Merge tag 'sound-5.3-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "Many updates in this development cycle are found in ASoC where it got
  a wide range of changes for the continued refactoring.

  Some highlights are below.

  ASoC:

   - Continued refactoring work by Morimoto-san toward the full
     componentization; the changes are seen allover the places

   - Support for force disconnecting muxes in DAPM

   - Continued development of ASoC Intel SOF stuff

   - New drivers for Cirrus Logic CS47L35, CS47L85 and CS47L90, Conexant
     CX2072X, Realtek RT1011 and RT1308

  HD-audio:

   - More fixes and adjustments for ASoC SOF HD-audio

   - Fix for resume problem on some Realtek codecs

  USB-audio:

   - A few fixes for the issues reported by syzbot USB fuzzer

   - Fix for UAC2 extension unit parser

   - Quirks for Line6 Helix, Emgaic Unitor 8

  FireWire:

   - Lots of code refactoring and fixes in most of its components"

* tag 'sound-5.3-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (626 commits)
  ALSA: firewire-lib: code refactoring for local variables
  ALSA: firewire-lib: code refactoring for post operation to data block counter
  ALSA: firewire-lib: code refactoring for error path of parser for CIP header
  ALSA: firewire-lib: fix different data block counter between probed event and transferred isochronous packet
  ALSA: firewire-lib: fix initial value of data block count for IR context without CIP_DBC_IS_END_EVENT
  ALSA: firewire-lib/fireface: fix initial value of data block counter for IR context with CIP_NO_HEADER
  ALSA: firewire-lib: fix invalid length of rx packet payload for tracepoint events
  ALSA: usb-audio: fix Line6 Helix audio format rates
  firewire-motu: fix wrong reference count for stream functionality at error path of rawmidi interface
  ALSA: firewire-digi00x: fix wrong reference count for stream functionality at error path of rawmidi interface
  ALSA: dice: fix wrong reference count for stream functionality at error path of rawmidi interface
  ALSA: oxfw: fix wrong reference count for stream functionality at error path of rawmidi interface
  ALSA: fireworks: fix wrong reference count for stream functionality at error path of rawmidi interface
  ALSA: bebob: fix wrong reference count for stream functionality at error path of rawmidi interface
  ASoC: SOF: Intel: implement runtime idle for CNL/APL
  ASoC: SOF: add runtime idle callback
  ASoC: hdac_hdmi: report codec link up/down status to bus
  ASoC: SOF: debug: fix possible memory leak in sof_dfsentry_write()
  ASoC: sunxi: sun50i-codec-analog: Add earpiece
  ASoC: rt5665: remove redundant assignment to variable idx
  ...

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

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

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

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

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

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

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

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

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

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

Documentation/devicetree/bindings/sound/cs42xx8.txt

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

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

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

Documentation/devicetree/bindings/sound/madera.txt

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

Documentation/devicetree/bindings/sound/max98357a.txt

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

Documentation/devicetree/bindings/sound/rt1011.txt

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

Documentation/devicetree/bindings/sound/rt1308.txt

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

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

@@ -18,7 +18,7 @@ Required properties:
     See Documentation/devicetree/bindings/dma/stm32-dma.txt.
   - dma-names: Identifier for each DMA request line. Must be "tx" and "rx".
   - pinctrl-names: should contain only value "default"
-  - pinctrl-0: see Documentation/devicetree/bindings/pinctrl/st,stm32-pinctrl.txt
+  - pinctrl-0: see Documentation/devicetree/bindings/pinctrl/st,stm32-pinctrl.yaml
 
 Optional properties:
   - resets: Reference to a reset controller asserting the reset controller

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

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

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

-* Allwinner A10 I2S controller
-
-The I2S bus (Inter-IC sound bus) is a serial link for digital
-audio data transfer between devices in the system.
-
-Required properties:
-
-- compatible: should be one of the following:
-   - "allwinner,sun4i-a10-i2s"
-   - "allwinner,sun6i-a31-i2s"
-   - "allwinner,sun8i-a83t-i2s"
-   - "allwinner,sun8i-h3-i2s"
-   - "allwinner,sun50i-a64-codec-i2s"
-- reg: physical base address of the controller and length of memory mapped
-  region.
-- interrupts: should contain the I2S interrupt.
-- dmas: DMA specifiers for tx and rx dma. See the DMA client binding,
-	Documentation/devicetree/bindings/dma/dma.txt
-- dma-names: should include "tx" and "rx".
-- clocks: a list of phandle + clock-specifer pairs, one for each entry in clock-names.
-- clock-names: should contain the following:
-   - "apb" : clock for the I2S bus interface
-   - "mod" : module clock for the I2S controller
-- #sound-dai-cells : Must be equal to 0
-
-Required properties for the following compatibles:
-	- "allwinner,sun6i-a31-i2s"
-	- "allwinner,sun8i-a83t-i2s"
-	- "allwinner,sun8i-h3-i2s"
-	- "allwinner,sun50i-a64-codec-i2s"
-- resets: phandle to the reset line for this codec
-
-Example:
-
-i2s0: i2s@1c22400 {
-	#sound-dai-cells = <0>;
-	compatible = "allwinner,sun4i-a10-i2s";
-	reg = <0x01c22400 0x400>;
-	interrupts = <GIC_SPI 16 IRQ_TYPE_LEVEL_HIGH>;
-	clocks = <&apb0_gates 3>, <&i2s0_clk>;
-	clock-names = "apb", "mod";
-	dmas = <&dma SUN4I_DMA_NORMAL 3>,
-	       <&dma SUN4I_DMA_NORMAL 3>;
-	dma-names = "rx", "tx";
-};

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

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

MAINTAINERS

@@ -1297,7 +1297,7 @@ ARM PRIMECELL SSP PL022 SPI DRIVER
 M:	Linus Walleij <linus.walleij@linaro.org>
 L:	linux-arm-kernel@lists.infradead.org (moderated for non-subscribers)
 S:	Maintained
-F:	Documentation/devicetree/bindings/spi/spi_pl022.txt
+F:	Documentation/devicetree/bindings/spi/spi-pl022.yaml
 F:	drivers/spi/spi-pl022.c
 
 ARM PRIMECELL UART PL010 AND PL011 DRIVERS
@@ -3942,13 +3942,18 @@ W:	https://github.com/CirrusLogic/linux-drivers/wiki
 S:	Supported
 F:	Documentation/devicetree/bindings/mfd/madera.txt
 F:	Documentation/devicetree/bindings/pinctrl/cirrus,madera-pinctrl.txt
+F:	Documentation/devicetree/bindings/sound/madera.txt
+F:	include/dt-bindings/sound/madera*
 F:	include/linux/irqchip/irq-madera*
 F:	include/linux/mfd/madera/*
+F:	include/sound/madera*
 F:	drivers/gpio/gpio-madera*
 F:	drivers/irqchip/irq-madera*
 F:	drivers/mfd/madera*
 F:	drivers/mfd/cs47l*
 F:	drivers/pinctrl/cirrus/*
+F:	sound/soc/codecs/cs47l*
+F:	sound/soc/codecs/madera*
 
 CLANG-FORMAT FILE
 M:	Miguel Ojeda <miguel.ojeda.sandonis@gmail.com>

drivers/gpu/drm/vc4/vc4_hdmi.c

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

include/dt-bindings/sound/madera.h

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

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

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

include/linux/mfd/madera/pdata.h

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

include/sound/hda_codec.h

@@ -18,6 +18,9 @@
 #include <sound/hda_verbs.h>
 #include <sound/hda_regmap.h>
 
+#define IS_BXT(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x5a98)
+#define IS_CFL(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0xa348)
+
 /*
  * Structures
  */
@@ -268,9 +271,6 @@ struct hda_codec {
 	unsigned long jackpoll_interval; /* In jiffies. Zero means no poll, rely on unsol events */
 	struct delayed_work jackpoll_work;
 
-	/* jack detection */
-	struct snd_array jacks;
-
 	int depop_delay; /* depop delay in ms, -1 for default delay time */
 
 	/* fix-up list */

include/sound/hdaudio.h

@@ -120,7 +120,7 @@ void snd_hdac_device_unregister(struct hdac_device *codec);
 int snd_hdac_device_set_chip_name(struct hdac_device *codec, const char *name);
 int snd_hdac_codec_modalias(struct hdac_device *hdac, char *buf, size_t size);
 
-int snd_hdac_refresh_widgets(struct hdac_device *codec, bool sysfs);
+int snd_hdac_refresh_widgets(struct hdac_device *codec);
 
 unsigned int snd_hdac_make_cmd(struct hdac_device *codec, hda_nid_t nid,
 			       unsigned int verb, unsigned int parm);
@@ -358,6 +358,9 @@ struct hdac_bus {
 	bool align_bdle_4k:1;		/* BDLE align 4K boundary */
 	bool reverse_assign:1;		/* assign devices in reverse order */
 	bool corbrp_self_clear:1;	/* CORBRP clears itself after reset */
+	bool polling_mode:1;
+
+	int poll_count;
 
 	int bdl_pos_adj;		/* BDL position adjustment */
 

include/sound/madera-pdata.h

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

include/sound/simple_card_utils.h

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

include/sound/soc.h

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

include/sound/sof/dai-intel.h

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

include/sound/sof/header.h

@@ -49,6 +49,7 @@
 #define SOF_IPC_GLB_DAI_MSG			SOF_GLB_TYPE(0x8U)
 #define SOF_IPC_GLB_TRACE_MSG			SOF_GLB_TYPE(0x9U)
 #define SOF_IPC_GLB_GDB_DEBUG                   SOF_GLB_TYPE(0xAU)
+#define SOF_IPC_GLB_TEST_MSG			SOF_GLB_TYPE(0xBU)
 
 /*
  * DSP Command Message Types
@@ -99,9 +100,13 @@
 #define SOF_IPC_STREAM_VORBIS_PARAMS		SOF_CMD_TYPE(0x010)
 #define SOF_IPC_STREAM_VORBIS_FREE		SOF_CMD_TYPE(0x011)
 
-/* trace and debug */
+/* trace */
 #define SOF_IPC_TRACE_DMA_PARAMS		SOF_CMD_TYPE(0x001)
 #define SOF_IPC_TRACE_DMA_POSITION		SOF_CMD_TYPE(0x002)
+#define SOF_IPC_TRACE_DMA_PARAMS_EXT		SOF_CMD_TYPE(0x003)
+
+/* debug */
+#define SOF_IPC_TEST_IPC_FLOOD                  SOF_CMD_TYPE(0x001)
 
 /* Get message component id */
 #define SOF_IPC_MESSAGE_ID(x)			((x) & 0xffff)

include/sound/sof/topology.h

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

include/sound/sof/trace.h

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

include/uapi/linux/usb/audio.h

@@ -450,6 +450,43 @@ static inline __u8 *uac_processing_unit_specific(struct uac_processing_unit_desc
 	}
 }
 
+/*
+ * Extension Unit (XU) has almost compatible layout with Processing Unit, but
+ * on UAC2, it has a different bmControls size (bControlSize); it's 1 byte for
+ * XU while 2 bytes for PU.  The last iExtension field is a one-byte index as
+ * well as iProcessing field of PU.
+ */
+static inline __u8 uac_extension_unit_bControlSize(struct uac_processing_unit_descriptor *desc,
+						   int protocol)
+{
+	switch (protocol) {
+	case UAC_VERSION_1:
+		return desc->baSourceID[desc->bNrInPins + 4];
+	case UAC_VERSION_2:
+		return 1; /* in UAC2, this value is constant */
+	case UAC_VERSION_3:
+		return 4; /* in UAC3, this value is constant */
+	default:
+		return 1;
+	}
+}
+
+static inline __u8 uac_extension_unit_iExtension(struct uac_processing_unit_descriptor *desc,
+						 int protocol)
+{
+	__u8 control_size = uac_extension_unit_bControlSize(desc, protocol);
+
+	switch (protocol) {
+	case UAC_VERSION_1:
+	case UAC_VERSION_2:
+	default:
+		return *(uac_processing_unit_bmControls(desc, protocol)
+			 + control_size);
+	case UAC_VERSION_3:
+		return 0; /* UAC3 does not have this field */
+	}
+}
+
 /* 4.5.2 Class-Specific AS Interface Descriptor */
 struct uac1_as_header_descriptor {
 	__u8  bLength;			/* in bytes: 7 */

include/uapi/sound/sof/abi.h

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

include/uapi/sound/sof/eq.h

-/* SPDX-License-Identifier: ((GPL-2.0 WITH Linux-syscall-note) OR BSD-3-Clause) */
-/*
- * This file is provided under a dual BSD/GPLv2 license.  When using or
- * redistributing this file, you may do so under either license.
- *
- * Copyright(c) 2018 Intel Corporation. All rights reserved.
- */
-
-#ifndef __INCLUDE_UAPI_SOUND_SOF_USER_EQ_H__
-#define __INCLUDE_UAPI_SOUND_SOF_USER_EQ_H__
-
-/* FIR EQ type */
-
-#define SOF_EQ_FIR_IDX_SWITCH	0
-
-#define SOF_EQ_FIR_MAX_SIZE 4096 /* Max size allowed for coef data in bytes */
-
-#define SOF_EQ_FIR_MAX_LENGTH 192 /* Max length for individual filter */
-
-#define SOF_EQ_FIR_MAX_RESPONSES 8 /* A blob can define max 8 FIR EQs */
-
-/*
- * eq_fir_configuration data structure contains this information
- *     uint32_t size
- *	   This is the number of bytes need to store the received EQ
- *	   configuration.
- *     uint16_t channels_in_config
- *         This describes the number of channels in this EQ config data. It
- *         can be different from PLATFORM_MAX_CHANNELS.
- *     uint16_t number_of_responses
- *         0=no responses, 1=one response defined, 2=two responses defined, etc.
- *     int16_t data[]
- *         assign_response[channels_in_config]
- *             0 = use first response, 1 = use 2nd response, etc.
- *             E.g. {0, 0, 0, 0, 1, 1, 1, 1} would apply to channels 0-3 the
- *	       same first defined response and for to channels 4-7 the second.
- *         coef_data[]
- *             Repeated data
- *             { filter_length, output_shift, h[] }
- *	       for every EQ response defined where vector h has filter_length
- *             number of coefficients. Coefficients in h[] are in Q1.15 format.
- *             E.g. 16384 (Q1.15) = 0.5. The shifts are number of right shifts.
- *
- * NOTE: The channels_in_config must be even to have coef_data aligned to
- * 32 bit word in RAM. Therefore a mono EQ assign must be duplicated to 2ch
- * even if it would never used. Similarly a 5ch EQ assign must be increased
- * to 6ch. EQ init will return an error if this is not met.
- *
- * NOTE: The filter_length must be multiple of four. Therefore the filter must
- * be padded from the end with zeros have this condition met.
- */
-
-struct sof_eq_fir_config {
-	uint32_t size;
-	uint16_t channels_in_config;
-	uint16_t number_of_responses;
-
-	/* reserved */
-	uint32_t reserved[4];
-
-	int16_t data[];
-} __packed;
-
-struct sof_eq_fir_coef_data {
-	int16_t length; /* Number of FIR taps */
-	int16_t out_shift; /* Amount of right shifts at output */
-
-	/* reserved */
-	uint32_t reserved[4];
-
-	int16_t coef[]; /* FIR coefficients */
-} __packed;
-
-/* In the struct above there's two 16 bit words (length, shift) and four
- * reserved 32 bit words before the actual FIR coefficients. This information
- * is used in parsing of the configuration blob.
- */
-#define SOF_EQ_FIR_COEF_NHEADER \
-	(sizeof(struct sof_eq_fir_coef_data) / sizeof(int16_t))
-
-/* IIR EQ type */
-
-#define SOF_EQ_IIR_IDX_SWITCH   0
-
-#define SOF_EQ_IIR_MAX_SIZE 1024 /* Max size allowed for coef data in bytes */
-
-#define SOF_EQ_IIR_MAX_RESPONSES 8 /* A blob can define max 8 IIR EQs */
-
-/* eq_iir_configuration
- *     uint32_t channels_in_config
- *         This describes the number of channels in this EQ config data. It
- *         can be different from PLATFORM_MAX_CHANNELS.
- *     uint32_t number_of_responses_defined
- *         0=no responses, 1=one response defined, 2=two responses defined, etc.
- *     int32_t data[]
- *         Data consist of two parts. First is the response assign vector that
- *	   has length of channels_in_config. The latter part is coefficient
- *         data.
- *         uint32_t assign_response[channels_in_config]
- *             -1 = not defined, 0 = use first response, 1 = use 2nd, etc.
- *             E.g. {0, 0, 0, 0, -1, -1, -1, -1} would apply to channels 0-3 the
- *             same first defined response and leave channels 4-7 unequalized.
- *         coefficient_data[]
- *             <1st EQ>
- *             uint32_t num_biquads
- *             uint32_t num_biquads_in_series
- *             <1st biquad>
- *             int32_t coef_a2       Q2.30 format
- *             int32_t coef_a1       Q2.30 format
- *             int32_t coef_b2       Q2.30 format
- *             int32_t coef_b1       Q2.30 format
- *             int32_t coef_b0       Q2.30 format
- *             int32_t output_shift  number of shifts right, shift left is negative
- *             int32_t output_gain   Q2.14 format
- *             <2nd biquad>
- *             ...
- *             <2nd EQ>
- *
- *         Note: A flat response biquad can be made with a section set to
- *         b0 = 1.0, gain = 1.0, and other parameters set to 0
- *         {0, 0, 0, 0, 1073741824, 0, 16484}
- */
-
-struct sof_eq_iir_config {
-	uint32_t size;
-	uint32_t channels_in_config;
-	uint32_t number_of_responses;
-
-	/* reserved */
-	uint32_t reserved[4];
-
-	int32_t data[]; /* eq_assign[channels], eq 0, eq 1, ... */
-} __packed;
-
-struct sof_eq_iir_header_df2t {
-	uint32_t num_sections;
-	uint32_t num_sections_in_series;
-
-	/* reserved */
-	uint32_t reserved[4];
-
-	int32_t biquads[]; /* Repeated biquad coefficients */
-} __packed;
-
-struct sof_eq_iir_biquad_df2t {
-	int32_t a2; /* Q2.30 */
-	int32_t a1; /* Q2.30 */
-	int32_t b2; /* Q2.30 */
-	int32_t b1; /* Q2.30 */
-	int32_t b0; /* Q2.30 */
-	int32_t output_shift; /* Number of right shifts */
-	int32_t output_gain;  /* Q2.14 */
-} __packed;
-
-/* A full 22th order equalizer with 11 biquads cover octave bands 1-11 in
- * in the 0 - 20 kHz bandwidth.
- */
-#define SOF_EQ_IIR_DF2T_BIQUADS_MAX 11
-
-/* The number of int32_t words in sof_eq_iir_header_df2t:
- *	num_sections, num_sections_in_series, reserved[4]
- */
-#define SOF_EQ_IIR_NHEADER_DF2T \
-	(sizeof(struct sof_eq_iir_header_df2t) / sizeof(int32_t))
-
-/* The number of int32_t words in sof_eq_iir_biquad_df2t:
- *	a2, a1, b2, b1, b0, output_shift, output_gain
- */
-#define SOF_EQ_IIR_NBIQUAD_DF2T \
-	(sizeof(struct sof_eq_iir_biquad_df2t) / sizeof(int32_t))
-
-#endif

include/uapi/sound/sof/manifest.h

-/* SPDX-License-Identifier: ((GPL-2.0 WITH Linux-syscall-note) OR BSD-3-Clause) */
-/*
- * This file is provided under a dual BSD/GPLv2 license.  When using or
- * redistributing this file, you may do so under either license.
- *
- * Copyright(c) 2018 Intel Corporation. All rights reserved.
- */
-
-#ifndef __INCLUDE_UAPI_SOUND_SOF_USER_MANIFEST_H__
-#define __INCLUDE_UAPI_SOUND_SOF_USER_MANIFEST_H__
-
-/* start offset for base FW module */
-#define SOF_MAN_ELF_TEXT_OFFSET		0x2000
-
-/* FW Extended Manifest Header id = $AE1 */
-#define SOF_MAN_EXT_HEADER_MAGIC	0x31454124
-
-/* module type load type */
-#define SOF_MAN_MOD_TYPE_BUILTIN	0
-#define SOF_MAN_MOD_TYPE_MODULE		1
-
-struct sof_man_module_type {
-	uint32_t load_type:4;	/* SOF_MAN_MOD_TYPE_ */
-	uint32_t auto_start:1;
-	uint32_t domain_ll:1;
-	uint32_t domain_dp:1;
-	uint32_t rsvd_:25;
-};
-
-/* segment flags.type */
-#define SOF_MAN_SEGMENT_TEXT		0
-#define SOF_MAN_SEGMENT_RODATA		1
-#define SOF_MAN_SEGMENT_DATA		1
-#define SOF_MAN_SEGMENT_BSS		2
-#define SOF_MAN_SEGMENT_EMPTY		15
-
-union sof_man_segment_flags {
-	uint32_t ul;
-	struct {
-		uint32_t contents:1;
-		uint32_t alloc:1;
-		uint32_t load:1;
-		uint32_t readonly:1;
-		uint32_t code:1;
-		uint32_t data:1;
-		uint32_t _rsvd0:2;
-		uint32_t type:4;	/* MAN_SEGMENT_ */
-		uint32_t _rsvd1:4;
-		uint32_t length:16;	/* of segment in pages */
-	} r;
-} __packed;
-
-/*
- * Module segment descriptor. Used by ROM - Immutable.
- */
-struct sof_man_segment_desc {
-	union sof_man_segment_flags flags;
-	uint32_t v_base_addr;
-	uint32_t file_offset;
-} __packed;
-
-/*
- * The firmware binary can be split into several modules.
- */
-
-#define SOF_MAN_MOD_ID_LEN		4
-#define SOF_MAN_MOD_NAME_LEN		8
-#define SOF_MAN_MOD_SHA256_LEN		32
-#define SOF_MAN_MOD_ID			{'$', 'A', 'M', 'E'}
-
-/*
- * Each module has an entry in the FW header. Used by ROM - Immutable.
- */
-struct sof_man_module {
-	uint8_t struct_id[SOF_MAN_MOD_ID_LEN];	/* SOF_MAN_MOD_ID */
-	uint8_t name[SOF_MAN_MOD_NAME_LEN];
-	uint8_t uuid[16];
-	struct sof_man_module_type type;
-	uint8_t hash[SOF_MAN_MOD_SHA256_LEN];
-	uint32_t entry_point;
-	uint16_t cfg_offset;
-	uint16_t cfg_count;
-	uint32_t affinity_mask;
-	uint16_t instance_max_count;	/* max number of instances */
-	uint16_t instance_bss_size;	/* instance (pages) */
-	struct sof_man_segment_desc segment[3];
-} __packed;
-
-/*
- * Each module has a configuration in the FW header. Used by ROM - Immutable.
- */
-struct sof_man_mod_config {
-	uint32_t par[4];	/* module parameters */
-	uint32_t is_pages;	/* actual size of instance .bss (pages) */
-	uint32_t cps;		/* cycles per second */
-	uint32_t ibs;		/* input buffer size (bytes) */
-	uint32_t obs;		/* output buffer size (bytes) */
-	uint32_t module_flags;	/* flags, reserved for future use */
-	uint32_t cpc;		/* cycles per single run */
-	uint32_t obls;		/* output block size, reserved for future use */
-} __packed;
-
-/*
- * FW Manifest Header
- */
-
-#define SOF_MAN_FW_HDR_FW_NAME_LEN	8
-#define SOF_MAN_FW_HDR_ID		{'$', 'A', 'M', '1'}
-#define SOF_MAN_FW_HDR_NAME		"ADSPFW"
-#define SOF_MAN_FW_HDR_FLAGS		0x0
-#define SOF_MAN_FW_HDR_FEATURES		0xff
-
-/*
- * The firmware has a standard header that is checked by the ROM on firmware
- * loading. preload_page_count is used by DMA code loader and is entire
- * image size on CNL. i.e. CNL: total size of the binary’s .text and .rodata
- * Used by ROM - Immutable.
- */
-struct sof_man_fw_header {
-	uint8_t header_id[4];
-	uint32_t header_len;
-	uint8_t name[SOF_MAN_FW_HDR_FW_NAME_LEN];
-	/* number of pages of preloaded image loaded by driver */
-	uint32_t preload_page_count;
-	uint32_t fw_image_flags;
-	uint32_t feature_mask;
-	uint16_t major_version;
-	uint16_t minor_version;
-	uint16_t hotfix_version;
-	uint16_t build_version;
-	uint32_t num_module_entries;
-	uint32_t hw_buf_base_addr;
-	uint32_t hw_buf_length;
-	/* target address for binary loading as offset in IMR - must be == base offset */
-	uint32_t load_offset;
-} __packed;
-
-/*
- * Firmware manifest descriptor. This can contain N modules and N module
- * configs. Used by ROM - Immutable.
- */
-struct sof_man_fw_desc {
-	struct sof_man_fw_header header;
-
-	/* Warning - hack for module arrays. For some unknown reason the we
-	 * have a variable size array of struct man_module followed by a
-	 * variable size array of struct mod_config. These should have been
-	 * merged into a variable array of a parent structure. We have to hack
-	 * around this in many places....
-	 *
-	 * struct sof_man_module man_module[];
-	 * struct sof_man_mod_config mod_config[];
-	 */
-
-} __packed;
-
-/*
- * Component Descriptor. Used by ROM - Immutable.
- */
-struct sof_man_component_desc {
-	uint32_t reserved[2];	/* all 0 */
-	uint32_t version;
-	uint8_t hash[SOF_MAN_MOD_SHA256_LEN];
-	uint32_t base_offset;
-	uint32_t limit_offset;
-	uint32_t attributes[4];
-} __packed;
-
-/*
- * Audio DSP extended metadata. Used by ROM - Immutable.
- */
-struct sof_man_adsp_meta_file_ext {
-	uint32_t ext_type;	/* always 17 for ADSP extension */
-	uint32_t ext_len;
-	uint32_t imr_type;
-	uint8_t reserved[16];	/* all 0 */
-	struct sof_man_component_desc comp_desc[1];
-} __packed;
-
-/*
- * Module Manifest for rimage module metadata. Not used by ROM.
- */
-struct sof_man_module_manifest {
-	struct sof_man_module module;
-	uint32_t text_size;
-} __packed;
-
-#endif

include/uapi/sound/sof/tokens.h

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

include/uapi/sound/sof/tone.h

-/* SPDX-License-Identifier: ((GPL-2.0 WITH Linux-syscall-note) OR BSD-3-Clause) */
-/*
-* This file is provided under a dual BSD/GPLv2 license.  When using or
-* redistributing this file, you may do so under either license.
-*
-* Copyright(c) 2018 Intel Corporation. All rights reserved.
-*/
-
-#ifndef __INCLUDE_UAPI_SOUND_SOF_USER_TONE_H__
-#define __INCLUDE_UAPI_SOUND_SOF_USER_TONE_H__
-
-#define SOF_TONE_IDX_FREQUENCY		0
-#define SOF_TONE_IDX_AMPLITUDE		1
-#define SOF_TONE_IDX_FREQ_MULT		2
-#define SOF_TONE_IDX_AMPL_MULT		3
-#define SOF_TONE_IDX_LENGTH		4
-#define SOF_TONE_IDX_PERIOD		5
-#define SOF_TONE_IDX_REPEATS		6
-#define SOF_TONE_IDX_LIN_RAMP_STEP	7
-
-#endif

include/uapi/sound/sof/trace.h

-/* SPDX-License-Identifier: ((GPL-2.0 WITH Linux-syscall-note) OR BSD-3-Clause) */
-/*
- * This file is provided under a dual BSD/GPLv2 license.  When using or
- * redistributing this file, you may do so under either license.
- *
- * Copyright(c) 2018 Intel Corporation. All rights reserved.
- */
-
-#ifndef __INCLUDE_UAPI_SOUND_SOF_USER_TRACE_H__
-#define __INCLUDE_UAPI_SOUND_SOF_USER_TRACE_H__
-
-/*
- * Host system time.
- *
- * This property is used by the driver to pass down information about
- * current system time. It is expressed in us.
- * FW translates timestamps (in log entries, probe pockets) to this time
- * domain.
- *
- * (cavs: SystemTime).
- */
-struct system_time {
-	uint32_t val_l;  /* Lower dword of current host time value */
-	uint32_t val_u;  /* Upper dword of current host time value */
-} __packed;
-
-#define LOG_ENABLE		1  /* Enable logging */
-#define LOG_DISABLE		0  /* Disable logging */
-
-#define LOG_LEVEL_CRITICAL	1  /* (FDK fatal) */
-#define LOG_LEVEL_VERBOSE	2
-
-/*
- * Layout of a log fifo.
- */
-struct log_buffer_layout {
-	uint32_t read_ptr;  /*read pointer */
-	uint32_t write_ptr; /* write pointer */
-	uint32_t buffer[0]; /* buffer */
-} __packed;
-
-/*
- * Log buffer status reported by FW.
- */
-struct log_buffer_status {
-	uint32_t core_id;  /* ID of core that logged to other half */
-} __packed;
-
-#define TRACE_ID_LENGTH 12
-
-/*
- *  Log entry header.
- *
- * The header is followed by an array of arguments (uint32_t[]).
- * Number of arguments is specified by the params_num field of log_entry
- */
-struct log_entry_header {
-	uint32_t id_0 : TRACE_ID_LENGTH;	/* e.g. Pipeline ID */
-	uint32_t id_1 : TRACE_ID_LENGTH;	/* e.g. Component ID */
-	uint32_t core_id : 8;		/* Reporting core's id */
-
-	uint64_t timestamp;		/* Timestamp (in dsp ticks) */
-	uint32_t log_entry_address;	/* Address of log entry in ELF */
-} __packed;
-
-#endif

sound/core/control.c

@@ -196,16 +196,12 @@ EXPORT_SYMBOL(snd_ctl_notify);
 static int snd_ctl_new(struct snd_kcontrol **kctl, unsigned int count,
 		       unsigned int access, struct snd_ctl_file *file)
 {
-	unsigned int size;
 	unsigned int idx;
 
 	if (count == 0 || count > MAX_CONTROL_COUNT)
 		return -EINVAL;
 
-	size  = sizeof(struct snd_kcontrol);
-	size += sizeof(struct snd_kcontrol_volatile) * count;
-
-	*kctl = kzalloc(size, GFP_KERNEL);
+	*kctl = kzalloc(struct_size(*kctl, vd, count), GFP_KERNEL);
 	if (!*kctl)
 		return -ENOMEM;
 

sound/core/oss/rate.c

@@ -323,8 +323,8 @@ int snd_pcm_plugin_build_rate(struct snd_pcm_substream *plug,
 
 	err = snd_pcm_plugin_build(plug, "rate conversion",
 				   src_format, dst_format,
-				   sizeof(struct rate_priv) +
-				   src_format->channels * sizeof(struct rate_channel),
+				   struct_size(data, channels,
+					       src_format->channels),
 				   &plugin);
 	if (err < 0)
 		return err;

sound/firewire/amdtp-am824.c

@@ -82,7 +82,7 @@ int amdtp_am824_set_parameters(struct amdtp_stream *s, unsigned int rate,
 	if (err < 0)
 		return err;
 
-	s->fdf = AMDTP_FDF_AM824 | s->sfc;
+	s->ctx_data.rx.fdf = AMDTP_FDF_AM824 | s->sfc;
 
 	p->pcm_channels = pcm_channels;
 	p->midi_ports = midi_ports;
@@ -320,7 +320,7 @@ static void read_midi_messages(struct amdtp_stream *s,
 	u8 *b;
 
 	for (f = 0; f < frames; f++) {
-		port = (8 - s->tx_first_dbc + s->data_block_counter + f) % 8;
+		port = (8 - s->ctx_data.tx.first_dbc + s->data_block_counter + f) % 8;
 		b = (u8 *)&buffer[p->midi_position];
 
 		len = b[0] - 0x80;

sound/firewire/amdtp-stream-trace.h

@@ -13,147 +13,16 @@
 
 #include <linux/tracepoint.h>
 
-TRACE_EVENT(in_packet,
-	TP_PROTO(const struct amdtp_stream *s, u32 cycles, u32 *cip_header, unsigned int payload_length, unsigned int index),
-	TP_ARGS(s, cycles, cip_header, payload_length, index),
-	TP_STRUCT__entry(
-		__field(unsigned int, second)
-		__field(unsigned int, cycle)
-		__field(int, channel)
-		__field(int, src)
-		__field(int, dest)
-		__field(u32, cip_header0)
-		__field(u32, cip_header1)
-		__field(unsigned int, payload_quadlets)
-		__field(unsigned int, packet_index)
-		__field(unsigned int, irq)
-		__field(unsigned int, index)
-	),
-	TP_fast_assign(
-		__entry->second = cycles / CYCLES_PER_SECOND;
-		__entry->cycle = cycles % CYCLES_PER_SECOND;
-		__entry->channel = s->context->channel;
-		__entry->src = fw_parent_device(s->unit)->node_id;
-		__entry->dest = fw_parent_device(s->unit)->card->node_id;
-		__entry->cip_header0 = cip_header[0];
-		__entry->cip_header1 = cip_header[1];
-		__entry->payload_quadlets = payload_length / 4;
-		__entry->packet_index = s->packet_index;
-		__entry->irq = !!in_interrupt();
-		__entry->index = index;
-	),
-	TP_printk(
-		"%02u %04u %04x %04x %02d %08x %08x %03u %02u %01u %02u",
-		__entry->second,
-		__entry->cycle,
-		__entry->src,
-		__entry->dest,
-		__entry->channel,
-		__entry->cip_header0,
-		__entry->cip_header1,
-		__entry->payload_quadlets,
-		__entry->packet_index,
-		__entry->irq,
-		__entry->index)
-);
-
-TRACE_EVENT(out_packet,
-	TP_PROTO(const struct amdtp_stream *s, u32 cycles, __be32 *cip_header, unsigned int payload_length, unsigned int index),
-	TP_ARGS(s, cycles, cip_header, payload_length, index),
-	TP_STRUCT__entry(
-		__field(unsigned int, second)
-		__field(unsigned int, cycle)
-		__field(int, channel)
-		__field(int, src)
-		__field(int, dest)
-		__field(u32, cip_header0)
-		__field(u32, cip_header1)
-		__field(unsigned int, payload_quadlets)
-		__field(unsigned int, packet_index)
-		__field(unsigned int, irq)
-		__field(unsigned int, index)
-	),
-	TP_fast_assign(
-		__entry->second = cycles / CYCLES_PER_SECOND;
-		__entry->cycle = cycles % CYCLES_PER_SECOND;
-		__entry->channel = s->context->channel;
-		__entry->src = fw_parent_device(s->unit)->card->node_id;
-		__entry->dest = fw_parent_device(s->unit)->node_id;
-		__entry->cip_header0 = be32_to_cpu(cip_header[0]);
-		__entry->cip_header1 = be32_to_cpu(cip_header[1]);
-		__entry->payload_quadlets = payload_length / 4;
-		__entry->packet_index = s->packet_index;
-		__entry->irq = !!in_interrupt();
-		__entry->index = index;
-	),
-	TP_printk(
-		"%02u %04u %04x %04x %02d %08x %08x %03u %02u %01u %02u",
-		__entry->second,
-		__entry->cycle,
-		__entry->src,
-		__entry->dest,
-		__entry->channel,
-		__entry->cip_header0,
-		__entry->cip_header1,
-		__entry->payload_quadlets,
-		__entry->packet_index,
-		__entry->irq,
-		__entry->index)
-);
-
-TRACE_EVENT(in_packet_without_header,
-	TP_PROTO(const struct amdtp_stream *s, u32 cycles, unsigned int payload_quadlets, unsigned int data_blocks, unsigned int index),
-	TP_ARGS(s, cycles, payload_quadlets, data_blocks, index),
-	TP_STRUCT__entry(
-		__field(unsigned int, second)
-		__field(unsigned int, cycle)
-		__field(int, channel)
-		__field(int, src)
-		__field(int, dest)
-		__field(unsigned int, payload_quadlets)
-		__field(unsigned int, data_blocks)
-		__field(unsigned int, data_block_counter)
-		__field(unsigned int, packet_index)
-		__field(unsigned int, irq)
-		__field(unsigned int, index)
-	),
-	TP_fast_assign(
-		__entry->second = cycles / CYCLES_PER_SECOND;
-		__entry->cycle = cycles % CYCLES_PER_SECOND;
-		__entry->channel = s->context->channel;
-		__entry->src = fw_parent_device(s->unit)->node_id;
-		__entry->dest = fw_parent_device(s->unit)->card->node_id;
-		__entry->payload_quadlets = payload_quadlets;
-		__entry->data_blocks = data_blocks,
-		__entry->data_block_counter = s->data_block_counter,
-		__entry->packet_index = s->packet_index;
-		__entry->irq = !!in_interrupt();
-		__entry->index = index;
-	),
-	TP_printk(
-		"%02u %04u %04x %04x %02d %03u %02u %03u %02u %01u %02u",
-		__entry->second,
-		__entry->cycle,
-		__entry->src,
-		__entry->dest,
-		__entry->channel,
-		__entry->payload_quadlets,
-		__entry->data_blocks,
-		__entry->data_block_counter,
-		__entry->packet_index,
-		__entry->irq,
-		__entry->index)
-);
-
-TRACE_EVENT(out_packet_without_header,
-	TP_PROTO(const struct amdtp_stream *s, u32 cycles, unsigned int payload_length, unsigned int data_blocks, unsigned int index),
-	TP_ARGS(s, cycles, payload_length, data_blocks, index),
+TRACE_EVENT(amdtp_packet,
+	TP_PROTO(const struct amdtp_stream *s, u32 cycles, const __be32 *cip_header, unsigned int payload_length, unsigned int data_blocks, unsigned int index),
+	TP_ARGS(s, cycles, cip_header, payload_length, data_blocks, index),
 	TP_STRUCT__entry(
 		__field(unsigned int, second)
 		__field(unsigned int, cycle)
 		__field(int, channel)
 		__field(int, src)
 		__field(int, dest)
+		__dynamic_array(u8, cip_header, cip_header ? 8 : 0)
 		__field(unsigned int, payload_quadlets)
 		__field(unsigned int, data_blocks)
 		__field(unsigned int, data_block_counter)
@@ -165,17 +34,26 @@ TRACE_EVENT(out_packet_without_header,
 		__entry->second = cycles / CYCLES_PER_SECOND;
 		__entry->cycle = cycles % CYCLES_PER_SECOND;
 		__entry->channel = s->context->channel;
-		__entry->src = fw_parent_device(s->unit)->card->node_id;
-		__entry->dest = fw_parent_device(s->unit)->node_id;
-		__entry->payload_quadlets = payload_length / 4;
-		__entry->data_blocks = data_blocks,
+		if (s->direction == AMDTP_IN_STREAM) {
+			__entry->src = fw_parent_device(s->unit)->node_id;
+			__entry->dest = fw_parent_device(s->unit)->card->node_id;
+		} else {
+			__entry->src = fw_parent_device(s->unit)->card->node_id;
+			__entry->dest = fw_parent_device(s->unit)->node_id;
+		}
+		if (cip_header) {
+			memcpy(__get_dynamic_array(cip_header), cip_header,
+			       __get_dynamic_array_len(cip_header));
+		}
+		__entry->payload_quadlets = payload_length / sizeof(__be32);
+		__entry->data_blocks = data_blocks;
 		__entry->data_block_counter = s->data_block_counter,
 		__entry->packet_index = s->packet_index;
 		__entry->irq = !!in_interrupt();
 		__entry->index = index;
 	),
 	TP_printk(
-		"%02u %04u %04x %04x %02d %03u %02u %03u %02u %01u %02u",
+		"%02u %04u %04x %04x %02d %03u %02u %03u %02u %01u %02u %s",
 		__entry->second,
 		__entry->cycle,
 		__entry->src,
@@ -186,7 +64,10 @@ TRACE_EVENT(out_packet_without_header,
 		__entry->data_block_counter,
 		__entry->packet_index,
 		__entry->irq,
-		__entry->index)
+		__entry->index,
+		__print_array(__get_dynamic_array(cip_header),
+			      __get_dynamic_array_len(cip_header),
+			      sizeof(u8)))
 );
 
 #endif

sound/firewire/amdtp-stream.h

@@ -108,10 +108,31 @@ struct amdtp_stream {
 	struct iso_packets_buffer buffer;
 	int packet_index;
 	int tag;
-	int (*handle_packet)(struct amdtp_stream *s,
-			unsigned int payload_quadlets, unsigned int cycle,
-			unsigned int index);
-	unsigned int max_payload_length;
+	union {
+		struct {
+			unsigned int ctx_header_size;
+
+			// limit for payload of iso packet.
+			unsigned int max_ctx_payload_length;
+
+			// For quirks of CIP headers.
+			// Fixed interval of dbc between previos/current
+			// packets.
+			unsigned int dbc_interval;
+			// Indicate the value of dbc field in a first packet.
+			unsigned int first_dbc;
+		} tx;
+		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;
+
+			// To generate CIP header.
+			unsigned int fdf;
+		} rx;
+	} ctx_data;
 
 	/* For CIP headers. */
 	unsigned int source_node_id_field;
@@ -119,19 +140,10 @@ struct amdtp_stream {
 	unsigned int data_block_counter;
 	unsigned int sph;
 	unsigned int fmt;
-	unsigned int fdf;
-	/* quirk: fixed interval of dbc between previos/current packets. */
-	unsigned int tx_dbc_interval;
-	/* quirk: indicate the value of dbc field in a first packet. */
-	unsigned int tx_first_dbc;
 
 	/* Internal flags. */
 	enum cip_sfc sfc;
 	unsigned int syt_interval;
-	unsigned int transfer_delay;
-	unsigned int data_block_state;
-	unsigned int last_syt_offset;
-	unsigned int syt_offset_state;
 
 	/* For a PCM substream processing. */
 	struct snd_pcm_substream *pcm;

sound/firewire/bebob/bebob.h

@@ -92,8 +92,6 @@ struct snd_bebob {
 	unsigned int midi_input_ports;
 	unsigned int midi_output_ports;
 
-	bool connected;
-
 	struct amdtp_stream tx_stream;
 	struct amdtp_stream rx_stream;
 	struct cmp_connection out_conn;
@@ -217,7 +215,8 @@ int snd_bebob_stream_get_clock_src(struct snd_bebob *bebob,
 				   enum snd_bebob_clock_type *src);
 int snd_bebob_stream_discover(struct snd_bebob *bebob);
 int snd_bebob_stream_init_duplex(struct snd_bebob *bebob);
-int snd_bebob_stream_start_duplex(struct snd_bebob *bebob, unsigned int rate);
+int snd_bebob_stream_reserve_duplex(struct snd_bebob *bebob, unsigned int rate);
+int snd_bebob_stream_start_duplex(struct snd_bebob *bebob);
 void snd_bebob_stream_stop_duplex(struct snd_bebob *bebob);
 void snd_bebob_stream_destroy_duplex(struct snd_bebob *bebob);
 

sound/firewire/bebob/bebob_midi.c

@@ -7,58 +7,31 @@
 
 #include "bebob.h"
 
-static int midi_capture_open(struct snd_rawmidi_substream *substream)
+static int midi_open(struct snd_rawmidi_substream *substream)
 {
 	struct snd_bebob *bebob = substream->rmidi->private_data;
 	int err;
 
 	err = snd_bebob_stream_lock_try(bebob);
 	if (err < 0)
-		goto end;
+		return err;
 
 	mutex_lock(&bebob->mutex);
-	bebob->substreams_counter++;
-	err = snd_bebob_stream_start_duplex(bebob, 0);
+	err = snd_bebob_stream_reserve_duplex(bebob, 0);
+	if (err >= 0) {
+		++bebob->substreams_counter;
+		err = snd_bebob_stream_start_duplex(bebob);
+		if (err < 0)
+			--bebob->substreams_counter;
+	}
 	mutex_unlock(&bebob->mutex);
 	if (err < 0)
 		snd_bebob_stream_lock_release(bebob);
-end:
-	return err;
-}
-
-static int midi_playback_open(struct snd_rawmidi_substream *substream)
-{
-	struct snd_bebob *bebob = substream->rmidi->private_data;
-	int err;
-
-	err = snd_bebob_stream_lock_try(bebob);
-	if (err < 0)
-		goto end;
 
-	mutex_lock(&bebob->mutex);
-	bebob->substreams_counter++;
-	err = snd_bebob_stream_start_duplex(bebob, 0);
-	mutex_unlock(&bebob->mutex);
-	if (err < 0)
-		snd_bebob_stream_lock_release(bebob);
-end:
 	return err;
 }
 
-static int midi_capture_close(struct snd_rawmidi_substream *substream)
-{
-	struct snd_bebob *bebob = substream->rmidi->private_data;
-
-	mutex_lock(&bebob->mutex);
-	bebob->substreams_counter--;
-	snd_bebob_stream_stop_duplex(bebob);
-	mutex_unlock(&bebob->mutex);
-
-	snd_bebob_stream_lock_release(bebob);
-	return 0;
-}
-
-static int midi_playback_close(struct snd_rawmidi_substream *substream)
+static int midi_close(struct snd_rawmidi_substream *substream)
 {
 	struct snd_bebob *bebob = substream->rmidi->private_data;
 
@@ -120,13 +93,13 @@ static void set_midi_substream_names(struct snd_bebob *bebob,
 int snd_bebob_create_midi_devices(struct snd_bebob *bebob)
 {
 	static const struct snd_rawmidi_ops capture_ops = {
-		.open		= midi_capture_open,
-		.close		= midi_capture_close,
+		.open		= midi_open,
+		.close		= midi_close,
 		.trigger	= midi_capture_trigger,
 	};
 	static const struct snd_rawmidi_ops playback_ops = {
-		.open		= midi_playback_open,
-		.close		= midi_playback_close,
+		.open		= midi_open,
+		.close		= midi_close,
 		.trigger	= midi_playback_trigger,
 	};
 	struct snd_rawmidi *rmidi;

sound/firewire/bebob/bebob_pcm.c

@@ -184,9 +184,8 @@ pcm_close(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-static int
-pcm_capture_hw_params(struct snd_pcm_substream *substream,
-		      struct snd_pcm_hw_params *hw_params)
+static int pcm_hw_params(struct snd_pcm_substream *substream,
+			 struct snd_pcm_hw_params *hw_params)
 {
 	struct snd_bebob *bebob = substream->private_data;
 	int err;
@@ -197,62 +196,31 @@ pcm_capture_hw_params(struct snd_pcm_substream *substream,
 		return err;
 
 	if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) {
-		mutex_lock(&bebob->mutex);
-		bebob->substreams_counter++;
-		mutex_unlock(&bebob->mutex);
-	}
+		unsigned int rate = params_rate(hw_params);
 
-	return 0;
-}
-static int
-pcm_playback_hw_params(struct snd_pcm_substream *substream,
-		       struct snd_pcm_hw_params *hw_params)
-{
-	struct snd_bebob *bebob = substream->private_data;
-	int err;
-
-	err = snd_pcm_lib_alloc_vmalloc_buffer(substream,
-					       params_buffer_bytes(hw_params));
-	if (err < 0)
-		return err;
-
-	if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) {
 		mutex_lock(&bebob->mutex);
-		bebob->substreams_counter++;
+		err = snd_bebob_stream_reserve_duplex(bebob, rate);
+		if (err >= 0)
+			++bebob->substreams_counter;
 		mutex_unlock(&bebob->mutex);
 	}
 
-	return 0;
+	return err;
 }
 
-static int
-pcm_capture_hw_free(struct snd_pcm_substream *substream)
+static int pcm_hw_free(struct snd_pcm_substream *substream)
 {
 	struct snd_bebob *bebob = substream->private_data;
 
-	if (substream->runtime->status->state != SNDRV_PCM_STATE_OPEN) {
-		mutex_lock(&bebob->mutex);
-		bebob->substreams_counter--;
-		mutex_unlock(&bebob->mutex);
-	}
-
-	snd_bebob_stream_stop_duplex(bebob);
+	mutex_lock(&bebob->mutex);
 
-	return snd_pcm_lib_free_vmalloc_buffer(substream);
-}
-static int
-pcm_playback_hw_free(struct snd_pcm_substream *substream)
-{
-	struct snd_bebob *bebob = substream->private_data;
-
-	if (substream->runtime->status->state != SNDRV_PCM_STATE_OPEN) {
-		mutex_lock(&bebob->mutex);
+	if (substream->runtime->status->state != SNDRV_PCM_STATE_OPEN)
 		bebob->substreams_counter--;
-		mutex_unlock(&bebob->mutex);
-	}
 
 	snd_bebob_stream_stop_duplex(bebob);
 
+	mutex_unlock(&bebob->mutex);
+
 	return snd_pcm_lib_free_vmalloc_buffer(substream);
 }
 
@@ -260,10 +228,9 @@ static int
 pcm_capture_prepare(struct snd_pcm_substream *substream)
 {
 	struct snd_bebob *bebob = substream->private_data;
-	struct snd_pcm_runtime *runtime = substream->runtime;
 	int err;
 
-	err = snd_bebob_stream_start_duplex(bebob, runtime->rate);
+	err = snd_bebob_stream_start_duplex(bebob);
 	if (err >= 0)
 		amdtp_stream_pcm_prepare(&bebob->tx_stream);
 
@@ -273,10 +240,9 @@ static int
 pcm_playback_prepare(struct snd_pcm_substream *substream)
 {
 	struct snd_bebob *bebob = substream->private_data;
-	struct snd_pcm_runtime *runtime = substream->runtime;
 	int err;
 
-	err = snd_bebob_stream_start_duplex(bebob, runtime->rate);
+	err = snd_bebob_stream_start_duplex(bebob);
 	if (err >= 0)
 		amdtp_stream_pcm_prepare(&bebob->rx_stream);
 
@@ -353,8 +319,8 @@ int snd_bebob_create_pcm_devices(struct snd_bebob *bebob)
 		.open		= pcm_open,
 		.close		= pcm_close,
 		.ioctl		= snd_pcm_lib_ioctl,
-		.hw_params	= pcm_capture_hw_params,
-		.hw_free	= pcm_capture_hw_free,
+		.hw_params	= pcm_hw_params,
+		.hw_free	= pcm_hw_free,
 		.prepare	= pcm_capture_prepare,
 		.trigger	= pcm_capture_trigger,
 		.pointer	= pcm_capture_pointer,
@@ -365,8 +331,8 @@ int snd_bebob_create_pcm_devices(struct snd_bebob *bebob)
 		.open		= pcm_open,
 		.close		= pcm_close,
 		.ioctl		= snd_pcm_lib_ioctl,
-		.hw_params	= pcm_playback_hw_params,
-		.hw_free	= pcm_playback_hw_free,
+		.hw_params	= pcm_hw_params,
+		.hw_free	= pcm_hw_free,
 		.prepare	= pcm_playback_prepare,
 		.trigger	= pcm_playback_trigger,
 		.pointer	= pcm_playback_pointer,

sound/firewire/cmp.c

@@ -185,6 +185,37 @@ void cmp_connection_destroy(struct cmp_connection *c)
 }
 EXPORT_SYMBOL(cmp_connection_destroy);
 
+int cmp_connection_reserve(struct cmp_connection *c,
+			   unsigned int max_payload_bytes)
+{
+	int err;
+
+	mutex_lock(&c->mutex);
+
+	if (WARN_ON(c->resources.allocated)) {
+		err = -EBUSY;
+		goto end;
+	}
+
+	c->speed = min(c->max_speed,
+		       fw_parent_device(c->resources.unit)->max_speed);
+
+	err = fw_iso_resources_allocate(&c->resources, max_payload_bytes,
+					c->speed);
+end:
+	mutex_unlock(&c->mutex);
+
+	return err;
+}
+EXPORT_SYMBOL(cmp_connection_reserve);
+
+void cmp_connection_release(struct cmp_connection *c)
+{
+	mutex_lock(&c->mutex);
+	fw_iso_resources_free(&c->resources);
+	mutex_unlock(&c->mutex);
+}
+EXPORT_SYMBOL(cmp_connection_release);
 
 static __be32 ipcr_set_modify(struct cmp_connection *c, __be32 ipcr)
 {
@@ -270,25 +301,18 @@ static int pcr_set_check(struct cmp_connection *c, __be32 pcr)
  * When this function succeeds, the caller is responsible for starting
  * transmitting packets.
  */
-int cmp_connection_establish(struct cmp_connection *c,
-			     unsigned int max_payload_bytes)
+int cmp_connection_establish(struct cmp_connection *c)
 {
 	int err;
 
-	if (WARN_ON(c->connected))
-		return -EISCONN;
-
-	c->speed = min(c->max_speed,
-		       fw_parent_device(c->resources.unit)->max_speed);
-
 	mutex_lock(&c->mutex);
 
-retry_after_bus_reset:
-	err = fw_iso_resources_allocate(&c->resources,
-					max_payload_bytes, c->speed);
-	if (err < 0)
-		goto err_mutex;
+	if (WARN_ON(c->connected)) {
+		mutex_unlock(&c->mutex);
+		return -EISCONN;
+	}
 
+retry_after_bus_reset:
 	if (c->direction == CMP_OUTPUT)
 		err = pcr_modify(c, opcr_set_modify, pcr_set_check,
 				 ABORT_ON_BUS_RESET);
@@ -297,21 +321,13 @@ int cmp_connection_establish(struct cmp_connection *c,
 				 ABORT_ON_BUS_RESET);
 
 	if (err == -EAGAIN) {
-		fw_iso_resources_free(&c->resources);
-		goto retry_after_bus_reset;
+		err = fw_iso_resources_update(&c->resources);
+		if (err >= 0)
+			goto retry_after_bus_reset;
 	}
-	if (err < 0)
-		goto err_resources;
-
-	c->connected = true;
-
-	mutex_unlock(&c->mutex);
-
-	return 0;
+	if (err >= 0)
+		c->connected = true;
 
-err_resources:
-	fw_iso_resources_free(&c->resources);
-err_mutex:
 	mutex_unlock(&c->mutex);
 
 	return err;
@@ -351,14 +367,12 @@ int cmp_connection_update(struct cmp_connection *c)
 				 SUCCEED_ON_BUS_RESET);
 
 	if (err < 0)
-		goto err_resources;
+		goto err_unconnect;
 
 	mutex_unlock(&c->mutex);
 
 	return 0;
 
-err_resources:
-	fw_iso_resources_free(&c->resources);
 err_unconnect:
 	c->connected = false;
 	mutex_unlock(&c->mutex);
@@ -395,8 +409,6 @@ void cmp_connection_break(struct cmp_connection *c)
 	if (err < 0)
 		cmp_error(c, "plug is still connected\n");
 
-	fw_iso_resources_free(&c->resources);
-
 	c->connected = false;
 
 	mutex_unlock(&c->mutex);

sound/firewire/cmp.h

@@ -42,8 +42,11 @@ int cmp_connection_init(struct cmp_connection *connection,
 int cmp_connection_check_used(struct cmp_connection *connection, bool *used);
 void cmp_connection_destroy(struct cmp_connection *connection);
 
-int cmp_connection_establish(struct cmp_connection *connection,
-			     unsigned int max_payload);
+int cmp_connection_reserve(struct cmp_connection *connection,
+			   unsigned int max_payload);
+void cmp_connection_release(struct cmp_connection *connection);
+
+int cmp_connection_establish(struct cmp_connection *connection);
 int cmp_connection_update(struct cmp_connection *connection);
 void cmp_connection_break(struct cmp_connection *connection);
 

sound/firewire/dice/Makefile

 # SPDX-License-Identifier: GPL-2.0-only
 snd-dice-objs := dice-transaction.o dice-stream.o dice-proc.o dice-midi.o \
 		 dice-pcm.o dice-hwdep.o dice.o dice-tcelectronic.o \
-		 dice-alesis.o dice-extension.o dice-mytek.o
+		 dice-alesis.o dice-extension.o dice-mytek.o dice-presonus.o
 obj-$(CONFIG_SND_DICE) += snd-dice.o

sound/firewire/dice/dice-midi.c

@@ -17,8 +17,13 @@ static int midi_open(struct snd_rawmidi_substream *substream)
 
 	mutex_lock(&dice->mutex);
 
-	dice->substreams_counter++;
-	err = snd_dice_stream_start_duplex(dice, 0);
+	err = snd_dice_stream_reserve_duplex(dice, 0);
+	if (err >= 0) {
+		++dice->substreams_counter;
+		err = snd_dice_stream_start_duplex(dice);
+		if (err < 0)
+			--dice->substreams_counter;
+	}
 
 	mutex_unlock(&dice->mutex);
 
@@ -34,7 +39,7 @@ static int midi_close(struct snd_rawmidi_substream *substream)
 
 	mutex_lock(&dice->mutex);
 
-	dice->substreams_counter--;
+	--dice->substreams_counter;
 	snd_dice_stream_stop_duplex(dice);
 
 	mutex_unlock(&dice->mutex);

sound/firewire/dice/dice-pcm.c

@@ -230,8 +230,8 @@ static int pcm_close(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-static int capture_hw_params(struct snd_pcm_substream *substream,
-			     struct snd_pcm_hw_params *hw_params)
+static int pcm_hw_params(struct snd_pcm_substream *substream,
+			 struct snd_pcm_hw_params *hw_params)
 {
 	struct snd_dice *dice = substream->private_data;
 	int err;
@@ -242,57 +242,26 @@ static int capture_hw_params(struct snd_pcm_substream *substream,
 		return err;
 
 	if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) {
-		mutex_lock(&dice->mutex);
-		dice->substreams_counter++;
-		mutex_unlock(&dice->mutex);
-	}
-
-	return 0;
-}
-static int playback_hw_params(struct snd_pcm_substream *substream,
-			      struct snd_pcm_hw_params *hw_params)
-{
-	struct snd_dice *dice = substream->private_data;
-	int err;
-
-	err = snd_pcm_lib_alloc_vmalloc_buffer(substream,
-					       params_buffer_bytes(hw_params));
-	if (err < 0)
-		return err;
+		unsigned int rate = params_rate(hw_params);
 
-	if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) {
 		mutex_lock(&dice->mutex);
-		dice->substreams_counter++;
+		err = snd_dice_stream_reserve_duplex(dice, rate);
+		if (err >= 0)
+			++dice->substreams_counter;
 		mutex_unlock(&dice->mutex);
 	}
 
-	return 0;
-}
-
-static int capture_hw_free(struct snd_pcm_substream *substream)
-{
-	struct snd_dice *dice = substream->private_data;
-
-	mutex_lock(&dice->mutex);
-
-	if (substream->runtime->status->state != SNDRV_PCM_STATE_OPEN)
-		dice->substreams_counter--;
-
-	snd_dice_stream_stop_duplex(dice);
-
-	mutex_unlock(&dice->mutex);
-
-	return snd_pcm_lib_free_vmalloc_buffer(substream);
+	return err;
 }
 
-static int playback_hw_free(struct snd_pcm_substream *substream)
+static int pcm_hw_free(struct snd_pcm_substream *substream)
 {
 	struct snd_dice *dice = substream->private_data;
 
 	mutex_lock(&dice->mutex);
 
 	if (substream->runtime->status->state != SNDRV_PCM_STATE_OPEN)
-		dice->substreams_counter--;
+		--dice->substreams_counter;
 
 	snd_dice_stream_stop_duplex(dice);
 
@@ -308,7 +277,7 @@ static int capture_prepare(struct snd_pcm_substream *substream)
 	int err;
 
 	mutex_lock(&dice->mutex);
-	err = snd_dice_stream_start_duplex(dice, substream->runtime->rate);
+	err = snd_dice_stream_start_duplex(dice);
 	mutex_unlock(&dice->mutex);
 	if (err >= 0)
 		amdtp_stream_pcm_prepare(stream);
@@ -322,7 +291,7 @@ static int playback_prepare(struct snd_pcm_substream *substream)
 	int err;
 
 	mutex_lock(&dice->mutex);
-	err = snd_dice_stream_start_duplex(dice, substream->runtime->rate);
+	err = snd_dice_stream_start_duplex(dice);
 	mutex_unlock(&dice->mutex);
 	if (err >= 0)
 		amdtp_stream_pcm_prepare(stream);
@@ -404,8 +373,8 @@ int snd_dice_create_pcm(struct snd_dice *dice)
 		.open      = pcm_open,
 		.close     = pcm_close,
 		.ioctl     = snd_pcm_lib_ioctl,
-		.hw_params = capture_hw_params,
-		.hw_free   = capture_hw_free,
+		.hw_params = pcm_hw_params,
+		.hw_free   = pcm_hw_free,
 		.prepare   = capture_prepare,
 		.trigger   = capture_trigger,
 		.pointer   = capture_pointer,
@@ -416,8 +385,8 @@ int snd_dice_create_pcm(struct snd_dice *dice)
 		.open      = pcm_open,
 		.close     = pcm_close,
 		.ioctl     = snd_pcm_lib_ioctl,
-		.hw_params = playback_hw_params,
-		.hw_free   = playback_hw_free,
+		.hw_params = pcm_hw_params,
+		.hw_free   = pcm_hw_free,
 		.prepare   = playback_prepare,
 		.trigger   = playback_trigger,
 		.pointer   = playback_pointer,

sound/firewire/dice/dice-presonus.c

+// SPDX-License-Identifier: GPL-2.0
+// dice-presonus.c - a part of driver for DICE based devices
+//
+// Copyright (c) 2019 Takashi Sakamoto
+//
+// Licensed under the terms of the GNU General Public License, version 2.
+
+#include "dice.h"
+
+struct dice_presonus_spec {
+	unsigned int tx_pcm_chs[MAX_STREAMS][SND_DICE_RATE_MODE_COUNT];
+	unsigned int rx_pcm_chs[MAX_STREAMS][SND_DICE_RATE_MODE_COUNT];
+	bool has_midi;
+};
+
+static const struct dice_presonus_spec dice_presonus_firesutio = {
+	.tx_pcm_chs = {{16, 16, 0}, {10, 2, 0} },
+	.rx_pcm_chs = {{16, 16, 0}, {10, 2, 0} },
+	.has_midi = true,
+};
+
+int snd_dice_detect_presonus_formats(struct snd_dice *dice)
+{
+	static const struct {
+		u32 model_id;
+		const struct dice_presonus_spec *spec;
+	} *entry, entries[] = {
+		{0x000008, &dice_presonus_firesutio},
+	};
+	struct fw_csr_iterator it;
+	int key, val, model_id;
+	int i;
+
+	model_id = 0;
+	fw_csr_iterator_init(&it, dice->unit->directory);
+	while (fw_csr_iterator_next(&it, &key, &val)) {
+		if (key == CSR_MODEL) {
+			model_id = val;
+			break;
+		}
+	}
+
+	for (i = 0; i < ARRAY_SIZE(entries); ++i) {
+		entry = entries + i;
+		if (entry->model_id == model_id)
+			break;
+	}
+	if (i == ARRAY_SIZE(entries))
+		return -ENODEV;
+
+	memcpy(dice->tx_pcm_chs, entry->spec->tx_pcm_chs,
+	       MAX_STREAMS * SND_DICE_RATE_MODE_COUNT * sizeof(unsigned int));
+	memcpy(dice->rx_pcm_chs, entry->spec->rx_pcm_chs,
+	       MAX_STREAMS * SND_DICE_RATE_MODE_COUNT * sizeof(unsigned int));
+
+	if (entry->spec->has_midi) {
+		dice->tx_midi_ports[0] = 1;
+		dice->rx_midi_ports[0] = 1;
+	}
+
+	return 0;
+}

sound/firewire/dice/dice.c

@@ -19,6 +19,7 @@ MODULE_LICENSE("GPL v2");
 #define OUI_MAUDIO		0x000d6c
 #define OUI_MYTEK		0x001ee8
 #define OUI_SSL			0x0050c2	// Actually ID reserved by IEEE.
+#define OUI_PRESONUS		0x000a92
 
 #define DICE_CATEGORY_ID	0x04
 #define WEISS_CATEGORY_ID	0x00
@@ -371,6 +372,14 @@ static const struct ieee1394_device_id dice_id_table[] = {
 		.vendor_id	= OUI_SSL,
 		.model_id	= 0x000070,
 	},
+	// Presonus FireStudio.
+	{
+		.match_flags	= IEEE1394_MATCH_VENDOR_ID |
+				  IEEE1394_MATCH_MODEL_ID,
+		.vendor_id	= OUI_PRESONUS,
+		.model_id	= 0x000008,
+		.driver_data	= (kernel_ulong_t)snd_dice_detect_presonus_formats,
+	},
 	{
 		.match_flags = IEEE1394_MATCH_VERSION,
 		.version     = DICE_INTERFACE,

sound/firewire/dice/dice.h

@@ -204,10 +204,11 @@ extern const unsigned int snd_dice_rates[SND_DICE_RATES_COUNT];
 
 int snd_dice_stream_get_rate_mode(struct snd_dice *dice, unsigned int rate,
 				  enum snd_dice_rate_mode *mode);
-int snd_dice_stream_start_duplex(struct snd_dice *dice, unsigned int rate);
+int snd_dice_stream_start_duplex(struct snd_dice *dice);
 void snd_dice_stream_stop_duplex(struct snd_dice *dice);
 int snd_dice_stream_init_duplex(struct snd_dice *dice);
 void snd_dice_stream_destroy_duplex(struct snd_dice *dice);
+int snd_dice_stream_reserve_duplex(struct snd_dice *dice, unsigned int rate);
 void snd_dice_stream_update_duplex(struct snd_dice *dice);
 int snd_dice_stream_detect_current_formats(struct snd_dice *dice);
 
@@ -226,5 +227,6 @@ int snd_dice_detect_tcelectronic_formats(struct snd_dice *dice);
 int snd_dice_detect_alesis_formats(struct snd_dice *dice);
 int snd_dice_detect_extension_formats(struct snd_dice *dice);
 int snd_dice_detect_mytek_formats(struct snd_dice *dice);
+int snd_dice_detect_presonus_formats(struct snd_dice *dice);
 
 #endif

sound/firewire/digi00x/amdtp-dot.c

@@ -127,7 +127,7 @@ int amdtp_dot_set_parameters(struct amdtp_stream *s, unsigned int rate,
 	if (err < 0)
 		return err;
 
-	s->fdf = AMDTP_FDF_AM824 | s->sfc;
+	s->ctx_data.rx.fdf = AMDTP_FDF_AM824 | s->sfc;
 
 	p->pcm_channels = pcm_channels;
 

sound/firewire/digi00x/digi00x-midi.c

@@ -17,8 +17,13 @@ static int midi_open(struct snd_rawmidi_substream *substream)
 		return err;
 
 	mutex_lock(&dg00x->mutex);
-	dg00x->substreams_counter++;
-	err = snd_dg00x_stream_start_duplex(dg00x, 0);
+	err = snd_dg00x_stream_reserve_duplex(dg00x, 0);
+	if (err >= 0) {
+		++dg00x->substreams_counter;
+		err = snd_dg00x_stream_start_duplex(dg00x);
+		if (err < 0)
+			--dg00x->substreams_counter;
+	}
 	mutex_unlock(&dg00x->mutex);
 	if (err < 0)
 		snd_dg00x_stream_lock_release(dg00x);
@@ -31,7 +36,7 @@ static int midi_close(struct snd_rawmidi_substream *substream)
 	struct snd_dg00x *dg00x = substream->rmidi->private_data;
 
 	mutex_lock(&dg00x->mutex);
-	dg00x->substreams_counter--;
+	--dg00x->substreams_counter;
 	snd_dg00x_stream_stop_duplex(dg00x);
 	mutex_unlock(&dg00x->mutex);
 

sound/firewire/digi00x/digi00x-pcm.c

@@ -154,8 +154,8 @@ static int pcm_close(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-static int pcm_capture_hw_params(struct snd_pcm_substream *substream,
-				 struct snd_pcm_hw_params *hw_params)
+static int pcm_hw_params(struct snd_pcm_substream *substream,
+			 struct snd_pcm_hw_params *hw_params)
 {
 	struct snd_dg00x *dg00x = substream->private_data;
 	int err;
@@ -166,58 +166,26 @@ static int pcm_capture_hw_params(struct snd_pcm_substream *substream,
 		return err;
 
 	if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) {
-		mutex_lock(&dg00x->mutex);
-		dg00x->substreams_counter++;
-		mutex_unlock(&dg00x->mutex);
-	}
+		unsigned int rate = params_rate(hw_params);
 
-	return 0;
-}
-
-static int pcm_playback_hw_params(struct snd_pcm_substream *substream,
-				  struct snd_pcm_hw_params *hw_params)
-{
-	struct snd_dg00x *dg00x = substream->private_data;
-	int err;
-
-	err = snd_pcm_lib_alloc_vmalloc_buffer(substream,
-					       params_buffer_bytes(hw_params));
-	if (err < 0)
-		return err;
-
-	if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) {
 		mutex_lock(&dg00x->mutex);
-		dg00x->substreams_counter++;
+		err = snd_dg00x_stream_reserve_duplex(dg00x, rate);
+		if (err >= 0)
+			++dg00x->substreams_counter;
 		mutex_unlock(&dg00x->mutex);
 	}
 
-	return 0;
-}
-
-static int pcm_capture_hw_free(struct snd_pcm_substream *substream)
-{
-	struct snd_dg00x *dg00x = substream->private_data;
-
-	mutex_lock(&dg00x->mutex);
-
-	if (substream->runtime->status->state != SNDRV_PCM_STATE_OPEN)
-		dg00x->substreams_counter--;
-
-	snd_dg00x_stream_stop_duplex(dg00x);
-
-	mutex_unlock(&dg00x->mutex);
-
-	return snd_pcm_lib_free_vmalloc_buffer(substream);
+	return err;
 }
 
-static int pcm_playback_hw_free(struct snd_pcm_substream *substream)
+static int pcm_hw_free(struct snd_pcm_substream *substream)
 {
 	struct snd_dg00x *dg00x = substream->private_data;
 
 	mutex_lock(&dg00x->mutex);
 
 	if (substream->runtime->status->state != SNDRV_PCM_STATE_OPEN)
-		dg00x->substreams_counter--;
+		--dg00x->substreams_counter;
 
 	snd_dg00x_stream_stop_duplex(dg00x);
 
@@ -229,12 +197,11 @@ static int pcm_playback_hw_free(struct snd_pcm_substream *substream)
 static int pcm_capture_prepare(struct snd_pcm_substream *substream)
 {
 	struct snd_dg00x *dg00x = substream->private_data;
-	struct snd_pcm_runtime *runtime = substream->runtime;
 	int err;
 
 	mutex_lock(&dg00x->mutex);
 
-	err = snd_dg00x_stream_start_duplex(dg00x, runtime->rate);
+	err = snd_dg00x_stream_start_duplex(dg00x);
 	if (err >= 0)
 		amdtp_stream_pcm_prepare(&dg00x->tx_stream);
 
@@ -246,12 +213,11 @@ static int pcm_capture_prepare(struct snd_pcm_substream *substream)
 static int pcm_playback_prepare(struct snd_pcm_substream *substream)
 {
 	struct snd_dg00x *dg00x = substream->private_data;
-	struct snd_pcm_runtime *runtime = substream->runtime;
 	int err;
 
 	mutex_lock(&dg00x->mutex);
 
-	err = snd_dg00x_stream_start_duplex(dg00x, runtime->rate);
+	err = snd_dg00x_stream_start_duplex(dg00x);
 	if (err >= 0) {
 		amdtp_stream_pcm_prepare(&dg00x->rx_stream);
 		amdtp_dot_reset(&dg00x->rx_stream);
@@ -332,8 +298,8 @@ int snd_dg00x_create_pcm_devices(struct snd_dg00x *dg00x)
 		.open		= pcm_open,
 		.close		= pcm_close,
 		.ioctl		= snd_pcm_lib_ioctl,
-		.hw_params	= pcm_capture_hw_params,
-		.hw_free	= pcm_capture_hw_free,
+		.hw_params	= pcm_hw_params,
+		.hw_free	= pcm_hw_free,
 		.prepare	= pcm_capture_prepare,
 		.trigger	= pcm_capture_trigger,
 		.pointer	= pcm_capture_pointer,
@@ -344,8 +310,8 @@ int snd_dg00x_create_pcm_devices(struct snd_dg00x *dg00x)
 		.open		= pcm_open,
 		.close		= pcm_close,
 		.ioctl		= snd_pcm_lib_ioctl,
-		.hw_params	= pcm_playback_hw_params,
-		.hw_free	= pcm_playback_hw_free,
+		.hw_params	= pcm_hw_params,
+		.hw_free	= pcm_hw_free,
 		.prepare	= pcm_playback_prepare,
 		.trigger	= pcm_playback_trigger,
 		.pointer	= pcm_playback_pointer,

sound/firewire/digi00x/digi00x-stream.c

@@ -124,11 +124,25 @@ int snd_dg00x_stream_get_external_rate(struct snd_dg00x *dg00x,
 
 static void finish_session(struct snd_dg00x *dg00x)
 {
-	__be32 data = cpu_to_be32(0x00000003);
+	__be32 data;
+
+	amdtp_stream_stop(&dg00x->tx_stream);
+	amdtp_stream_stop(&dg00x->rx_stream);
 
+	data = cpu_to_be32(0x00000003);
 	snd_fw_transaction(dg00x->unit, TCODE_WRITE_QUADLET_REQUEST,
 			   DG00X_ADDR_BASE + DG00X_OFFSET_STREAMING_SET,
 			   &data, sizeof(data), 0);
+
+	// Unregister isochronous channels for both direction.
+	data = 0;
+	snd_fw_transaction(dg00x->unit, TCODE_WRITE_QUADLET_REQUEST,
+			   DG00X_ADDR_BASE + DG00X_OFFSET_ISOC_CHANNELS,
+			   &data, sizeof(data), 0);
+
+	// Just after finishing the session, the device may lost transmitting
+	// functionality for a short time.
+	msleep(50);
 }
 
 static int begin_session(struct snd_dg00x *dg00x)
@@ -137,11 +151,20 @@ static int begin_session(struct snd_dg00x *dg00x)
 	u32 curr;
 	int err;
 
+	// Register isochronous channels for both direction.
+	data = cpu_to_be32((dg00x->tx_resources.channel << 16) |
+			   dg00x->rx_resources.channel);
+	err = snd_fw_transaction(dg00x->unit, TCODE_WRITE_QUADLET_REQUEST,
+				 DG00X_ADDR_BASE + DG00X_OFFSET_ISOC_CHANNELS,
+				 &data, sizeof(data), 0);
+	if (err < 0)
+		return err;
+
 	err = snd_fw_transaction(dg00x->unit, TCODE_READ_QUADLET_REQUEST,
 				 DG00X_ADDR_BASE + DG00X_OFFSET_STREAMING_STATE,
 				 &data, sizeof(data), 0);
 	if (err < 0)
-		goto error;
+		return err;
 	curr = be32_to_cpu(data);
 
 	if (curr == 0)
@@ -156,39 +179,23 @@ static int begin_session(struct snd_dg00x *dg00x)
 					 DG00X_OFFSET_STREAMING_SET,
 					 &data, sizeof(data), 0);
 		if (err < 0)
-			goto error;
+			break;
 
 		msleep(20);
 		curr--;
 	}
 
-	return 0;
-error:
-	finish_session(dg00x);
 	return err;
 }
 
-static void release_resources(struct snd_dg00x *dg00x)
+static int keep_resources(struct snd_dg00x *dg00x, struct amdtp_stream *stream,
+			  unsigned int rate)
 {
-	__be32 data = 0;
-
-	/* Unregister isochronous channels for both direction. */
-	snd_fw_transaction(dg00x->unit, TCODE_WRITE_QUADLET_REQUEST,
-			   DG00X_ADDR_BASE + DG00X_OFFSET_ISOC_CHANNELS,
-			   &data, sizeof(data), 0);
-
-	/* Release isochronous resources. */
-	fw_iso_resources_free(&dg00x->tx_resources);
-	fw_iso_resources_free(&dg00x->rx_resources);
-}
-
-static int keep_resources(struct snd_dg00x *dg00x, unsigned int rate)
-{
-	unsigned int i;
-	__be32 data;
+	struct fw_iso_resources *resources;
+	int i;
 	int err;
 
-	/* Check sampling rate. */
+	// Check sampling rate.
 	for (i = 0; i < SND_DG00X_RATE_COUNT; i++) {
 		if (snd_dg00x_stream_rates[i] == rate)
 			break;
@@ -196,41 +203,19 @@ static int keep_resources(struct snd_dg00x *dg00x, unsigned int rate)
 	if (i == SND_DG00X_RATE_COUNT)
 		return -EINVAL;
 
-	/* Keep resources for out-stream. */
-	err = amdtp_dot_set_parameters(&dg00x->rx_stream, rate,
-				       snd_dg00x_stream_pcm_channels[i]);
-	if (err < 0)
-		return err;
-	err = fw_iso_resources_allocate(&dg00x->rx_resources,
-				amdtp_stream_get_max_payload(&dg00x->rx_stream),
-				fw_parent_device(dg00x->unit)->max_speed);
-	if (err < 0)
-		return err;
+	if (stream == &dg00x->tx_stream)
+		resources = &dg00x->tx_resources;
+	else
+		resources = &dg00x->rx_resources;
 
-	/* Keep resources for in-stream. */
-	err = amdtp_dot_set_parameters(&dg00x->tx_stream, rate,
+	err = amdtp_dot_set_parameters(stream, rate,
 				       snd_dg00x_stream_pcm_channels[i]);
 	if (err < 0)
 		return err;
-	err = fw_iso_resources_allocate(&dg00x->tx_resources,
-				amdtp_stream_get_max_payload(&dg00x->tx_stream),
-				fw_parent_device(dg00x->unit)->max_speed);
-	if (err < 0)
-		goto error;
 
-	/* Register isochronous channels for both direction. */
-	data = cpu_to_be32((dg00x->tx_resources.channel << 16) |
-			   dg00x->rx_resources.channel);
-	err = snd_fw_transaction(dg00x->unit, TCODE_WRITE_QUADLET_REQUEST,
-				 DG00X_ADDR_BASE + DG00X_OFFSET_ISOC_CHANNELS,
-				 &data, sizeof(data), 0);
-	if (err < 0)
-		goto error;
-
-	return 0;
-error:
-	release_resources(dg00x);
-	return err;
+	return fw_iso_resources_allocate(resources,
+				amdtp_stream_get_max_payload(stream),
+				fw_parent_device(dg00x->unit)->max_speed);
 }
 
 int snd_dg00x_stream_init_duplex(struct snd_dg00x *dg00x)
@@ -272,43 +257,68 @@ void snd_dg00x_stream_destroy_duplex(struct snd_dg00x *dg00x)
 	fw_iso_resources_destroy(&dg00x->tx_resources);
 }
 
-int snd_dg00x_stream_start_duplex(struct snd_dg00x *dg00x, unsigned int rate)
+int snd_dg00x_stream_reserve_duplex(struct snd_dg00x *dg00x, unsigned int rate)
 {
 	unsigned int curr_rate;
-	int err = 0;
-
-	if (dg00x->substreams_counter == 0)
-		goto end;
+	int err;
 
-	/* Check current sampling rate. */
 	err = snd_dg00x_stream_get_local_rate(dg00x, &curr_rate);
 	if (err < 0)
-		goto error;
+		return err;
 	if (rate == 0)
 		rate = curr_rate;
-	if (curr_rate != rate ||
-	    amdtp_streaming_error(&dg00x->tx_stream) ||
-	    amdtp_streaming_error(&dg00x->rx_stream)) {
+
+	if (dg00x->substreams_counter == 0 || curr_rate != rate) {
 		finish_session(dg00x);
 
-		amdtp_stream_stop(&dg00x->tx_stream);
-		amdtp_stream_stop(&dg00x->rx_stream);
-		release_resources(dg00x);
-	}
+		fw_iso_resources_free(&dg00x->tx_resources);
+		fw_iso_resources_free(&dg00x->rx_resources);
 
-	/*
-	 * No packets are transmitted without receiving packets, reagardless of
-	 * which source of clock is used.
-	 */
-	if (!amdtp_stream_running(&dg00x->rx_stream)) {
 		err = snd_dg00x_stream_set_local_rate(dg00x, rate);
+		if (err < 0)
+			return err;
+
+		err = keep_resources(dg00x, &dg00x->rx_stream, rate);
+		if (err < 0)
+			return err;
+
+		err = keep_resources(dg00x, &dg00x->tx_stream, rate);
+		if (err < 0) {
+			fw_iso_resources_free(&dg00x->rx_resources);
+			return err;
+		}
+	}
+
+	return 0;
+}
+
+int snd_dg00x_stream_start_duplex(struct snd_dg00x *dg00x)
+{
+	unsigned int generation = dg00x->rx_resources.generation;
+	int err = 0;
+
+	if (dg00x->substreams_counter == 0)
+		return 0;
+
+	if (amdtp_streaming_error(&dg00x->tx_stream) ||
+	    amdtp_streaming_error(&dg00x->rx_stream))
+		finish_session(dg00x);
+
+	if (generation != fw_parent_device(dg00x->unit)->card->generation) {
+		err = fw_iso_resources_update(&dg00x->tx_resources);
 		if (err < 0)
 			goto error;
 
-		err = keep_resources(dg00x, rate);
+		err = fw_iso_resources_update(&dg00x->rx_resources);
 		if (err < 0)
 			goto error;
+	}
 
+	/*
+	 * No packets are transmitted without receiving packets, reagardless of
+	 * which source of clock is used.
+	 */
+	if (!amdtp_stream_running(&dg00x->rx_stream)) {
 		err = begin_session(dg00x);
 		if (err < 0)
 			goto error;
@@ -343,33 +353,22 @@ int snd_dg00x_stream_start_duplex(struct snd_dg00x *dg00x, unsigned int rate)
 			goto error;
 		}
 	}
-end:
-	return err;
+
+	return 0;
 error:
 	finish_session(dg00x);
 
-	amdtp_stream_stop(&dg00x->tx_stream);
-	amdtp_stream_stop(&dg00x->rx_stream);
-	release_resources(dg00x);
-
 	return err;
 }
 
 void snd_dg00x_stream_stop_duplex(struct snd_dg00x *dg00x)
 {
-	if (dg00x->substreams_counter > 0)
-		return;
-
-	amdtp_stream_stop(&dg00x->tx_stream);
-	amdtp_stream_stop(&dg00x->rx_stream);
-	finish_session(dg00x);
-	release_resources(dg00x);
+	if (dg00x->substreams_counter == 0) {
+		finish_session(dg00x);
 
-	/*
-	 * Just after finishing the session, the device may lost transmitting
-	 * functionality for a short time.
-	 */
-	msleep(50);
+		fw_iso_resources_free(&dg00x->tx_resources);
+		fw_iso_resources_free(&dg00x->rx_resources);
+	}
 }
 
 void snd_dg00x_stream_update_duplex(struct snd_dg00x *dg00x)

sound/firewire/digi00x/digi00x.h

@@ -139,7 +139,8 @@ int snd_dg00x_stream_get_clock(struct snd_dg00x *dg00x,
 int snd_dg00x_stream_check_external_clock(struct snd_dg00x *dg00x,
 					  bool *detect);
 int snd_dg00x_stream_init_duplex(struct snd_dg00x *dg00x);
-int snd_dg00x_stream_start_duplex(struct snd_dg00x *dg00x, unsigned int rate);
+int snd_dg00x_stream_reserve_duplex(struct snd_dg00x *dg00x, unsigned int rate);
+int snd_dg00x_stream_start_duplex(struct snd_dg00x *dg00x);
 void snd_dg00x_stream_stop_duplex(struct snd_dg00x *dg00x);
 void snd_dg00x_stream_update_duplex(struct snd_dg00x *dg00x);
 void snd_dg00x_stream_destroy_duplex(struct snd_dg00x *dg00x);

sound/firewire/fireface/ff-pcm.c

@@ -198,8 +198,8 @@ static int pcm_close(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-static int pcm_capture_hw_params(struct snd_pcm_substream *substream,
-				 struct snd_pcm_hw_params *hw_params)
+static int pcm_hw_params(struct snd_pcm_substream *substream,
+			 struct snd_pcm_hw_params *hw_params)
 {
 	struct snd_ff *ff = substream->private_data;
 	int err;
@@ -210,58 +210,26 @@ static int pcm_capture_hw_params(struct snd_pcm_substream *substream,
 		return err;
 
 	if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) {
-		mutex_lock(&ff->mutex);
-		ff->substreams_counter++;
-		mutex_unlock(&ff->mutex);
-	}
-
-	return 0;
-}
-
-static int pcm_playback_hw_params(struct snd_pcm_substream *substream,
-				  struct snd_pcm_hw_params *hw_params)
-{
-	struct snd_ff *ff = substream->private_data;
-	int err;
-
-	err = snd_pcm_lib_alloc_vmalloc_buffer(substream,
-					       params_buffer_bytes(hw_params));
-	if (err < 0)
-		return err;
+		unsigned int rate = params_rate(hw_params);
 
-	if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) {
 		mutex_lock(&ff->mutex);
-		ff->substreams_counter++;
+		err = snd_ff_stream_reserve_duplex(ff, rate);
+		if (err >= 0)
+			++ff->substreams_counter;
 		mutex_unlock(&ff->mutex);
 	}
 
 	return 0;
 }
 
-static int pcm_capture_hw_free(struct snd_pcm_substream *substream)
-{
-	struct snd_ff *ff = substream->private_data;
-
-	mutex_lock(&ff->mutex);
-
-	if (substream->runtime->status->state != SNDRV_PCM_STATE_OPEN)
-		ff->substreams_counter--;
-
-	snd_ff_stream_stop_duplex(ff);
-
-	mutex_unlock(&ff->mutex);
-
-	return snd_pcm_lib_free_vmalloc_buffer(substream);
-}
-
-static int pcm_playback_hw_free(struct snd_pcm_substream *substream)
+static int pcm_hw_free(struct snd_pcm_substream *substream)
 {
 	struct snd_ff *ff = substream->private_data;
 
 	mutex_lock(&ff->mutex);
 
 	if (substream->runtime->status->state != SNDRV_PCM_STATE_OPEN)
-		ff->substreams_counter--;
+		--ff->substreams_counter;
 
 	snd_ff_stream_stop_duplex(ff);
 
@@ -374,8 +342,8 @@ int snd_ff_create_pcm_devices(struct snd_ff *ff)
 		.open		= pcm_open,
 		.close		= pcm_close,
 		.ioctl		= snd_pcm_lib_ioctl,
-		.hw_params	= pcm_capture_hw_params,
-		.hw_free	= pcm_capture_hw_free,
+		.hw_params	= pcm_hw_params,
+		.hw_free	= pcm_hw_free,
 		.prepare	= pcm_capture_prepare,
 		.trigger	= pcm_capture_trigger,
 		.pointer	= pcm_capture_pointer,
@@ -386,8 +354,8 @@ int snd_ff_create_pcm_devices(struct snd_ff *ff)
 		.open		= pcm_open,
 		.close		= pcm_close,
 		.ioctl		= snd_pcm_lib_ioctl,
-		.hw_params	= pcm_playback_hw_params,
-		.hw_free	= pcm_playback_hw_free,
+		.hw_params	= pcm_hw_params,
+		.hw_free	= pcm_hw_free,
 		.prepare	= pcm_playback_prepare,
 		.trigger	= pcm_playback_trigger,
 		.pointer	= pcm_playback_pointer,

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

@@ -97,25 +97,64 @@ static int latter_switch_fetching_mode(struct snd_ff *ff, bool enable)
 				  LATTER_FETCH_MODE, &reg, sizeof(reg), 0);
 }
 
-static int keep_resources(struct snd_ff *ff, unsigned int rate)
+static int latter_allocate_resources(struct snd_ff *ff, unsigned int rate)
 {
 	enum snd_ff_stream_mode mode;
+	unsigned int code;
+	__le32 reg;
+	unsigned int count;
 	int i;
 	int err;
 
-	// Check whether the given value is supported or not.
-	for (i = 0; i < CIP_SFC_COUNT; i++) {
-		if (amdtp_rate_table[i] == rate)
+	// Set the number of data blocks transferred in a second.
+	if (rate % 32000 == 0)
+		code = 0x00;
+	else if (rate % 44100 == 0)
+		code = 0x02;
+	else if (rate % 48000 == 0)
+		code = 0x04;
+	else
+		return -EINVAL;
+
+	if (rate >= 64000 && rate < 128000)
+		code |= 0x08;
+	else if (rate >= 128000 && rate < 192000)
+		code |= 0x10;
+
+	reg = cpu_to_le32(code);
+	err = snd_fw_transaction(ff->unit, TCODE_WRITE_QUADLET_REQUEST,
+				 LATTER_STF, &reg, sizeof(reg), 0);
+	if (err < 0)
+		return err;
+
+	// Confirm to shift transmission clock.
+	count = 0;
+	while (count++ < 10) {
+		unsigned int curr_rate;
+		enum snd_ff_clock_src src;
+
+		err = latter_get_clock(ff, &curr_rate, &src);
+		if (err < 0)
+			return err;
+
+		if (curr_rate == rate)
 			break;
 	}
-	if (i >= CIP_SFC_COUNT)
+	if (count == 10)
+		return -ETIMEDOUT;
+
+	for (i = 0; i < ARRAY_SIZE(amdtp_rate_table); ++i) {
+		if (rate == amdtp_rate_table[i])
+			break;
+	}
+	if (i == ARRAY_SIZE(amdtp_rate_table))
 		return -EINVAL;
 
 	err = snd_ff_stream_get_multiplier_mode(i, &mode);
 	if (err < 0)
 		return err;
 
-	/* Keep resources for in-stream. */
+	// Keep resources for in-stream.
 	ff->tx_resources.channels_mask = 0x00000000000000ffuLL;
 	err = fw_iso_resources_allocate(&ff->tx_resources,
 			amdtp_stream_get_max_payload(&ff->tx_stream),
@@ -123,7 +162,7 @@ static int keep_resources(struct snd_ff *ff, unsigned int rate)
 	if (err < 0)
 		return err;
 
-	/* Keep resources for out-stream. */
+	// Keep resources for out-stream.
 	ff->rx_resources.channels_mask = 0x00000000000000ffuLL;
 	err = fw_iso_resources_allocate(&ff->rx_resources,
 			amdtp_stream_get_max_payload(&ff->rx_stream),
@@ -136,60 +175,30 @@ static int keep_resources(struct snd_ff *ff, unsigned int rate)
 
 static int latter_begin_session(struct snd_ff *ff, unsigned int rate)
 {
-	static const struct {
-		unsigned int stf;
-		unsigned int code;
-		unsigned int flag;
-	} *entry, rate_table[] = {
-		{ 32000,  0x00, 0x92, },
-		{ 44100,  0x02, 0x92, },
-		{ 48000,  0x04, 0x92, },
-		{ 64000,  0x08, 0x8e, },
-		{ 88200,  0x0a, 0x8e, },
-		{ 96000,  0x0c, 0x8e, },
-		{ 128000, 0x10, 0x8c, },
-		{ 176400, 0x12, 0x8c, },
-		{ 192000, 0x14, 0x8c, },
-	};
+	unsigned int generation = ff->rx_resources.generation;
+	unsigned int flag;
 	u32 data;
 	__le32 reg;
-	unsigned int count;
-	int i;
 	int err;
 
-	for (i = 0; i < ARRAY_SIZE(rate_table); ++i) {
-		entry = rate_table + i;
-		if (entry->stf == rate)
-			break;
-	}
-	if (i == ARRAY_SIZE(rate_table))
+	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;
 
-	reg = cpu_to_le32(entry->code);
-	err = snd_fw_transaction(ff->unit, TCODE_WRITE_QUADLET_REQUEST,
-				 LATTER_STF, &reg, sizeof(reg), 0);
-	if (err < 0)
-		return err;
-
-	// Confirm to shift transmission clock.
-	count = 0;
-	while (count++ < 10) {
-		unsigned int curr_rate;
-		enum snd_ff_clock_src src;
-
-		err = latter_get_clock(ff, &curr_rate, &src);
+	if (generation != fw_parent_device(ff->unit)->card->generation) {
+		err = fw_iso_resources_update(&ff->tx_resources);
 		if (err < 0)
 			return err;
 
-		if (curr_rate == rate)
-			break;
+		err = fw_iso_resources_update(&ff->rx_resources);
+		if (err < 0)
+			return err;
 	}
-	if (count == 10)
-		return -ETIMEDOUT;
-
-	err = keep_resources(ff, rate);
-	if (err < 0)
-		return err;
 
 	data = (ff->tx_resources.channel << 8) | ff->rx_resources.channel;
 	reg = cpu_to_le32(data);
@@ -200,7 +209,7 @@ static int latter_begin_session(struct snd_ff *ff, unsigned int rate)
 
 	// Always use the maximum number of data channels in data block of
 	// packet.
-	reg = cpu_to_le32(entry->flag);
+	reg = cpu_to_le32(flag);
 	return snd_fw_transaction(ff->unit, TCODE_WRITE_QUADLET_REQUEST,
 				  LATTER_ISOC_START, &reg, sizeof(reg), 0);
 }
@@ -424,6 +433,7 @@ const struct snd_ff_protocol snd_ff_protocol_latter = {
 	.fill_midi_msg		= latter_fill_midi_msg,
 	.get_clock		= latter_get_clock,
 	.switch_fetching_mode	= latter_switch_fetching_mode,
+	.allocate_resources	= latter_allocate_resources,
 	.begin_session		= latter_begin_session,
 	.finish_session		= latter_finish_session,
 	.dump_status		= latter_dump_status,

sound/firewire/tascam/amdtp-tascam.c

@@ -223,7 +223,7 @@ int amdtp_tscm_init(struct amdtp_stream *s, struct fw_unit *unit,
 		return 0;
 
 	/* Use fixed value for FDF field. */
-	s->fdf = 0x00;
+	s->ctx_data.rx.fdf = 0x00;
 
 	/* This protocol uses fixed number of data channels for PCM samples. */
 	p = s->protocol;