != 0;
+	return component->active;
 }
 
 /* component pin */

include/sound/soc-dai.h

@@ -154,21 +154,59 @@ int snd_soc_dai_startup(struct snd_soc_dai *dai,
 			struct snd_pcm_substream *substream);
 void snd_soc_dai_shutdown(struct snd_soc_dai *dai,
 			  struct snd_pcm_substream *substream);
-int snd_soc_dai_prepare(struct snd_soc_dai *dai,
-			struct snd_pcm_substream *substream);
-int snd_soc_dai_trigger(struct snd_soc_dai *dai,
-			struct snd_pcm_substream *substream, int cmd);
-int snd_soc_dai_bespoke_trigger(struct snd_soc_dai *dai,
-			struct snd_pcm_substream *substream, int cmd);
 snd_pcm_sframes_t snd_soc_dai_delay(struct snd_soc_dai *dai,
 				    struct snd_pcm_substream *substream);
 void snd_soc_dai_suspend(struct snd_soc_dai *dai);
 void snd_soc_dai_resume(struct snd_soc_dai *dai);
-int snd_soc_dai_probe(struct snd_soc_dai *dai);
-int snd_soc_dai_remove(struct snd_soc_dai *dai);
 int snd_soc_dai_compress_new(struct snd_soc_dai *dai,
 			     struct snd_soc_pcm_runtime *rtd, int num);
 bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int stream);
+void snd_soc_dai_action(struct snd_soc_dai *dai,
+			int stream, int action);
+static inline void snd_soc_dai_activate(struct snd_soc_dai *dai,
+					int stream)
+{
+	snd_soc_dai_action(dai, stream,  1);
+}
+static inline void snd_soc_dai_deactivate(struct snd_soc_dai *dai,
+					  int stream)
+{
+	snd_soc_dai_action(dai, stream, -1);
+}
+int snd_soc_dai_active(struct snd_soc_dai *dai);
+
+int snd_soc_pcm_dai_probe(struct snd_soc_pcm_runtime *rtd, int order);
+int snd_soc_pcm_dai_remove(struct snd_soc_pcm_runtime *rtd, int order);
+int snd_soc_pcm_dai_new(struct snd_soc_pcm_runtime *rtd);
+int snd_soc_pcm_dai_prepare(struct snd_pcm_substream *substream);
+int snd_soc_pcm_dai_trigger(struct snd_pcm_substream *substream, int cmd);
+int snd_soc_pcm_dai_bespoke_trigger(struct snd_pcm_substream *substream,
+				    int cmd);
+
+int snd_soc_dai_compr_startup(struct snd_soc_dai *dai,
+			      struct snd_compr_stream *cstream);
+void snd_soc_dai_compr_shutdown(struct snd_soc_dai *dai,
+				struct snd_compr_stream *cstream);
+int snd_soc_dai_compr_trigger(struct snd_soc_dai *dai,
+			      struct snd_compr_stream *cstream, int cmd);
+int snd_soc_dai_compr_set_params(struct snd_soc_dai *dai,
+				 struct snd_compr_stream *cstream,
+				 struct snd_compr_params *params);
+int snd_soc_dai_compr_get_params(struct snd_soc_dai *dai,
+				 struct snd_compr_stream *cstream,
+				 struct snd_codec *params);
+int snd_soc_dai_compr_ack(struct snd_soc_dai *dai,
+			  struct snd_compr_stream *cstream,
+			  size_t bytes);
+int snd_soc_dai_compr_pointer(struct snd_soc_dai *dai,
+			      struct snd_compr_stream *cstream,
+			      struct snd_compr_tstamp *tstamp);
+int snd_soc_dai_compr_set_metadata(struct snd_soc_dai *dai,
+				   struct snd_compr_stream *cstream,
+				   struct snd_compr_metadata *metadata);
+int snd_soc_dai_compr_get_metadata(struct snd_soc_dai *dai,
+				   struct snd_compr_stream *cstream,
+				   struct snd_compr_metadata *metadata);
 
 struct snd_soc_dai_ops {
 	/*
@@ -326,8 +364,6 @@ struct snd_soc_dai {
 	/* DAI runtime info */
 	unsigned int stream_active[SNDRV_PCM_STREAM_LAST + 1]; /* usage count */
 
-	unsigned int active;
-
 	struct snd_soc_dapm_widget *playback_widget;
 	struct snd_soc_dapm_widget *capture_widget;
 
@@ -443,4 +479,10 @@ static inline void *snd_soc_dai_get_sdw_stream(struct snd_soc_dai *dai,
 		return ERR_PTR(-ENOTSUPP);
 }
 
+static inline unsigned int
+snd_soc_dai_stream_active(struct snd_soc_dai *dai, int stream)
+{
+	return dai->stream_active[stream];
+}
+
 #endif

include/sound/soc-dapm.h

@@ -689,7 +689,7 @@ struct snd_soc_dapm_context {
 /* A list of widgets associated with an object, typically a snd_kcontrol */
 struct snd_soc_dapm_widget_list {
 	int num_widgets;
-	struct snd_soc_dapm_widget *widgets[0];
+	struct snd_soc_dapm_widget *widgets[];
 };
 
 #define for_each_dapm_widgets(list, i, widget)				\

include/sound/soc-link.h

+/* SPDX-License-Identifier: GPL-2.0
+ *
+ * soc-link.h
+ *
+ * Copyright (C) 2019 Renesas Electronics Corp.
+ * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+ */
+#ifndef __SOC_LINK_H
+#define __SOC_LINK_H
+
+int snd_soc_link_init(struct snd_soc_pcm_runtime *rtd);
+int snd_soc_link_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
+				    struct snd_pcm_hw_params *params);
+
+int snd_soc_link_startup(struct snd_pcm_substream *substream);
+void snd_soc_link_shutdown(struct snd_pcm_substream *substream);
+int snd_soc_link_prepare(struct snd_pcm_substream *substream);
+int snd_soc_link_hw_params(struct snd_pcm_substream *substream,
+			   struct snd_pcm_hw_params *params);
+void snd_soc_link_hw_free(struct snd_pcm_substream *substream);
+int snd_soc_link_trigger(struct snd_pcm_substream *substream, int cmd);
+
+int snd_soc_link_compr_startup(struct snd_compr_stream *cstream);
+void snd_soc_link_compr_shutdown(struct snd_compr_stream *cstream);
+int snd_soc_link_compr_set_params(struct snd_compr_stream *cstream);
+
+#endif /* __SOC_LINK_H */

include/sound/soc.h

@@ -414,11 +414,6 @@ enum snd_soc_pcm_subclass {
 	SND_SOC_PCM_CLASS_BE	= 1,
 };
 
-enum snd_soc_card_subclass {
-	SND_SOC_CARD_CLASS_INIT		= 0,
-	SND_SOC_CARD_CLASS_RUNTIME	= 1,
-};
-
 int snd_soc_register_card(struct snd_soc_card *card);
 int snd_soc_unregister_card(struct snd_soc_card *card);
 int devm_snd_soc_register_card(struct device *dev, struct snd_soc_card *card);
@@ -468,8 +463,19 @@ struct snd_soc_pcm_runtime *snd_soc_get_pcm_runtime(struct snd_soc_card *card,
 				struct snd_soc_dai_link *dai_link);
 
 bool snd_soc_runtime_ignore_pmdown_time(struct snd_soc_pcm_runtime *rtd);
-void snd_soc_runtime_activate(struct snd_soc_pcm_runtime *rtd, int stream);
-void snd_soc_runtime_deactivate(struct snd_soc_pcm_runtime *rtd, int stream);
+
+void snd_soc_runtime_action(struct snd_soc_pcm_runtime *rtd,
+			    int stream, int action);
+static inline void snd_soc_runtime_activate(struct snd_soc_pcm_runtime *rtd,
+				     int stream)
+{
+	snd_soc_runtime_action(rtd, stream, 1);
+}
+static inline void snd_soc_runtime_deactivate(struct snd_soc_pcm_runtime *rtd,
+				       int stream)
+{
+	snd_soc_runtime_action(rtd, stream, -1);
+}
 
 int snd_soc_runtime_calc_hw(struct snd_soc_pcm_runtime *rtd,
 			    struct snd_pcm_hardware *hw, int stream);
@@ -498,10 +504,6 @@ int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream,
 	const struct snd_pcm_hardware *hw);
 
 /* Jack reporting */
-int snd_soc_card_jack_new(struct snd_soc_card *card, const char *id, int type,
-	struct snd_soc_jack *jack, struct snd_soc_jack_pin *pins,
-	unsigned int num_pins);
-
 void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask);
 int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count,
 			  struct snd_soc_jack_pin *pins);
@@ -571,8 +573,6 @@ static inline int snd_soc_set_ac97_ops(struct snd_ac97_bus_ops *ops)
 struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template,
 				  void *data, const char *long_name,
 				  const char *prefix);
-struct snd_kcontrol *snd_soc_card_get_kcontrol(struct snd_soc_card *soc_card,
-					       const char *name);
 int snd_soc_add_component_controls(struct snd_soc_component *component,
 	const struct snd_kcontrol_new *controls, unsigned int num_controls);
 int snd_soc_add_card_controls(struct snd_soc_card *soc_card,
@@ -790,9 +790,6 @@ struct snd_soc_dai_link {
 	const struct snd_soc_pcm_stream *params;
 	unsigned int num_params;
 
-	struct snd_soc_dapm_widget *playback_widget;
-	struct snd_soc_dapm_widget *capture_widget;
-
 	unsigned int dai_fmt;           /* format to set on init */
 
 	enum snd_soc_dpcm_trigger trigger[2]; /* trigger type for DPCM */
@@ -809,7 +806,7 @@ struct snd_soc_dai_link {
 	const struct snd_soc_compr_ops *compr_ops;
 
 	/* Mark this pcm with non atomic ops */
-	bool nonatomic;
+	unsigned int nonatomic:1;
 
 	/* For unidirectional dai links */
 	unsigned int playback_only:1;
@@ -1005,9 +1002,6 @@ struct snd_soc_card {
 
 	spinlock_t dpcm_lock;
 
-	bool instantiated;
-	bool topology_shortname_created;
-
 	int (*probe)(struct snd_soc_card *card);
 	int (*late_probe)(struct snd_soc_card *card);
 	int (*remove)(struct snd_soc_card *card);
@@ -1068,8 +1062,6 @@ struct snd_soc_card {
 	int num_of_dapm_widgets;
 	const struct snd_soc_dapm_route *of_dapm_routes;
 	int num_of_dapm_routes;
-	bool fully_routed;
-	bool disable_route_checks;
 
 	/* lists of probed devices belonging to this card */
 	struct list_head component_dev_list;
@@ -1096,6 +1088,13 @@ struct snd_soc_card {
 #endif
 	u32 pop_time;
 
+	/* bit field */
+	unsigned int instantiated:1;
+	unsigned int topology_shortname_created:1;
+	unsigned int fully_routed:1;
+	unsigned int disable_route_checks:1;
+	unsigned int probed:1;
+
 	void *drvdata;
 };
 #define for_each_card_prelinks(card, i, link)				\
@@ -1146,16 +1145,21 @@ struct snd_soc_pcm_runtime {
 	/* runtime devices */
 	struct snd_pcm *pcm;
 	struct snd_compr *compr;
-	struct snd_soc_dai *codec_dai;
-	struct snd_soc_dai *cpu_dai;
-	struct snd_soc_dai **dais;
 
-	struct snd_soc_dai **codec_dais;
+	/*
+	 * dais = cpu_dai + codec_dai
+	 * see
+	 *	soc_new_pcm_runtime()
+	 *	asoc_rtd_to_cpu()
+	 *	asoc_rtd_to_codec()
+	 */
+	struct snd_soc_dai **dais;
 	unsigned int num_codecs;
-
-	struct snd_soc_dai **cpu_dais;
 	unsigned int num_cpus;
 
+	struct snd_soc_dapm_widget *playback_widget;
+	struct snd_soc_dapm_widget *capture_widget;
+
 	struct delayed_work delayed_work;
 	void (*close_delayed_work_func)(struct snd_soc_pcm_runtime *rtd);
 #ifdef CONFIG_DEBUG_FS
@@ -1170,28 +1174,28 @@ struct snd_soc_pcm_runtime {
 	unsigned int fe_compr:1; /* for Dynamic PCM */
 
 	int num_components;
-	struct snd_soc_component *components[0]; /* CPU/Codec/Platform */
+	struct snd_soc_component *components[]; /* CPU/Codec/Platform */
 };
 /* see soc_new_pcm_runtime()  */
 #define asoc_rtd_to_cpu(rtd, n)   (rtd)->dais[n]
 #define asoc_rtd_to_codec(rtd, n) (rtd)->dais[n + (rtd)->num_cpus]
 
 #define for_each_rtd_components(rtd, i, component)			\
-	for ((i) = 0;							\
+	for ((i) = 0, component = NULL;					\
 	     ((i) < rtd->num_components) && ((component) = rtd->components[i]);\
 	     (i)++)
 #define for_each_rtd_cpu_dais(rtd, i, dai)				\
 	for ((i) = 0;							\
-	     ((i) < rtd->num_cpus) && ((dai) = rtd->cpu_dais[i]);	\
+	     ((i) < rtd->num_cpus) && ((dai) = asoc_rtd_to_cpu(rtd, i)); \
 	     (i)++)
 #define for_each_rtd_cpu_dais_rollback(rtd, i, dai)		\
-	for (; (--(i) >= 0) && ((dai) = rtd->cpu_dais[i]);)
+	for (; (--(i) >= 0) && ((dai) = asoc_rtd_to_cpu(rtd, i));)
 #define for_each_rtd_codec_dais(rtd, i, dai)				\
 	for ((i) = 0;							\
-	     ((i) < rtd->num_codecs) && ((dai) = rtd->codec_dais[i]);	\
+	     ((i) < rtd->num_codecs) && ((dai) = asoc_rtd_to_codec(rtd, i)); \
 	     (i)++)
 #define for_each_rtd_codec_dais_rollback(rtd, i, dai)		\
-	for (; (--(i) >= 0) && ((dai) = rtd->codec_dais[i]);)
+	for (; (--(i) >= 0) && ((dai) = asoc_rtd_to_codec(rtd, i));)
 #define for_each_rtd_dais(rtd, i, dai)					\
 	for ((i) = 0;							\
 	     ((i) < (rtd)->num_cpus + (rtd)->num_codecs) &&		\
@@ -1252,29 +1256,16 @@ struct soc_enum {
 #endif
 };
 
-/* device driver data */
-
-static inline void snd_soc_card_set_drvdata(struct snd_soc_card *card,
-		void *data)
-{
-	card->drvdata = data;
-}
-
-static inline void *snd_soc_card_get_drvdata(struct snd_soc_card *card)
-{
-	return card->drvdata;
-}
-
 static inline bool snd_soc_volsw_is_stereo(struct soc_mixer_control *mc)
 {
 	if (mc->reg == mc->rreg && mc->shift == mc->rshift)
-		return 0;
+		return false;
 	/*
 	 * mc->reg == mc->rreg && mc->shift != mc->rshift, or
 	 * mc->reg != mc->rreg means that the control is
 	 * stereo (bits in one register or in two registers)
 	 */
-	return 1;
+	return true;
 }
 
 static inline unsigned int snd_soc_enum_val_to_item(struct soc_enum *e,
@@ -1377,20 +1368,6 @@ struct snd_soc_dai *snd_soc_find_dai(
 
 #include <sound/soc-dai.h>
 
-static inline
-struct snd_soc_dai *snd_soc_card_get_codec_dai(struct snd_soc_card *card,
-					       const char *dai_name)
-{
-	struct snd_soc_pcm_runtime *rtd;
-
-	list_for_each_entry(rtd, &card->rtd_list, list) {
-		if (!strcmp(rtd->codec_dai->name, dai_name))
-			return rtd->codec_dai;
-	}
-
-	return NULL;
-}
-
 static inline
 int snd_soc_fixup_dai_links_platform_name(struct snd_soc_card *card,
 					  const char *platform_name)
@@ -1436,5 +1413,6 @@ static inline void snd_soc_dapm_mutex_unlock(struct snd_soc_dapm_context *dapm)
 }
 
 #include <sound/soc-component.h>
+#include <sound/soc-card.h>
 
 #endif

include/sound/sof.h

-/* SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) */
+/* SPDX-License-Identifier: (GPL-2.0-only 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.
@@ -27,6 +27,9 @@ struct snd_sof_pdata {
 
 	struct device *dev;
 
+	/* indicate how many first bytes shouldn't be loaded into DSP memory. */
+	size_t fw_offset;
+
 	/*
 	 * notification callback used if the hardware initialization
 	 * can take time or is handled in a workqueue. This callback

include/sound/sof/channel_map.h

-/* SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) */
+/* SPDX-License-Identifier: (GPL-2.0-only 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.

include/sound/sof/control.h

-/* SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) */
+/* SPDX-License-Identifier: (GPL-2.0-only 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.

include/sound/sof/dai-imx.h

-/* SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) */
+/* SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) */
 /*
  * Copyright 2019 NXP
  *

include/sound/sof/dai-intel.h

-/* SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) */
+/* SPDX-License-Identifier: (GPL-2.0-only 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.
@@ -49,6 +49,9 @@
 /* bclk idle */
 #define SOF_DAI_INTEL_SSP_CLKCTRL_BCLK_IDLE_HIGH	BIT(5)
 
+/* DMIC max. four controllers for eight microphone channels */
+#define SOF_DAI_INTEL_DMIC_NUM_CTRL			4
+
 /* SSP Configuration Request - SOF_IPC_DAI_SSP_CONFIG */
 struct sof_ipc_dai_ssp_params {
 	struct sof_ipc_hdr hdr;
@@ -85,15 +88,19 @@ struct sof_ipc_dai_ssp_params {
 struct sof_ipc_dai_hda_params {
 	struct sof_ipc_hdr hdr;
 	uint32_t link_dma_ch;
+	uint32_t rate;
+	uint32_t channels;
 } __packed;
 
 /* ALH Configuration Request - SOF_IPC_DAI_ALH_CONFIG */
 struct sof_ipc_dai_alh_params {
 	struct sof_ipc_hdr hdr;
 	uint32_t stream_id;
+	uint32_t rate;
+	uint32_t channels;
 
 	/* reserved for future use */
-	uint32_t reserved[15];
+	uint32_t reserved[13];
 } __packed;
 
 /* DMIC Configuration Request - SOF_IPC_DAI_DMIC_CONFIG */
@@ -135,7 +142,7 @@ struct sof_ipc_dai_dmic_pdm_ctrl {
  * version number used in configuration data is checked vs. version used by
  * device driver src/drivers/dmic.c need to match. It is incremented from
  * initial value 1 if updates done for the to driver would alter the operation
- * of the microhone.
+ * of the microphone.
  *
  * Note: The microphone clock (pdmclk_min, pdmclk_max, duty_min, duty_max)
  * parameters need to be set as defined in microphone data sheet. E.g. clock
@@ -170,12 +177,13 @@ struct sof_ipc_dai_dmic_params {
 	uint32_t fifo_fs;	/**< FIFO sample rate in Hz (8000..96000) */
 	uint32_t reserved_1;	/**< Reserved */
 	uint16_t fifo_bits;	/**< FIFO word length (16 or 32) */
-	uint16_t reserved_2;	/**< Reserved */
+	uint16_t fifo_bits_b;	/**< Deprecated since firmware ABI 3.0.1 */
 
 	uint16_t duty_min;	/**< Min. mic clock duty cycle in % (20..80) */
 	uint16_t duty_max;	/**< Max. mic clock duty cycle in % (min..80) */
 
-	uint32_t num_pdm_active; /**< Number of active pdm controllers */
+	uint32_t num_pdm_active; /**< Number of active pdm controllers. */
+				 /**< Range is 1..SOF_DAI_INTEL_DMIC_NUM_CTRL */
 
 	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) */
@@ -184,8 +192,8 @@ struct sof_ipc_dai_dmic_params {
 	/* reserved for future use */
 	uint32_t reserved[5];
 
-	/**< variable number of pdm controller config */
-	struct sof_ipc_dai_dmic_pdm_ctrl pdm[0];
+	/**< PDM controllers configuration */
+	struct sof_ipc_dai_dmic_pdm_ctrl pdm[SOF_DAI_INTEL_DMIC_NUM_CTRL];
 } __packed;
 
 #endif

include/sound/sof/dai.h

-/* SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) */
+/* SPDX-License-Identifier: (GPL-2.0-only 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.

include/sound/sof/ext_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) 2020 Intel Corporation. All rights reserved.
+ */
+
+/*
+ * Extended manifest is a place to store metadata about firmware, known during
+ * compilation time - for example firmware version or used compiler.
+ * Given information are read on host side before firmware startup.
+ * This part of output binary is not signed.
+ */
+
+#ifndef __SOF_FIRMWARE_EXT_MANIFEST_H__
+#define __SOF_FIRMWARE_EXT_MANIFEST_H__
+
+#include <linux/bits.h>
+#include <linux/compiler.h>
+#include <linux/types.h>
+#include <sound/sof/info.h>
+
+/* In ASCII `XMan` */
+#define SOF_EXT_MAN_MAGIC_NUMBER	0x6e614d58
+
+/* Build u32 number in format MMmmmppp */
+#define SOF_EXT_MAN_BUILD_VERSION(MAJOR, MINOR, PATH) ((uint32_t)( \
+	((MAJOR) << 24) | \
+	((MINOR) << 12) | \
+	(PATH)))
+
+/* check extended manifest version consistency */
+#define SOF_EXT_MAN_VERSION_INCOMPATIBLE(host_ver, cli_ver) ( \
+	((host_ver) & GENMASK(31, 24)) != \
+	((cli_ver) & GENMASK(31, 24)))
+
+/* used extended manifest header version */
+#define SOF_EXT_MAN_VERSION		SOF_EXT_MAN_BUILD_VERSION(1, 0, 0)
+
+/* extended manifest header, deleting any field breaks backward compatibility */
+struct sof_ext_man_header {
+	uint32_t magic;		/*< identification number, */
+				/*< EXT_MAN_MAGIC_NUMBER */
+	uint32_t full_size;	/*< [bytes] full size of ext_man, */
+				/*< (header + content + padding) */
+	uint32_t header_size;	/*< [bytes] makes header extensionable, */
+				/*< after append new field to ext_man header */
+				/*< then backward compatible won't be lost */
+	uint32_t header_version; /*< value of EXT_MAN_VERSION */
+				/*< not related with following content */
+
+	/* just after this header should be list of ext_man_elem_* elements */
+} __packed;
+
+/* Now define extended manifest elements */
+
+/* Extended manifest elements types */
+enum sof_ext_man_elem_type {
+	SOF_EXT_MAN_ELEM_FW_VERSION		= 0,
+	SOF_EXT_MAN_ELEM_WINDOW			= SOF_IPC_EXT_WINDOW,
+	SOF_EXT_MAN_ELEM_CC_VERSION		= SOF_IPC_EXT_CC_INFO,
+};
+
+/* extended manifest element header */
+struct sof_ext_man_elem_header {
+	uint32_t type;		/*< SOF_EXT_MAN_ELEM_ */
+	uint32_t size;		/*< in bytes, including header size */
+
+	/* just after this header should be type dependent content */
+} __packed;
+
+/* FW version */
+struct sof_ext_man_fw_version {
+	struct sof_ext_man_elem_header hdr;
+	/* use sof_ipc struct because of code re-use */
+	struct sof_ipc_fw_version version;
+	uint32_t flags;
+} __packed;
+
+/* extended data memory windows for IPC, trace and debug */
+struct sof_ext_man_window {
+	struct sof_ext_man_elem_header hdr;
+	/* use sof_ipc struct because of code re-use */
+	struct sof_ipc_window ipc_window;
+} __packed;
+
+/* Used C compiler description */
+struct sof_ext_man_cc_version {
+	struct sof_ext_man_elem_header hdr;
+	/* use sof_ipc struct because of code re-use */
+	struct sof_ipc_cc_version cc_version;
+} __packed;
+
+#endif /* __SOF_FIRMWARE_EXT_MANIFEST_H__ */

include/sound/sof/header.h

-/* SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) */
+/* SPDX-License-Identifier: (GPL-2.0-only 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.

include/sound/sof/info.h

-/* SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) */
+/* SPDX-License-Identifier: (GPL-2.0-only 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.
@@ -31,6 +31,8 @@ enum sof_ipc_ext_data {
 	SOF_IPC_EXT_UNUSED		= 0,
 	SOF_IPC_EXT_WINDOW		= 1,
 	SOF_IPC_EXT_CC_INFO		= 2,
+	SOF_IPC_EXT_PROBE_INFO		= 3,
+	SOF_IPC_EXT_USER_ABI_INFO	= 4,
 };
 
 /* FW version - SOF_IPC_GLB_VERSION */
@@ -109,9 +111,27 @@ struct sof_ipc_cc_version {
 	/* reserved for future use */
 	uint32_t reserved[4];
 
-	char name[16]; /* null terminated compiler name */
-	char optim[4]; /* null terminated compiler -O flag value */
-	char desc[]; /* null terminated compiler description */
+	uint8_t name[16]; /* null terminated compiler name */
+	uint8_t optim[4]; /* null terminated compiler -O flag value */
+	uint8_t desc[32]; /* null terminated compiler description */
 } __packed;
 
+/* extended data: Probe setup */
+struct sof_ipc_probe_support {
+	struct sof_ipc_ext_data_hdr ext_hdr;
+
+	uint32_t probe_points_max;
+	uint32_t injection_dmas_max;
+
+	/* reserved for future use */
+	uint32_t reserved[2];
+} __packed;
+
+/* extended data: user abi version(s) */
+struct sof_ipc_user_abi_version {
+	struct sof_ipc_ext_data_hdr ext_hdr;
+
+	uint32_t abi_dbg_version;
+}  __packed;
+
 #endif

include/sound/sof/pm.h

-/* SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) */
+/* SPDX-License-Identifier: (GPL-2.0-only 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.

include/sound/sof/stream.h

-/* SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) */
+/* SPDX-License-Identifier: (GPL-2.0-only 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.

include/sound/sof/topology.h

-/* SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) */
+/* SPDX-License-Identifier: (GPL-2.0-only 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.
@@ -37,6 +37,8 @@ enum sof_comp_type {
 	SOF_COMP_SELECTOR,		/**< channel selector component */
 	SOF_COMP_DEMUX,
 	SOF_COMP_ASRC,		/**< Asynchronous sample rate converter */
+	SOF_COMP_DCBLOCK,
+	SOF_COMP_SMART_AMP,             /**< smart amplifier component */
 	/* 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 */
@@ -75,11 +77,23 @@ struct sof_ipc_comp {
 #define SOF_MEM_CAPS_CACHE			(1 << 6) /**< cacheable */
 #define SOF_MEM_CAPS_EXEC			(1 << 7) /**< executable */
 
+/*
+ * overrun will cause ring buffer overwrite, instead of XRUN.
+ */
+#define SOF_BUF_OVERRUN_PERMITTED	BIT(0)
+
+/*
+ * underrun will cause readback of 0s, instead of XRUN.
+ */
+#define SOF_BUF_UNDERRUN_PERMITTED	BIT(1)
+
 /* create new component buffer - SOF_IPC_TPLG_BUFFER_NEW */
 struct sof_ipc_buffer {
 	struct sof_ipc_comp comp;
 	uint32_t size;		/**< buffer size in bytes */
 	uint32_t caps;		/**< SOF_MEM_CAPS_ */
+	uint32_t flags;		/**< SOF_BUF_ flags defined above */
+	uint32_t reserved;	/**< reserved for future use */
 } __packed;
 
 /* generic component config data - must always be after struct sof_ipc_comp */
@@ -206,6 +220,8 @@ enum sof_ipc_process_type {
 	SOF_PROCESS_CHAN_SELECTOR,	/**< Channel Selector */
 	SOF_PROCESS_MUX,
 	SOF_PROCESS_DEMUX,
+	SOF_PROCESS_DCBLOCK,
+	SOF_PROCESS_SMART_AMP,	/**< Smart Amplifier */
 };
 
 /* generic "effect", "codec" or proprietary processing component */
@@ -218,7 +234,7 @@ struct sof_ipc_comp_process {
 	/* reserved for future use */
 	uint32_t reserved[7];
 
-	unsigned char data[0];
+	uint8_t data[0];
 } __packed;
 
 /* frees components, buffers and pipelines

include/sound/sof/trace.h

-/* SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) */
+/* SPDX-License-Identifier: (GPL-2.0-only 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.
@@ -72,7 +72,7 @@ struct sof_ipc_dma_trace_posn {
 struct sof_ipc_panic_info {
 	struct sof_ipc_hdr hdr;
 	uint32_t code;			/* SOF_IPC_PANIC_ */
-	char filename[SOF_TRACE_FILENAME_SIZE];
+	uint8_t filename[SOF_TRACE_FILENAME_SIZE];
 	uint32_t linenum;
 }  __packed;
 

include/sound/sof/xtensa.h

-/* SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) */
+/* SPDX-License-Identifier: (GPL-2.0-only 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.

include/uapi/sound/skl-tplg-interface.h

@@ -18,6 +18,8 @@
  */
 #define SKL_CONTROL_TYPE_BYTE_TLV	0x100
 #define SKL_CONTROL_TYPE_MIC_SELECT	0x102
+#define SKL_CONTROL_TYPE_MULTI_IO_SELECT	0x103
+#define SKL_CONTROL_TYPE_MULTI_IO_SELECT_DMIC	0x104
 
 #define HDA_SST_CFG_MAX	900 /* size of copier cfg*/
 #define MAX_IN_QUEUE 8

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 13
+#define SOF_ABI_MINOR 16
 #define SOF_ABI_PATCH 0
 
 /* SOF ABI version number. Format within 32bit word is MMmmmppp */

include/uapi/sound/sof/tokens.h

@@ -126,4 +126,12 @@
 #define SOF_TKN_MUTE_LED_USE			1300
 #define SOF_TKN_MUTE_LED_DIRECTION		1301
 
+/* ALH */
+#define SOF_TKN_INTEL_ALH_RATE			1400
+#define SOF_TKN_INTEL_ALH_CH			1401
+
+/* HDA */
+#define SOF_TKN_INTEL_HDA_RATE			1500
+#define SOF_TKN_INTEL_HDA_CH			1501
+
 #endif

sound/core/oss/pcm_plugin.h

@@ -64,7 +64,7 @@ struct snd_pcm_plugin {
 	char *buf;
 	snd_pcm_uframes_t buf_frames;
 	struct snd_pcm_plugin_channel *buf_channels;
-	char extra_data[0];
+	char extra_data[];
 };
 
 int snd_pcm_plugin_build(struct snd_pcm_substream *handle,

sound/core/seq/oss/seq_oss_timer.h

@@ -44,14 +44,4 @@ snd_seq_oss_timer_cur_tick(struct seq_oss_timer *timer)
 	return timer->cur_tick;
 }
 
-
-/*
- * is realtime event?
- */
-static inline int
-snd_seq_oss_timer_is_realtime(struct seq_oss_timer *timer)
-{
-	return timer->realtime;
-}
-
 #endif

sound/drivers/portman2x4.c

@@ -457,7 +457,7 @@ static int portman_probe(struct parport *p)
 
 	/* Set for RXDATA0 where no damage will be done. */
 	/* 5 */
-	parport_write_control(p, RXDATA0 + STROBE);	/* Write Strobe=1 to command reg. */
+	parport_write_control(p, RXDATA0 | STROBE);	/* Write Strobe=1 to command reg. */
 
 	/* 6 */
 	if ((parport_read_status(p) & ESTB) != ESTB)

sound/firewire/Kconfig

@@ -150,8 +150,12 @@ config SND_FIREWIRE_MOTU
 	 Say Y here to enable support for FireWire devices which MOTU produced:
 	  * 828mk2
 	  * Traveler
-	  * 828mk3
+	  * Ultralite
+	  * 8pre
+	  * 828mk3 (FireWire only)
+	  * 828mk3 (Hybrid)
 	  * Audio Express
+	  * 4pre
 
 	 To compile this driver as a module, choose M here: the module
 	 will be called snd-firewire-motu.
@@ -164,6 +168,8 @@ config SND_FIREFACE
 	 Say Y here to include support for RME fireface series.
 	  * Fireface 400
 	  * Fireface 800
+	  * Fireface UFX
 	  * Fireface UCX
+	  * Fireface 802
 
 endif # SND_FIREWIRE

sound/firewire/amdtp-am824.c

@@ -82,7 +82,8 @@ int amdtp_am824_set_parameters(struct amdtp_stream *s, unsigned int rate,
 	if (err < 0)
 		return err;
 
-	s->ctx_data.rx.fdf = AMDTP_FDF_AM824 | s->sfc;
+	if (s->direction == AMDTP_OUT_STREAM)
+		s->ctx_data.rx.fdf = AMDTP_FDF_AM824 | s->sfc;
 
 	p->pcm_channels = pcm_channels;
 	p->midi_ports = midi_ports;

sound/firewire/amdtp-stream.h

@@ -108,6 +108,8 @@ typedef unsigned int (*amdtp_stream_process_ctx_payloads_t)(
 						const struct pkt_desc *desc,
 						unsigned int packets,
 						struct snd_pcm_substream *pcm);
+
+struct amdtp_domain;
 struct amdtp_stream {
 	struct fw_unit *unit;
 	enum cip_flags flags;
@@ -136,9 +138,7 @@ struct amdtp_stream {
 		struct {
 			// To calculate CIP data blocks and tstamp.
 			unsigned int transfer_delay;
-			unsigned int data_block_state;
-			unsigned int last_syt_offset;
-			unsigned int syt_offset_state;
+			unsigned int seq_index;
 
 			// To generate CIP header.
 			unsigned int fdf;
@@ -180,6 +180,7 @@ struct amdtp_stream {
 	int channel;
 	int speed;
 	struct list_head list;
+	struct amdtp_domain *domain;
 };
 
 int amdtp_stream_init(struct amdtp_stream *s, struct fw_unit *unit,
@@ -273,6 +274,11 @@ static inline bool amdtp_stream_wait_callback(struct amdtp_stream *s,
 				  msecs_to_jiffies(timeout)) > 0;
 }
 
+struct seq_desc {
+	unsigned int syt_offset;
+	unsigned int data_blocks;
+};
+
 struct amdtp_domain {
 	struct list_head streams;
 
@@ -280,6 +286,14 @@ struct amdtp_domain {
 	unsigned int events_per_buffer;
 
 	struct amdtp_stream *irq_target;
+
+	struct seq_desc *seq_descs;
+	unsigned int seq_size;
+	unsigned int seq_tail;
+
+	unsigned int data_block_state;
+	unsigned int syt_offset_state;
+	unsigned int last_syt_offset;
 };
 
 int amdtp_domain_init(struct amdtp_domain *d);

sound/firewire/fireface/ff-protocol-latter.c

@@ -16,7 +16,8 @@
 #define LATTER_SYNC_STATUS	0x0000801c0000ULL
 
 static int parse_clock_bits(u32 data, unsigned int *rate,
-			    enum snd_ff_clock_src *src)
+			    enum snd_ff_clock_src *src,
+			    enum snd_ff_unit_version unit_version)
 {
 	static const struct {
 		unsigned int rate;
@@ -43,6 +44,11 @@ static int parse_clock_bits(u32 data, unsigned int *rate,
 	};
 	int i;
 
+	if (unit_version != SND_FF_UNIT_VERSION_UCX) {
+		// e.g. 0x00fe0f20 but expected 0x00eff002.
+		data = ((data & 0xf0f0f0f0) >> 4) | ((data & 0x0f0f0f0f) << 4);
+	}
+
 	for (i = 0; i < ARRAY_SIZE(rate_entries); ++i) {
 		rate_entry = rate_entries + i;
 		if ((data & 0x0f000000) == rate_entry->flag) {
@@ -79,7 +85,7 @@ static int latter_get_clock(struct snd_ff *ff, unsigned int *rate,
 		return err;
 	data = le32_to_cpu(reg);
 
-	return parse_clock_bits(data, rate, src);
+	return parse_clock_bits(data, rate, src, ff->unit_version);
 }
 
 static int latter_switch_fetching_mode(struct snd_ff *ff, bool enable)
@@ -107,18 +113,18 @@ static int latter_allocate_resources(struct snd_ff *ff, unsigned int rate)
 	int err;
 
 	// Set the number of data blocks transferred in a second.
-	if (rate % 32000 == 0)
-		code = 0x00;
+	if (rate % 48000 == 0)
+		code = 0x04;
 	else if (rate % 44100 == 0)
 		code = 0x02;
-	else if (rate % 48000 == 0)
-		code = 0x04;
+	else if (rate % 32000 == 0)
+		code = 0x00;
 	else
 		return -EINVAL;
 
 	if (rate >= 64000 && rate < 128000)
 		code |= 0x08;
-	else if (rate >= 128000 && rate < 192000)
+	else if (rate >= 128000)
 		code |= 0x10;
 
 	reg = cpu_to_le32(code);
@@ -140,7 +146,7 @@ static int latter_allocate_resources(struct snd_ff *ff, unsigned int rate)
 		if (curr_rate == rate)
 			break;
 	}
-	if (count == 10)
+	if (count > 10)
 		return -ETIMEDOUT;
 
 	for (i = 0; i < ARRAY_SIZE(amdtp_rate_table); ++i) {
@@ -181,14 +187,30 @@ static int latter_begin_session(struct snd_ff *ff, unsigned int rate)
 	__le32 reg;
 	int err;
 
-	if (rate >= 32000 && rate <= 48000)
-		flag = 0x92;
-	else if (rate >= 64000 && rate <= 96000)
-		flag = 0x8e;
-	else if (rate >= 128000 && rate <= 192000)
-		flag = 0x8c;
-	else
-		return -EINVAL;
+	if (ff->unit_version == SND_FF_UNIT_VERSION_UCX) {
+		// For Fireface UCX. Always use the maximum number of data
+		// channels in data block of packet.
+		if (rate >= 32000 && rate <= 48000)
+			flag = 0x92;
+		else if (rate >= 64000 && rate <= 96000)
+			flag = 0x8e;
+		else if (rate >= 128000 && rate <= 192000)
+			flag = 0x8c;
+		else
+			return -EINVAL;
+	} else {
+		// For Fireface UFX and 802. Due to bandwidth limitation on
+		// IEEE 1394a (400 Mbps), Analog 1-12 and AES are available
+		// without any ADAT at quadruple speed.
+		if (rate >= 32000 && rate <= 48000)
+			flag = 0x9e;
+		else if (rate >= 64000 && rate <= 96000)
+			flag = 0x96;
+		else if (rate >= 128000 && rate <= 192000)
+			flag = 0x8e;
+		else
+			return -EINVAL;
+	}
 
 	if (generation != fw_parent_device(ff->unit)->card->generation) {
 		err = fw_iso_resources_update(&ff->tx_resources);
@@ -207,8 +229,6 @@ static int latter_begin_session(struct snd_ff *ff, unsigned int rate)
 	if (err < 0)
 		return err;
 
-	// Always use the maximum number of data channels in data block of
-	// packet.
 	reg = cpu_to_le32(flag);
 	return snd_fw_transaction(ff->unit, TCODE_WRITE_QUADLET_REQUEST,
 				  LATTER_ISOC_START, &reg, sizeof(reg), 0);
@@ -263,7 +283,7 @@ static void latter_dump_status(struct snd_ff *ff, struct snd_info_buffer *buffer
 		}
 	}
 
-	err = parse_clock_bits(data, &rate, &src);
+	err = parse_clock_bits(data, &rate, &src, ff->unit_version);
 	if (err < 0)
 		return;
 	label = snd_ff_proc_get_clk_label(src);

sound/firewire/fireface/ff-stream.c

@@ -184,7 +184,6 @@ int snd_ff_stream_start_duplex(struct snd_ff *ff, unsigned int rate)
 	 */
 	if (!amdtp_stream_running(&ff->rx_stream)) {
 		int spd = fw_parent_device(ff->unit)->max_speed;
-		unsigned int ir_delay_cycle;
 
 		err = ff->spec->protocol->begin_session(ff, rate);
 		if (err < 0)
@@ -200,14 +199,7 @@ int snd_ff_stream_start_duplex(struct snd_ff *ff, unsigned int rate)
 		if (err < 0)
 			goto error;
 
-		// The device postpones start of transmission mostly for several
-		// cycles after receiving packets firstly.
-		if (ff->spec->protocol == &snd_ff_protocol_ff800)
-			ir_delay_cycle = 800;	// = 100 msec
-		else
-			ir_delay_cycle = 16;	// = 2 msec
-
-		err = amdtp_domain_start(&ff->domain, ir_delay_cycle);
+		err = amdtp_domain_start(&ff->domain, 0);
 		if (err < 0)
 			goto error;
 

sound/firewire/fireface/ff.c

@@ -16,12 +16,22 @@ MODULE_LICENSE("GPL v2");
 static void name_card(struct snd_ff *ff)
 {
 	struct fw_device *fw_dev = fw_parent_device(ff->unit);
+	const char *const names[] = {
+		[SND_FF_UNIT_VERSION_FF800]	= "Fireface800",
+		[SND_FF_UNIT_VERSION_FF400]	= "Fireface400",
+		[SND_FF_UNIT_VERSION_UFX]	= "FirefaceUFX",
+		[SND_FF_UNIT_VERSION_UCX]	= "FirefaceUCX",
+		[SND_FF_UNIT_VERSION_802]	= "Fireface802",
+	};
+	const char *name;
+
+	name = names[ff->unit_version];
 
 	strcpy(ff->card->driver, "Fireface");
-	strcpy(ff->card->shortname, ff->spec->name);
-	strcpy(ff->card->mixername, ff->spec->name);
+	strcpy(ff->card->shortname, name);
+	strcpy(ff->card->mixername, name);
 	snprintf(ff->card->longname, sizeof(ff->card->longname),
-		 "RME %s, GUID %08x%08x at %s, S%d", ff->spec->name,
+		 "RME %s, GUID %08x%08x at %s, S%d", name,
 		 fw_dev->config_rom[3], fw_dev->config_rom[4],
 		 dev_name(&ff->unit->device), 100 << fw_dev->max_speed);
 }
@@ -101,6 +111,7 @@ static int snd_ff_probe(struct fw_unit *unit,
 	spin_lock_init(&ff->lock);
 	init_waitqueue_head(&ff->hwdep_wait);
 
+	ff->unit_version = entry->version;
 	ff->spec = (const struct snd_ff_spec *)entry->driver_data;
 
 	/* Register this sound card later. */
@@ -145,7 +156,6 @@ static void snd_ff_remove(struct fw_unit *unit)
 }
 
 static const struct snd_ff_spec spec_ff800 = {
-	.name = "Fireface800",
 	.pcm_capture_channels = {28, 20, 12},
 	.pcm_playback_channels = {28, 20, 12},
 	.midi_in_ports = 1,
@@ -157,7 +167,6 @@ static const struct snd_ff_spec spec_ff800 = {
 };
 
 static const struct snd_ff_spec spec_ff400 = {
-	.name = "Fireface400",
 	.pcm_capture_channels = {18, 14, 10},
 	.pcm_playback_channels = {18, 14, 10},
 	.midi_in_ports = 2,
@@ -169,7 +178,6 @@ static const struct snd_ff_spec spec_ff400 = {
 };
 
 static const struct snd_ff_spec spec_ucx = {
-	.name = "FirefaceUCX",
 	.pcm_capture_channels = {18, 14, 12},
 	.pcm_playback_channels = {18, 14, 12},
 	.midi_in_ports = 2,
@@ -180,6 +188,17 @@ static const struct snd_ff_spec spec_ucx = {
 	.midi_rx_addrs = {0xffff00000030ull, 0xffff00000030ull},
 };
 
+static const struct snd_ff_spec spec_ufx_802 = {
+	.pcm_capture_channels = {30, 22, 14},
+	.pcm_playback_channels = {30, 22, 14},
+	.midi_in_ports = 1,
+	.midi_out_ports = 1,
+	.protocol = &snd_ff_protocol_latter,
+	.midi_high_addr = 0xffff00000034ull,
+	.midi_addr_range = 0x80,
+	.midi_rx_addrs = {0xffff00000030ull, 0xffff00000030ull},
+};
+
 static const struct ieee1394_device_id snd_ff_id_table[] = {
 	/* Fireface 800 */
 	{
@@ -189,7 +208,7 @@ static const struct ieee1394_device_id snd_ff_id_table[] = {
 				  IEEE1394_MATCH_MODEL_ID,
 		.vendor_id	= OUI_RME,
 		.specifier_id	= OUI_RME,
-		.version	= 0x000001,
+		.version	= SND_FF_UNIT_VERSION_FF800,
 		.model_id	= 0x101800,
 		.driver_data	= (kernel_ulong_t)&spec_ff800,
 	},
@@ -201,10 +220,22 @@ static const struct ieee1394_device_id snd_ff_id_table[] = {
 				  IEEE1394_MATCH_MODEL_ID,
 		.vendor_id	= OUI_RME,
 		.specifier_id	= OUI_RME,
-		.version	= 0x000002,
+		.version	= SND_FF_UNIT_VERSION_FF400,
 		.model_id	= 0x101800,
 		.driver_data	= (kernel_ulong_t)&spec_ff400,
 	},
+	// Fireface UFX.
+	{
+		.match_flags	= IEEE1394_MATCH_VENDOR_ID |
+				  IEEE1394_MATCH_SPECIFIER_ID |
+				  IEEE1394_MATCH_VERSION |
+				  IEEE1394_MATCH_MODEL_ID,
+		.vendor_id	= OUI_RME,
+		.specifier_id	= OUI_RME,
+		.version	= SND_FF_UNIT_VERSION_UFX,
+		.model_id	= 0x101800,
+		.driver_data	= (kernel_ulong_t)&spec_ufx_802,
+	},
 	// Fireface UCX.
 	{
 		.match_flags	= IEEE1394_MATCH_VENDOR_ID |
@@ -213,10 +244,22 @@ static const struct ieee1394_device_id snd_ff_id_table[] = {
 				  IEEE1394_MATCH_MODEL_ID,
 		.vendor_id	= OUI_RME,
 		.specifier_id	= OUI_RME,
-		.version	= 0x000004,
+		.version	= SND_FF_UNIT_VERSION_UCX,
 		.model_id	= 0x101800,
 		.driver_data	= (kernel_ulong_t)&spec_ucx,
 	},
+	// Fireface 802.
+	{
+		.match_flags	= IEEE1394_MATCH_VENDOR_ID |
+				  IEEE1394_MATCH_SPECIFIER_ID |
+				  IEEE1394_MATCH_VERSION |
+				  IEEE1394_MATCH_MODEL_ID,
+		.vendor_id	= OUI_RME,
+		.specifier_id	= OUI_RME,
+		.version	= SND_FF_UNIT_VERSION_802,
+		.model_id	= 0x101800,
+		.driver_data	= (kernel_ulong_t)&spec_ufx_802,
+	},
 	{}
 };
 MODULE_DEVICE_TABLE(ieee1394, snd_ff_id_table);

sound/firewire/fireface/ff.h

@@ -34,6 +34,14 @@
 #define SND_FF_IN_MIDI_PORTS		2
 #define SND_FF_OUT_MIDI_PORTS		2
 
+enum snd_ff_unit_version {
+	SND_FF_UNIT_VERSION_FF800	= 0x000001,
+	SND_FF_UNIT_VERSION_FF400	= 0x000002,
+	SND_FF_UNIT_VERSION_UFX		= 0x000003,
+	SND_FF_UNIT_VERSION_UCX		= 0x000004,
+	SND_FF_UNIT_VERSION_802		= 0x000005,
+};
+
 enum snd_ff_stream_mode {
 	SND_FF_STREAM_MODE_LOW = 0,
 	SND_FF_STREAM_MODE_MID,
@@ -43,8 +51,6 @@ enum snd_ff_stream_mode {
 
 struct snd_ff_protocol;
 struct snd_ff_spec {
-	const char *const name;
-
 	const unsigned int pcm_capture_channels[SND_FF_STREAM_MODE_COUNT];
 	const unsigned int pcm_playback_channels[SND_FF_STREAM_MODE_COUNT];
 
@@ -66,6 +72,7 @@ struct snd_ff {
 	bool registered;
 	struct delayed_work dwork;
 
+	enum snd_ff_unit_version unit_version;
 	const struct snd_ff_spec *spec;
 
 	/* To handle MIDI tx. */

sound/firewire/fireworks/fireworks.h

@@ -177,7 +177,7 @@ struct snd_efw_phys_meters {
 	u32 in_meters;
 	u32 reserved4;
 	u32 reserved5;
-	u32 values[0];
+	u32 values[];
 } __packed;
 enum snd_efw_clock_source {
 	SND_EFW_CLOCK_SOURCE_INTERNAL	= 0,

sound/firewire/motu/amdtp-motu.c

@@ -76,15 +76,11 @@ int amdtp_motu_set_parameters(struct amdtp_stream *s, unsigned int rate,
 	if (i == ARRAY_SIZE(snd_motu_clock_rates))
 		return -EINVAL;
 
-	pcm_chunks = formats->fixed_part_pcm_chunks[mode] +
-		     formats->differed_part_pcm_chunks[mode];
+	// Each data block includes SPH in its head. Data chunks follow with
+	// 3 byte alignment. Padding follows with zero to conform to quadlet
+	// alignment.
+	pcm_chunks = formats->pcm_chunks[mode];
 	data_chunks = formats->msg_chunks + pcm_chunks;
-
-	/*
-	 * Each data block includes SPH in its head. Data chunks follow with
-	 * 3 byte alignment. Padding follows with zero to conform to quadlet
-	 * alignment.
-	 */
 	data_block_quadlets = 1 + DIV_ROUND_UP(data_chunks * 3, 4);
 
 	err = amdtp_stream_set_parameters(s, rate, data_block_quadlets);
@@ -440,7 +436,7 @@ static unsigned int process_it_ctx_payloads(struct amdtp_stream *s,
 
 int amdtp_motu_init(struct amdtp_stream *s, struct fw_unit *unit,
 		    enum amdtp_stream_direction dir,
-		    const struct snd_motu_protocol *const protocol)
+		    const struct snd_motu_spec *spec)
 {
 	amdtp_stream_process_ctx_payloads_t process_ctx_payloads;
 	int fmt = CIP_FMT_MOTU;
@@ -454,14 +450,15 @@ int amdtp_motu_init(struct amdtp_stream *s, struct fw_unit *unit,
 		 * Units of version 3 transmits packets with invalid CIP header
 		 * against IEC 61883-1.
 		 */
-		if (protocol == &snd_motu_protocol_v3) {
+		if (spec->protocol_version == SND_MOTU_PROTOCOL_V3) {
 			flags |= CIP_WRONG_DBS |
 				 CIP_SKIP_DBC_ZERO_CHECK |
 				 CIP_HEADER_WITHOUT_EOH;
 			fmt = CIP_FMT_MOTU_TX_V3;
 		}
 
-		if (protocol == &snd_motu_protocol_v2) {
+		if (spec == &snd_motu_spec_8pre ||
+		    spec == &snd_motu_spec_ultralite) {
 			// 8pre has some quirks.
 			flags |= CIP_WRONG_DBS |
 				 CIP_SKIP_DBC_ZERO_CHECK;

sound/firewire/motu/motu-pcm.c

@@ -26,8 +26,7 @@ static int motu_rate_constraint(struct snd_pcm_hw_params *params,
 		rate = snd_motu_clock_rates[i];
 		mode = i / 2;
 
-		pcm_channels = formats->fixed_part_pcm_chunks[mode] +
-			       formats->differed_part_pcm_chunks[mode];
+		pcm_channels = formats->pcm_chunks[mode];
 		if (!snd_interval_test(c, pcm_channels))
 			continue;
 
@@ -59,8 +58,7 @@ static int motu_channels_constraint(struct snd_pcm_hw_params *params,
 		if (!snd_interval_test(r, rate))
 			continue;
 
-		pcm_channels = formats->fixed_part_pcm_chunks[mode] +
-			       formats->differed_part_pcm_chunks[mode];
+		pcm_channels = formats->pcm_chunks[mode];
 		channels.min = min(channels.min, pcm_channels);
 		channels.max = max(channels.max, pcm_channels);
 	}
@@ -82,8 +80,7 @@ static void limit_channels_and_rates(struct snd_motu *motu,
 		rate = snd_motu_clock_rates[i];
 		mode = i / 2;
 
-		pcm_channels = formats->fixed_part_pcm_chunks[mode] +
-			       formats->differed_part_pcm_chunks[mode];
+		pcm_channels = formats->pcm_chunks[mode];
 		if (pcm_channels == 0)
 			continue;
 
@@ -133,7 +130,6 @@ static int init_hw_info(struct snd_motu *motu,
 static int pcm_open(struct snd_pcm_substream *substream)
 {
 	struct snd_motu *motu = substream->private_data;
-	const struct snd_motu_protocol *const protocol = motu->spec->protocol;
 	struct amdtp_domain *d = &motu->domain;
 	enum snd_motu_clock_source src;
 	int err;
@@ -152,7 +148,7 @@ static int pcm_open(struct snd_pcm_substream *substream)
 	if (err < 0)
 		goto err_locked;
 
-	err = protocol->get_clock_source(motu, &src);
+	err = snd_motu_protocol_get_clock_source(motu, &src);
 	if (err < 0)
 		goto err_locked;
 
@@ -166,7 +162,7 @@ static int pcm_open(struct snd_pcm_substream *substream)
 		unsigned int frames_per_buffer = d->events_per_buffer;
 		unsigned int rate;
 
-		err = protocol->get_clock_rate(motu, &rate);
+		err = snd_motu_protocol_get_clock_rate(motu, &rate);
 		if (err < 0)
 			goto err_locked;
 

sound/firewire/motu/motu-proc.c

@@ -28,13 +28,12 @@ static void proc_read_clock(struct snd_info_entry *entry,
 {
 
 	struct snd_motu *motu = entry->private_data;
-	const struct snd_motu_protocol *const protocol = motu->spec->protocol;
 	unsigned int rate;
 	enum snd_motu_clock_source source;
 
-	if (protocol->get_clock_rate(motu, &rate) < 0)
+	if (snd_motu_protocol_get_clock_rate(motu, &rate) < 0)
 		return;
-	if (protocol->get_clock_source(motu, &source) < 0)
+	if (snd_motu_protocol_get_clock_source(motu, &source) < 0)
 		return;
 
 	snd_iprintf(buffer, "Rate:\t%d\n", rate);
@@ -45,15 +44,14 @@ static void proc_read_format(struct snd_info_entry *entry,
 			     struct snd_info_buffer *buffer)
 {
 	struct snd_motu *motu = entry->private_data;
-	const struct snd_motu_protocol *const protocol = motu->spec->protocol;
 	unsigned int mode;
 	struct snd_motu_packet_format *formats;
 	int i;
 
-	if (protocol->cache_packet_formats(motu) < 0)
+	if (snd_motu_protocol_cache_packet_formats(motu) < 0)
 		return;
 
-	snd_iprintf(buffer, "tx:\tmsg\tfixed\tdiffered\n");
+	snd_iprintf(buffer, "tx:\tmsg\tfixed\ttotal\n");
 	for (i = 0; i < SND_MOTU_CLOCK_RATE_COUNT; ++i) {
 		mode = i >> 1;
 
@@ -62,11 +60,11 @@ static void proc_read_format(struct snd_info_entry *entry,
 			    "%u:\t%u\t%u\t%u\n",
 			    snd_motu_clock_rates[i],
 			    formats->msg_chunks,
-			    formats->fixed_part_pcm_chunks[mode],
-			    formats->differed_part_pcm_chunks[mode]);
+			    motu->spec->tx_fixed_pcm_chunks[mode],
+			    formats->pcm_chunks[mode]);
 	}
 
-	snd_iprintf(buffer, "rx:\tmsg\tfixed\tdiffered\n");
+	snd_iprintf(buffer, "rx:\tmsg\tfixed\ttotal\n");
 	for (i = 0; i < SND_MOTU_CLOCK_RATE_COUNT; ++i) {
 		mode = i >> 1;
 
@@ -75,8 +73,8 @@ static void proc_read_format(struct snd_info_entry *entry,
 			    "%u:\t%u\t%u\t%u\n",
 			    snd_motu_clock_rates[i],
 			    formats->msg_chunks,
-			    formats->fixed_part_pcm_chunks[mode],
-			    formats->differed_part_pcm_chunks[mode]);
+			    motu->spec->rx_fixed_pcm_chunks[mode],
+			    formats->pcm_chunks[mode]);
 	}
 }
 

sound/firewire/motu/motu-stream.c

@@ -88,7 +88,7 @@ static void finish_session(struct snd_motu *motu)
 	u32 data;
 	int err;
 
-	err = motu->spec->protocol->switch_fetching_mode(motu, false);
+	err = snd_motu_protocol_switch_fetching_mode(motu, false);
 	if (err < 0)
 		return;
 
@@ -110,7 +110,7 @@ int snd_motu_stream_cache_packet_formats(struct snd_motu *motu)
 {
 	int err;
 
-	err = motu->spec->protocol->cache_packet_formats(motu);
+	err = snd_motu_protocol_cache_packet_formats(motu);
 	if (err < 0)
 		return err;
 
@@ -140,7 +140,7 @@ int snd_motu_stream_reserve_duplex(struct snd_motu *motu, unsigned int rate,
 	unsigned int curr_rate;
 	int err;
 
-	err = motu->spec->protocol->get_clock_rate(motu, &curr_rate);
+	err = snd_motu_protocol_get_clock_rate(motu, &curr_rate);
 	if (err < 0)
 		return err;
 	if (rate == 0)
@@ -153,7 +153,7 @@ int snd_motu_stream_reserve_duplex(struct snd_motu *motu, unsigned int rate,
 		fw_iso_resources_free(&motu->tx_resources);
 		fw_iso_resources_free(&motu->rx_resources);
 
-		err = motu->spec->protocol->set_clock_rate(motu, rate);
+		err = snd_motu_protocol_set_clock_rate(motu, rate);
 		if (err < 0) {
 			dev_err(&motu->unit->device,
 				"fail to set sampling rate: %d\n", err);
@@ -201,9 +201,9 @@ static int ensure_packet_formats(struct snd_motu *motu)
 	data &= ~(TX_PACKET_EXCLUDE_DIFFERED_DATA_CHUNKS |
 		  RX_PACKET_EXCLUDE_DIFFERED_DATA_CHUNKS|
 		  TX_PACKET_TRANSMISSION_SPEED_MASK);
-	if (motu->tx_packet_formats.differed_part_pcm_chunks[0] == 0)
+	if (motu->spec->tx_fixed_pcm_chunks[0] == motu->tx_packet_formats.pcm_chunks[0])
 		data |= TX_PACKET_EXCLUDE_DIFFERED_DATA_CHUNKS;
-	if (motu->rx_packet_formats.differed_part_pcm_chunks[0] == 0)
+	if (motu->spec->rx_fixed_pcm_chunks[0] == motu->rx_packet_formats.pcm_chunks[0])
 		data |= RX_PACKET_EXCLUDE_DIFFERED_DATA_CHUNKS;
 	data |= fw_parent_device(motu->unit)->max_speed;
 
@@ -272,7 +272,7 @@ int snd_motu_stream_start_duplex(struct snd_motu *motu)
 			goto stop_streams;
 		}
 
-		err = motu->spec->protocol->switch_fetching_mode(motu, true);
+		err = snd_motu_protocol_switch_fetching_mode(motu, true);
 		if (err < 0) {
 			dev_err(&motu->unit->device,
 				"fail to enable frame fetching: %d\n", err);
@@ -317,7 +317,7 @@ static int init_stream(struct snd_motu *motu, struct amdtp_stream *s)
 	if (err < 0)
 		return err;
 
-	err = amdtp_motu_init(s, motu->unit, dir, motu->spec->protocol);
+	err = amdtp_motu_init(s, motu->unit, dir, motu->spec);
 	if (err < 0)
 		fw_iso_resources_destroy(resources);
 

sound/firewire/motu/motu.c

@@ -172,105 +172,6 @@ static void motu_bus_update(struct fw_unit *unit)
 	snd_motu_transaction_reregister(motu);
 }
 
-const struct snd_motu_spec snd_motu_spec_828mk2 = {
-	.name = "828mk2",
-	.protocol = &snd_motu_protocol_v2,
-	.flags = SND_MOTU_SPEC_SUPPORT_CLOCK_X2 |
-		 SND_MOTU_SPEC_TX_MICINST_CHUNK |
-		 SND_MOTU_SPEC_TX_RETURN_CHUNK |
-		 SND_MOTU_SPEC_RX_SEPARATED_MAIN |
-		 SND_MOTU_SPEC_HAS_OPT_IFACE_A |
-		 SND_MOTU_SPEC_RX_MIDI_2ND_Q |
-		 SND_MOTU_SPEC_TX_MIDI_2ND_Q,
-
-	.analog_in_ports = 8,
-	.analog_out_ports = 8,
-};
-
-static const struct snd_motu_spec motu_traveler = {
-	.name = "Traveler",
-	.protocol = &snd_motu_protocol_v2,
-	.flags = SND_MOTU_SPEC_SUPPORT_CLOCK_X2 |
-		 SND_MOTU_SPEC_SUPPORT_CLOCK_X4 |
-		 SND_MOTU_SPEC_TX_RETURN_CHUNK |
-		 SND_MOTU_SPEC_HAS_AESEBU_IFACE |
-		 SND_MOTU_SPEC_HAS_OPT_IFACE_A |
-		 SND_MOTU_SPEC_RX_MIDI_2ND_Q |
-		 SND_MOTU_SPEC_TX_MIDI_2ND_Q,
-
-	.analog_in_ports = 8,
-	.analog_out_ports = 8,
-};
-
-static const struct snd_motu_spec motu_ultralite = {
-	.name = "UltraLite",
-	.protocol = &snd_motu_protocol_v2,
-	.flags = SND_MOTU_SPEC_SUPPORT_CLOCK_X2 |
-		 SND_MOTU_SPEC_TX_MICINST_CHUNK | // padding.
-		 SND_MOTU_SPEC_TX_RETURN_CHUNK |
-		 SND_MOTU_SPEC_RX_MIDI_2ND_Q |
-		 SND_MOTU_SPEC_TX_MIDI_2ND_Q |
-		 SND_MOTU_SPEC_RX_SEPARATED_MAIN,
-	.analog_in_ports = 8,
-	.analog_out_ports = 8,
-};
-
-static const struct snd_motu_spec motu_8pre = {
-	.name = "8pre",
-	.protocol = &snd_motu_protocol_v2,
-	// In tx, use coax chunks for mix-return 1/2. In rx, use coax chunks for
-	// dummy 1/2.
-	.flags = SND_MOTU_SPEC_SUPPORT_CLOCK_X2 |
-		 SND_MOTU_SPEC_HAS_OPT_IFACE_A |
-		 SND_MOTU_SPEC_HAS_OPT_IFACE_B |
-		 SND_MOTU_SPEC_RX_MIDI_2ND_Q |
-		 SND_MOTU_SPEC_TX_MIDI_2ND_Q,
-	.analog_in_ports = 8,
-	.analog_out_ports = 2,
-};
-
-static const struct snd_motu_spec motu_828mk3 = {
-	.name = "828mk3",
-	.protocol = &snd_motu_protocol_v3,
-	.flags = SND_MOTU_SPEC_SUPPORT_CLOCK_X2 |
-		 SND_MOTU_SPEC_SUPPORT_CLOCK_X4 |
-		 SND_MOTU_SPEC_TX_MICINST_CHUNK |
-		 SND_MOTU_SPEC_TX_RETURN_CHUNK |
-		 SND_MOTU_SPEC_TX_REVERB_CHUNK |
-		 SND_MOTU_SPEC_RX_SEPARATED_MAIN |
-		 SND_MOTU_SPEC_HAS_OPT_IFACE_A |
-		 SND_MOTU_SPEC_HAS_OPT_IFACE_B |
-		 SND_MOTU_SPEC_RX_MIDI_3RD_Q |
-		 SND_MOTU_SPEC_TX_MIDI_3RD_Q,
-
-	.analog_in_ports = 8,
-	.analog_out_ports = 8,
-};
-
-static const struct snd_motu_spec motu_audio_express = {
-	.name = "AudioExpress",
-	.protocol = &snd_motu_protocol_v3,
-	.flags = SND_MOTU_SPEC_SUPPORT_CLOCK_X2 |
-		 SND_MOTU_SPEC_TX_MICINST_CHUNK |
-		 SND_MOTU_SPEC_TX_RETURN_CHUNK |
-		 SND_MOTU_SPEC_RX_SEPARATED_MAIN |
-		 SND_MOTU_SPEC_RX_MIDI_2ND_Q |
-		 SND_MOTU_SPEC_TX_MIDI_3RD_Q,
-	.analog_in_ports = 2,
-	.analog_out_ports = 4,
-};
-
-static const struct snd_motu_spec motu_4pre = {
-	.name = "4pre",
-	.protocol = &snd_motu_protocol_v3,
-	.flags = SND_MOTU_SPEC_SUPPORT_CLOCK_X2 |
-		 SND_MOTU_SPEC_TX_MICINST_CHUNK |
-		 SND_MOTU_SPEC_TX_RETURN_CHUNK |
-		 SND_MOTU_SPEC_RX_SEPARATED_MAIN,
-	.analog_in_ports = 2,
-	.analog_out_ports = 2,
-};
-
 #define SND_MOTU_DEV_ENTRY(model, data)			\
 {							\
 	.match_flags	= IEEE1394_MATCH_VENDOR_ID |	\
@@ -284,13 +185,14 @@ static const struct snd_motu_spec motu_4pre = {
 
 static const struct ieee1394_device_id motu_id_table[] = {
 	SND_MOTU_DEV_ENTRY(0x000003, &snd_motu_spec_828mk2),
-	SND_MOTU_DEV_ENTRY(0x000009, &motu_traveler),
-	SND_MOTU_DEV_ENTRY(0x00000d, &motu_ultralite),
-	SND_MOTU_DEV_ENTRY(0x00000f, &motu_8pre),
-	SND_MOTU_DEV_ENTRY(0x000015, &motu_828mk3),	/* FireWire only. */
-	SND_MOTU_DEV_ENTRY(0x000035, &motu_828mk3),	/* Hybrid. */
-	SND_MOTU_DEV_ENTRY(0x000033, &motu_audio_express),
-	SND_MOTU_DEV_ENTRY(0x000045, &motu_4pre),
+	SND_MOTU_DEV_ENTRY(0x000009, &snd_motu_spec_traveler),
+	SND_MOTU_DEV_ENTRY(0x00000d, &snd_motu_spec_ultralite),
+	SND_MOTU_DEV_ENTRY(0x00000f, &snd_motu_spec_8pre),
+	SND_MOTU_DEV_ENTRY(0x000015, &snd_motu_spec_828mk3), // FireWire only.
+	SND_MOTU_DEV_ENTRY(0x000019, &snd_motu_spec_ultralite_mk3), // FireWire only.
+	SND_MOTU_DEV_ENTRY(0x000035, &snd_motu_spec_828mk3), // Hybrid.
+	SND_MOTU_DEV_ENTRY(0x000033, &snd_motu_spec_audio_express),
+	SND_MOTU_DEV_ENTRY(0x000045, &snd_motu_spec_4pre),
 	{ }
 };
 MODULE_DEVICE_TABLE(ieee1394, motu_id_table);

sound/firewire/motu/motu.h

@@ -36,8 +36,7 @@ struct snd_motu_packet_format {
 	unsigned char pcm_byte_offset;
 
 	unsigned char msg_chunks;
-	unsigned char fixed_part_pcm_chunks[3];
-	unsigned char differed_part_pcm_chunks[3];
+	unsigned char pcm_chunks[3];
 };
 
 struct snd_motu {
@@ -74,19 +73,10 @@ struct snd_motu {
 };
 
 enum snd_motu_spec_flags {
-	SND_MOTU_SPEC_SUPPORT_CLOCK_X2	= 0x0001,
-	SND_MOTU_SPEC_SUPPORT_CLOCK_X4	= 0x0002,
-	SND_MOTU_SPEC_TX_MICINST_CHUNK	= 0x0004,
-	SND_MOTU_SPEC_TX_RETURN_CHUNK	= 0x0008,
-	SND_MOTU_SPEC_TX_REVERB_CHUNK	= 0x0010,
-	SND_MOTU_SPEC_HAS_AESEBU_IFACE	= 0x0020,
-	SND_MOTU_SPEC_HAS_OPT_IFACE_A	= 0x0040,
-	SND_MOTU_SPEC_HAS_OPT_IFACE_B	= 0x0080,
-	SND_MOTU_SPEC_RX_MIDI_2ND_Q	= 0x0100,
-	SND_MOTU_SPEC_RX_MIDI_3RD_Q	= 0x0200,
-	SND_MOTU_SPEC_TX_MIDI_2ND_Q	= 0x0400,
-	SND_MOTU_SPEC_TX_MIDI_3RD_Q	= 0x0800,
-	SND_MOTU_SPEC_RX_SEPARATED_MAIN	= 0x1000,
+	SND_MOTU_SPEC_RX_MIDI_2ND_Q	= 0x0001,
+	SND_MOTU_SPEC_RX_MIDI_3RD_Q	= 0x0002,
+	SND_MOTU_SPEC_TX_MIDI_2ND_Q	= 0x0004,
+	SND_MOTU_SPEC_TX_MIDI_3RD_Q	= 0x0008,
 };
 
 #define SND_MOTU_CLOCK_RATE_COUNT	6
@@ -108,33 +98,33 @@ enum snd_motu_clock_source {
 	SND_MOTU_CLOCK_SOURCE_UNKNOWN,
 };
 
-struct snd_motu_protocol {
-	int (*get_clock_rate)(struct snd_motu *motu, unsigned int *rate);
-	int (*set_clock_rate)(struct snd_motu *motu, unsigned int rate);
-	int (*get_clock_source)(struct snd_motu *motu,
-				enum snd_motu_clock_source *source);
-	int (*switch_fetching_mode)(struct snd_motu *motu, bool enable);
-	int (*cache_packet_formats)(struct snd_motu *motu);
+enum snd_motu_protocol_version {
+	SND_MOTU_PROTOCOL_V2,
+	SND_MOTU_PROTOCOL_V3,
 };
 
 struct snd_motu_spec {
 	const char *const name;
+	enum snd_motu_protocol_version protocol_version;
 	enum snd_motu_spec_flags flags;
 
-	unsigned char analog_in_ports;
-	unsigned char analog_out_ports;
-
-	const struct snd_motu_protocol *const protocol;
+	unsigned char tx_fixed_pcm_chunks[3];
+	unsigned char rx_fixed_pcm_chunks[3];
 };
 
-extern const struct snd_motu_protocol snd_motu_protocol_v2;
-extern const struct snd_motu_protocol snd_motu_protocol_v3;
-
 extern const struct snd_motu_spec snd_motu_spec_828mk2;
+extern const struct snd_motu_spec snd_motu_spec_traveler;
+extern const struct snd_motu_spec snd_motu_spec_ultralite;
+extern const struct snd_motu_spec snd_motu_spec_8pre;
+
+extern const struct snd_motu_spec snd_motu_spec_828mk3;
+extern const struct snd_motu_spec snd_motu_spec_ultralite_mk3;
+extern const struct snd_motu_spec snd_motu_spec_audio_express;
+extern const struct snd_motu_spec snd_motu_spec_4pre;
 
 int amdtp_motu_init(struct amdtp_stream *s, struct fw_unit *unit,
 		    enum amdtp_stream_direction dir,
-		    const struct snd_motu_protocol *const protocol);
+		    const struct snd_motu_spec *spec);
 int amdtp_motu_set_parameters(struct amdtp_stream *s, unsigned int rate,
 			      unsigned int midi_ports,
 			      struct snd_motu_packet_format *formats);
@@ -169,4 +159,79 @@ int snd_motu_create_pcm_devices(struct snd_motu *motu);
 int snd_motu_create_midi_devices(struct snd_motu *motu);
 
 int snd_motu_create_hwdep_device(struct snd_motu *motu);
+
+int snd_motu_protocol_v2_get_clock_rate(struct snd_motu *motu,
+					unsigned int *rate);
+int snd_motu_protocol_v2_set_clock_rate(struct snd_motu *motu,
+					unsigned int rate);
+int snd_motu_protocol_v2_get_clock_source(struct snd_motu *motu,
+					  enum snd_motu_clock_source *src);
+int snd_motu_protocol_v2_switch_fetching_mode(struct snd_motu *motu,
+					      bool enable);
+int snd_motu_protocol_v2_cache_packet_formats(struct snd_motu *motu);
+
+int snd_motu_protocol_v3_get_clock_rate(struct snd_motu *motu,
+					unsigned int *rate);
+int snd_motu_protocol_v3_set_clock_rate(struct snd_motu *motu,
+					unsigned int rate);
+int snd_motu_protocol_v3_get_clock_source(struct snd_motu *motu,
+					  enum snd_motu_clock_source *src);
+int snd_motu_protocol_v3_switch_fetching_mode(struct snd_motu *motu,
+					      bool enable);
+int snd_motu_protocol_v3_cache_packet_formats(struct snd_motu *motu);
+
+static inline int snd_motu_protocol_get_clock_rate(struct snd_motu *motu,
+						   unsigned int *rate)
+{
+	if (motu->spec->protocol_version == SND_MOTU_PROTOCOL_V2)
+		return snd_motu_protocol_v2_get_clock_rate(motu, rate);
+	else if (motu->spec->protocol_version == SND_MOTU_PROTOCOL_V3)
+		return snd_motu_protocol_v3_get_clock_rate(motu, rate);
+	else
+		return -ENXIO;
+}
+
+static inline int snd_motu_protocol_set_clock_rate(struct snd_motu *motu,
+						   unsigned int rate)
+{
+	if (motu->spec->protocol_version == SND_MOTU_PROTOCOL_V2)
+		return snd_motu_protocol_v2_set_clock_rate(motu, rate);
+	else if (motu->spec->protocol_version == SND_MOTU_PROTOCOL_V3)
+		return snd_motu_protocol_v3_set_clock_rate(motu, rate);
+	else
+		return -ENXIO;
+}
+
+static inline int snd_motu_protocol_get_clock_source(struct snd_motu *motu,
+					enum snd_motu_clock_source *source)
+{
+	if (motu->spec->protocol_version == SND_MOTU_PROTOCOL_V2)
+		return snd_motu_protocol_v2_get_clock_source(motu, source);
+	else if (motu->spec->protocol_version == SND_MOTU_PROTOCOL_V3)
+		return snd_motu_protocol_v3_get_clock_source(motu, source);
+	else
+		return -ENXIO;
+}
+
+static inline int snd_motu_protocol_switch_fetching_mode(struct snd_motu *motu,
+							 bool enable)
+{
+	if (motu->spec->protocol_version == SND_MOTU_PROTOCOL_V2)
+		return snd_motu_protocol_v2_switch_fetching_mode(motu, enable);
+	else if (motu->spec->protocol_version == SND_MOTU_PROTOCOL_V3)
+		return snd_motu_protocol_v3_switch_fetching_mode(motu, enable);
+	else
+		return -ENXIO;
+}
+
+static inline int snd_motu_protocol_cache_packet_formats(struct snd_motu *motu)
+{
+	if (motu->spec->protocol_version == SND_MOTU_PROTOCOL_V2)
+		return snd_motu_protocol_v2_cache_packet_formats(motu);
+	else if (motu->spec->protocol_version == SND_MOTU_PROTOCOL_V3)
+		return snd_motu_protocol_v3_cache_packet_formats(motu);
+	else
+		return -ENXIO;
+}
+
 #endif

sound/hda/ext/hdac_ext_bus.c

@@ -62,7 +62,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_exit);
 
 static void default_release(struct device *dev)
 {
-	snd_hdac_ext_bus_device_exit(container_of(dev, struct hdac_device, dev));
+	snd_hdac_ext_bus_device_exit(dev_to_hdac_dev(dev));
 }
 
 /**

sound/hda/hdac_bus.c

@@ -81,7 +81,6 @@ int snd_hdac_bus_exec_verb(struct hdac_bus *bus, unsigned int addr,
 	mutex_unlock(&bus->cmd_mutex);
 	return err;
 }
-EXPORT_SYMBOL_GPL(snd_hdac_bus_exec_verb);
 
 /**
  * snd_hdac_bus_exec_verb_unlocked - unlocked version
@@ -150,7 +149,6 @@ void snd_hdac_bus_queue_event(struct hdac_bus *bus, u32 res, u32 res_ex)
 
 	schedule_work(&bus->unsol_work);
 }
-EXPORT_SYMBOL_GPL(snd_hdac_bus_queue_event);
 
 /*
  * process queued unsolicited events
@@ -162,6 +160,7 @@ static void snd_hdac_bus_process_unsol_events(struct work_struct *work)
 	struct hdac_driver *drv;
 	unsigned int rp, caddr, res;
 
+	spin_lock_irq(&bus->reg_lock);
 	while (bus->unsol_rp != bus->unsol_wp) {
 		rp = (bus->unsol_rp + 1) % HDA_UNSOL_QUEUE_SIZE;
 		bus->unsol_rp = rp;
@@ -173,10 +172,13 @@ static void snd_hdac_bus_process_unsol_events(struct work_struct *work)
 		codec = bus->caddr_tbl[caddr & 0x0f];
 		if (!codec || !codec->dev.driver)
 			continue;
+		spin_unlock_irq(&bus->reg_lock);
 		drv = drv_to_hdac_driver(codec->dev.driver);
 		if (drv->unsol_event)
 			drv->unsol_event(codec, res);
+		spin_lock_irq(&bus->reg_lock);
 	}
+	spin_unlock_irq(&bus->reg_lock);
 }
 
 /**

sound/hda/hdac_controller.c

@@ -9,6 +9,7 @@
 #include <sound/core.h>
 #include <sound/hdaudio.h>
 #include <sound/hda_register.h>
+#include "local.h"
 
 /* clear CORB read pointer properly */
 static void azx_clear_corbrp(struct hdac_bus *bus)
@@ -527,6 +528,18 @@ bool snd_hdac_bus_init_chip(struct hdac_bus *bus, bool full_reset)
 	}
 
 	bus->chip_init = true;
+
+	/*
+	 * Default value of '8' is as per the HD audio specification (Rev 1.0a).
+	 * Following relation is used to derive STRIPE control value.
+	 *  For sample rate <= 48K:
+	 *   { ((num_channels * bits_per_sample) / number of SDOs) >= 8 }
+	 *  For sample rate > 48K:
+	 *   { ((num_channels * bits_per_sample * rate/48000) /
+	 *	number of SDOs) >= 8 }
+	 */
+	bus->sdo_limit = 8;
+
 	return true;
 }
 EXPORT_SYMBOL_GPL(snd_hdac_bus_init_chip);

sound/hda/hdac_device.c

@@ -20,7 +20,7 @@ static int get_codec_vendor_name(struct hdac_device *codec);
 
 static void default_release(struct device *dev)
 {
-	snd_hdac_device_exit(container_of(dev, struct hdac_device, dev));
+	snd_hdac_device_exit(dev_to_hdac_dev(dev));
 }
 
 /**

sound/hda/hdac_stream.c

@@ -38,7 +38,7 @@ int snd_hdac_get_stream_stripe_ctl(struct hdac_bus *bus,
 		else
 			value = (channels * bits_per_sample) / sdo_line;
 
-		if (value >= 8)
+		if (value >= bus->sdo_limit)
 			break;
 	}
 

sound/hda/intel-dsp-config.c

 // SPDX-License-Identifier: GPL-2.0
 // Copyright (c) 2019 Jaroslav Kysela <perex@perex.cz>
 
+#include <linux/acpi.h>
 #include <linux/bits.h>
 #include <linux/dmi.h>
 #include <linux/module.h>
 #include <linux/pci.h>
+#include <linux/soundwire/sdw.h>
+#include <linux/soundwire/sdw_intel.h>
 #include <sound/core.h>
 #include <sound/intel-dsp-config.h>
 #include <sound/intel-nhlt.h>
@@ -14,9 +17,14 @@ static int dsp_driver;
 module_param(dsp_driver, int, 0444);
 MODULE_PARM_DESC(dsp_driver, "Force the DSP driver for Intel DSP (0=auto, 1=legacy, 2=SST, 3=SOF)");
 
-#define FLAG_SST		BIT(0)
-#define FLAG_SOF		BIT(1)
-#define FLAG_SOF_ONLY_IF_DMIC	BIT(16)
+#define FLAG_SST			BIT(0)
+#define FLAG_SOF			BIT(1)
+#define FLAG_SST_ONLY_IF_DMIC		BIT(15)
+#define FLAG_SOF_ONLY_IF_DMIC		BIT(16)
+#define FLAG_SOF_ONLY_IF_SOUNDWIRE	BIT(17)
+
+#define FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE (FLAG_SOF_ONLY_IF_DMIC | \
+					    FLAG_SOF_ONLY_IF_SOUNDWIRE)
 
 struct config_entry {
 	u32 flags;
@@ -100,6 +108,10 @@ static const struct config_entry config_table[] = {
 			{}
 		}
 	},
+	{
+		.flags = FLAG_SST | FLAG_SST_ONLY_IF_DMIC,
+		.device = 0x9d70,
+	},
 #endif
 /* Kabylake-LP */
 #if IS_ENABLED(CONFIG_SND_SOC_INTEL_KBL)
@@ -116,6 +128,10 @@ static const struct config_entry config_table[] = {
 			{}
 		}
 	},
+	{
+		.flags = FLAG_SST | FLAG_SST_ONLY_IF_DMIC,
+		.device = 0x9d71,
+	},
 #endif
 
 /*
@@ -166,7 +182,7 @@ static const struct config_entry config_table[] = {
 		}
 	},
 	{
-		.flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC,
+		.flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE,
 		.device = 0x9dc8,
 	},
 #endif
@@ -187,7 +203,7 @@ static const struct config_entry config_table[] = {
 		}
 	},
 	{
-		.flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC,
+		.flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE,
 		.device = 0xa348,
 	},
 #endif
@@ -204,18 +220,50 @@ static const struct config_entry config_table[] = {
 					DMI_MATCH(DMI_SYS_VENDOR, "Google"),
 				}
 			},
+			{
+				.matches = {
+					DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"),
+					DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "09C6")
+				},
+			},
+			{
+				/* early version of SKU 09C6 */
+				.matches = {
+					DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"),
+					DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0983")
+				},
+			},
 			{}
 		}
 	},
 	{
-		.flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC,
+		.flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE,
 		.device = 0x02c8,
 	},
 #endif
 /* Cometlake-H */
 #if IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_H)
 	{
-		.flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC,
+		.flags = FLAG_SOF,
+		.device = 0x06c8,
+		.dmi_table = (const struct dmi_system_id []) {
+			{
+				.matches = {
+					DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"),
+					DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "098F"),
+				},
+			},
+			{
+				.matches = {
+					DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"),
+					DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0990"),
+				},
+			},
+			{}
+		}
+	},
+	{
+		.flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE,
 		.device = 0x06c8,
 	},
 #endif
@@ -236,7 +284,7 @@ static const struct config_entry config_table[] = {
 		}
 	},
 	{
-		.flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC,
+		.flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE,
 		.device = 0x34c8,
 	},
 #endif
@@ -256,9 +304,8 @@ static const struct config_entry config_table[] = {
 			{}
 		}
 	},
-
 	{
-		.flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC,
+		.flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE,
 		.device = 0xa0c8,
 	},
 #endif
@@ -303,6 +350,28 @@ static int snd_intel_dsp_check_dmic(struct pci_dev *pci)
 	return ret;
 }
 
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE)
+static int snd_intel_dsp_check_soundwire(struct pci_dev *pci)
+{
+	struct sdw_intel_acpi_info info;
+	acpi_handle handle;
+	int ret;
+
+	handle = ACPI_HANDLE(&pci->dev);
+
+	ret = sdw_intel_acpi_scan(handle, &info);
+	if (ret < 0)
+		return ret;
+
+	return info.link_mask;
+}
+#else
+static int snd_intel_dsp_check_soundwire(struct pci_dev *pci)
+{
+	return 0;
+}
+#endif
+
 int snd_intel_dsp_driver_probe(struct pci_dev *pci)
 {
 	const struct config_entry *cfg;
@@ -336,22 +405,36 @@ int snd_intel_dsp_driver_probe(struct pci_dev *pci)
 		return SND_INTEL_DSP_DRIVER_ANY;
 
 	if (cfg->flags & FLAG_SOF) {
-		if (cfg->flags & FLAG_SOF_ONLY_IF_DMIC) {
+		if (cfg->flags & FLAG_SOF_ONLY_IF_SOUNDWIRE &&
+		    snd_intel_dsp_check_soundwire(pci) > 0) {
+			dev_info(&pci->dev, "SoundWire enabled on CannonLake+ platform, using SOF driver\n");
+			return SND_INTEL_DSP_DRIVER_SOF;
+		}
+		if (cfg->flags & FLAG_SOF_ONLY_IF_DMIC &&
+		    snd_intel_dsp_check_dmic(pci)) {
+			dev_info(&pci->dev, "Digital mics found on Skylake+ platform, using SOF driver\n");
+			return SND_INTEL_DSP_DRIVER_SOF;
+		}
+		if (!(cfg->flags & FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE))
+			return SND_INTEL_DSP_DRIVER_SOF;
+	}
+
+
+	if (cfg->flags & FLAG_SST) {
+		if (cfg->flags & FLAG_SST_ONLY_IF_DMIC) {
 			if (snd_intel_dsp_check_dmic(pci)) {
-				dev_info(&pci->dev, "Digital mics found on Skylake+ platform, using SOF driver\n");
-				return SND_INTEL_DSP_DRIVER_SOF;
+				dev_info(&pci->dev, "Digital mics found on Skylake+ platform, using SST driver\n");
+				return SND_INTEL_DSP_DRIVER_SST;
 			}
 		} else {
-			return SND_INTEL_DSP_DRIVER_SOF;
+			return SND_INTEL_DSP_DRIVER_SST;
 		}
 	}
 
-	if (cfg->flags & FLAG_SST)
-		return SND_INTEL_DSP_DRIVER_SST;
-
 	return SND_INTEL_DSP_DRIVER_LEGACY;
 }
 EXPORT_SYMBOL_GPL(snd_intel_dsp_driver_probe);
 
 MODULE_LICENSE("GPL v2");
 MODULE_DESCRIPTION("Intel DSP config driver");
+MODULE_IMPORT_NS(SOUNDWIRE_INTEL_INIT);

sound/hda/local.h

@@ -36,6 +36,9 @@ void hda_widget_sysfs_exit(struct hdac_device *codec);
 int snd_hdac_bus_add_device(struct hdac_bus *bus, struct hdac_device *codec);
 void snd_hdac_bus_remove_device(struct hdac_bus *bus,
 				struct hdac_device *codec);
+void snd_hdac_bus_queue_event(struct hdac_bus *bus, u32 res, u32 res_ex);
+int snd_hdac_bus_exec_verb(struct hdac_bus *bus, unsigned int addr,
+			   unsigned int cmd, unsigned int *res);
 
 int snd_hdac_exec_verb(struct hdac_device *codec, unsigned int cmd,
 		       unsigned int flags, unsigned int *res);

sound/isa/ad1816a/ad1816a.c

@@ -54,7 +54,7 @@ MODULE_PARM_DESC(clockfreq, "Clock frequency for ad1816a driver (default = 0).")
 static const struct pnp_card_device_id snd_ad1816a_pnpids[] = {
 	/* Analog Devices AD1815 */
 	{ .id = "ADS7150", .devs = { { .id = "ADS7150" }, { .id = "ADS7151" } } },
-	/* Analog Device AD1816? */
+	/* Analog Devices AD1816? */
 	{ .id = "ADS7180", .devs = { { .id = "ADS7180" }, { .id = "ADS7181" } } },
 	/* Analog Devices AD1816A - added by Kenneth Platz <kxp@atl.hp.com> */
 	{ .id = "ADS7181", .devs = { { .id = "ADS7180" }, { .id = "ADS7181" } } },

sound/isa/es1688/es1688.c

@@ -267,8 +267,10 @@ static int snd_es968_pnp_detect(struct pnp_card_link *pcard,
 		return error;
 	}
 	error = snd_es1688_probe(card, dev);
-	if (error < 0)
+	if (error < 0) {
+		snd_card_free(card);
 		return error;
+	}
 	pnp_set_card_drvdata(pcard, card);
 	snd_es968_pnp_is_probed = 1;
 	return 0;

sound/pci/ac97/ac97_patch.c

@@ -1356,7 +1356,7 @@ static int patch_cx20551(struct snd_ac97 *ac97)
 }
 
 /*
- * Analog Device AD18xx, AD19xx codecs
+ * Analog Devices AD18xx, AD19xx codecs
  */
 #ifdef CONFIG_PM
 static void ad18xx_resume(struct snd_ac97 *ac97)