Merge tag 'sound-4.19-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "It's been busy summer weeks and hence lots of changes, partly for a
  few new drivers and partly for a wide range of fixes.

  Here are highlights:

  ALSA Core:
   - Fix rawmidi buffer management, code cleanup / refactoring
   - Fix the SG-buffer page handling with incorrect fallback size
   - Fix the stall at virmidi trigger callback with a large buffer; also
     offloading and code-refactoring along with it
   - Various ALSA sequencer code cleanups

  ASoC:
   - Deploy the standard snd_pcm_stop_xrun() helper in several drivers
   - Support for providing name prefixes to generic component nodes
   - Quite a few fixes for DPCM as it gains a bit wider use and more
     robust testing
   - Generalization of the DIO2125 support to a simple amplifier driver
   - Accessory detection support for the audio graph card
   - DT support for PXA AC'97 devices
   - Quirks for a number of new x86 systems
   - Support for AM Logic Meson, Everest ES7154, Intel systems with
     RT5682, Qualcomm QDSP6 and WCD9335, Realtek RT5682 and TI TAS5707

  HD-audio:
   - Code refactoring in HD-audio ext codec codes to drop own classes;
     preliminary works for the upcoming legacy codec support
   - Generalized DRM audio component for the upcoming radeon / amdgpu
     support
   - Unification of mic mute-LED and GPIO support for various codecs
   - Further improvement of CA0132 codec support including Recon3D
   - Proper vga_switcheroo handling for AMD i-GPU
   - Update of model list in documentation
   - Fixups for another HP Spectre x360, Conexant codecs, power-save
     blacklist update

  USB-audio:
   - Fix the invalid sample rate setup with external clock
   - Support of UAC3 selector units and processing units
   - Basic UAC3 power-domain support
   - Support for Encore mDSD and Thesycon-based DSD devices
   - Preparation for future complete callback changes

  Firewire:
   - Add support for MOTU Traveler

  Misc:
   - The endianess notation fixes in various drivers
   - Add fall-through comment in lots of drivers
   - Various sparse warning fixes, e.g. about PCM format types"

* tag 'sound-4.19-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (529 commits)
  ASoC: adav80x: mark expected switch fall-through
  ASoC: da7219: Add delays to capture path to remove DC offset noise
  ALSA: usb-audio: Mark expected switch fall-through
  ALSA: mixart: Mark expected switch fall-through
  ALSA: opl3: Mark expected switch fall-through
  ALSA: hda/ca0132 - Add exit commands for Recon3D
  ALSA: hda/ca0132 - Change mixer controls for Recon3D
  ALSA: hda/ca0132 - Add Recon3D input and output select commands
  ALSA: hda/ca0132 - Add DSP setup defaults for Recon3D
  ALSA: hda/ca0132 - Add Recon3D startup functions and setup
  ALSA: hda/ca0132 - Add bool variable to enable/disable pci region2 mmio
  ALSA: hda/ca0132 - Add Recon3D pincfg
  ALSA: hda/ca0132 - Add quirk ID and enum for Recon3D
  ALSA: hda/ca0132 - Add alt_functions unsolicited response
  ALSA: hda/ca0132 - Clean up ca0132_init function.
  ALSA: hda/ca0132 - Create mmio gpio function to make code clearer
  ASoC: wm_adsp: Make DSP name configurable by codec driver
  ASoC: wm_adsp: Declare firmware controls from codec driver
  ASoC: max98373: Added software reset register to readable registers
  ASoC: wm_adsp: Correct DSP pointer for preloader control
  ...

Documentation/devicetree/bindings/sound/ac97-bus.txt

@@ -0,0 +1,32 @@
+Generic AC97 Device Properties
+
+This documents describes the devicetree bindings for an ac97 controller child
+node describing ac97 codecs.
+
+Required properties:
+-compatible : Must be "ac97,vendor_id1,vendor_id2
+	      The ids shall be the 4 characters hexadecimal encoding, such as
+	      given by "%04x" formatting of printf
+-reg	    : Must be the ac97 codec number, between 0 and 3
+
+Example:
+ac97: sound@40500000 {
+	compatible = "marvell,pxa270-ac97";
+	reg = < 0x40500000 0x1000 >;
+	interrupts = <14>;
+	reset-gpios = <&gpio 95 GPIO_ACTIVE_HIGH>;
+	#sound-dai-cells = <1>;
+	pinctrl-names = "default";
+	pinctrl-0 = < &pinctrl_ac97_default >;
+	clocks = <&clks CLK_AC97>, <&clks CLK_AC97CONF>;
+	clock-names = "AC97CLK", "AC97CONFCLK";
+
+	#address-cells = <1>;
+	#size-cells = <0>;
+	audio-codec@0 {
+		reg = <0>;
+		compatible = "ac97,574d,4c13";
+		clocks = <&fixed_wm9713_clock>;
+		clock-names = "ac97_clk";
+	}
+};

Documentation/devicetree/bindings/sound/amlogic,axg-fifo.txt

@@ -0,0 +1,23 @@
+* Amlogic Audio FIFO controllers
+
+Required properties:
+- compatible: 'amlogic,axg-toddr' or
+	      'amlogic,axg-frddr'
+- reg: physical base address of the controller and length of memory
+       mapped region.
+- interrupts: interrupt specifier for the fifo.
+- clocks: phandle to the fifo peripheral clock provided by the audio
+	  clock controller.
+- resets: phandle to memory ARB line provided by the arb reset controller.
+- #sound-dai-cells: must be 0.
+
+Example of FRDDR A on the A113 SoC:
+
+frddr_a: audio-controller@1c0 {
+	compatible = "amlogic,axg-frddr";
+	reg = <0x0 0x1c0 0x0 0x1c>;
+	#sound-dai-cells = <0>;
+	interrupts = <GIC_SPI 88 IRQ_TYPE_EDGE_RISING>;
+	clocks = <&clkc_audio AUD_CLKID_FRDDR_A>;
+	resets = <&arb AXG_ARB_FRDDR_A>;
+};

Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.txt

@@ -0,0 +1,124 @@
+Amlogic AXG sound card:
+
+Required properties:
+
+- compatible: "amlogic,axg-sound-card"
+- model : User specified audio sound card name, one string
+
+Optional properties:
+
+- audio-aux-devs : List of phandles pointing to auxiliary devices
+- audio-widgets : Please refer to widgets.txt.
+- audio-routing : A list of the connections between audio components.
+
+Subnodes:
+
+- dai-link: Container for dai-link level properties and the CODEC
+	    sub-nodes. There should be at least one (and probably more)
+	    subnode of this type.
+
+Required dai-link properties:
+
+- sound-dai: phandle and port of the CPU DAI.
+
+Required TDM Backend dai-link properties:
+- dai-format : CPU/CODEC common audio format
+
+Optional TDM Backend dai-link properties:
+- dai-tdm-slot-rx-mask-{0,1,2,3}: Receive direction slot masks
+- dai-tdm-slot-tx-mask-{0,1,2,3}: Transmit direction slot masks
+				  When omitted, mask is assumed to have to no
+				  slots. A valid must have at one slot, so at
+				  least one these mask should be provided with
+				  an enabled slot.
+- dai-tdm-slot-num : Please refer to tdm-slot.txt.
+		     If omitted, slot number is set to accommodate the largest
+		     mask provided.
+- dai-tdm-slot-width : Please refer to tdm-slot.txt. default to 32 if omitted.
+- mclk-fs : Multiplication factor between stream rate and mclk
+
+Backend dai-link subnodes:
+
+- codec: dai-link representing backend links should have at least one subnode.
+	 One subnode for each codec of the dai-link.
+	 dai-link representing frontend links have no codec, therefore have no
+	 subnodes
+
+Required codec subnodes properties:
+
+- sound-dai: phandle and port of the CODEC DAI.
+
+Optional codec subnodes properties:
+
+- dai-tdm-slot-tx-mask : Please refer to tdm-slot.txt.
+- dai-tdm-slot-rx-mask : Please refer to tdm-slot.txt.
+
+Example:
+
+sound {
+	compatible = "amlogic,axg-sound-card";
+	model = "AXG-S420";
+	audio-aux-devs = <&tdmin_a>, <&tdmout_c>;
+	audio-widgets = "Line", "Lineout",
+			"Line", "Linein",
+			"Speaker", "Speaker1 Left",
+			"Speaker", "Speaker1 Right";
+			"Speaker", "Speaker2 Left",
+			"Speaker", "Speaker2 Right";
+	audio-routing = "TDMOUT_C IN 0", "FRDDR_A OUT 2",
+			"SPDIFOUT IN 0", "FRDDR_A OUT 3",
+			"TDM_C Playback", "TDMOUT_C OUT",
+			"TDMIN_A IN 2", "TDM_C Capture",
+			"TDMIN_A IN 5", "TDM_C Loopback",
+			"TODDR_A IN 0", "TDMIN_A OUT",
+			"Lineout", "Lineout AOUTL",
+			"Lineout", "Lineout AOUTR",
+			"Speaker1 Left", "SPK1 OUT_A",
+			"Speaker2 Left", "SPK2 OUT_A",
+			"Speaker1 Right", "SPK1 OUT_B",
+			"Speaker2 Right", "SPK2 OUT_B",
+			"Linein AINL", "Linein",
+			"Linein AINR", "Linein";
+
+	dai-link@0 {
+		sound-dai = <&frddr_a>;
+	};
+
+	dai-link@1 {
+		sound-dai = <&toddr_a>;
+	};
+
+	dai-link@2 {
+		sound-dai = <&tdmif_c>;
+		dai-format = "i2s";
+		dai-tdm-slot-tx-mask-2 = <1 1>;
+		dai-tdm-slot-tx-mask-3 = <1 1>;
+		dai-tdm-slot-rx-mask-1 = <1 1>;
+		mclk-fs = <256>;
+
+		codec@0 {
+			sound-dai = <&lineout>;
+		};
+
+		codec@1 {
+			sound-dai = <&speaker_amp1>;
+		};
+
+		codec@2 {
+			sound-dai = <&speaker_amp2>;
+		};
+
+		codec@3 {
+			sound-dai = <&linein>;
+		};
+
+	};
+
+	dai-link@3 {
+		sound-dai = <&spdifout>;
+
+		codec {
+			sound-dai = <&spdif_dit>;
+		};
+	};
+};

Documentation/devicetree/bindings/sound/amlogic,axg-spdifout.txt

@@ -0,0 +1,20 @@
+* Amlogic Audio SPDIF Output
+
+Required properties:
+- compatible: 'amlogic,axg-spdifout'
+- clocks: list of clock phandle, one for each entry clock-names.
+- clock-names: should contain the following:
+  * "pclk" : peripheral clock.
+  * "mclk" : master clock
+- #sound-dai-cells: must be 0.
+
+Example on the A113 SoC:
+
+spdifout: audio-controller@480 {
+	compatible = "amlogic,axg-spdifout";
+	reg = <0x0 0x480 0x0 0x50>;
+	#sound-dai-cells = <0>;
+	clocks = <&clkc_audio AUD_CLKID_SPDIFOUT>,
+		 <&clkc_audio AUD_CLKID_SPDIFOUT_CLK>;
+	clock-names = "pclk", "mclk";
+};

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

@@ -0,0 +1,28 @@
+* Amlogic Audio TDM formatters
+
+Required properties:
+- compatible: 'amlogic,axg-tdmin' or
+	      'amlogic,axg-tdmout'
+- reg: physical base address of the controller and length of memory
+       mapped region.
+- clocks: list of clock phandle, one for each entry clock-names.
+- clock-names: should contain the following:
+  * "pclk"     : peripheral clock.
+  * "sclk"     : bit clock.
+  * "sclk_sel" : bit clock input multiplexer.
+  * "lrclk"    : sample clock
+  * "lrclk_sel": sample clock input multiplexer
+
+Example of TDMOUT_A on the A113 SoC:
+
+tdmout_a: audio-controller@500 {
+	compatible = "amlogic,axg-tdmout";
+	reg = <0x0 0x500 0x0 0x40>;
+	clocks = <&clkc_audio AUD_CLKID_TDMOUT_A>,
+		 <&clkc_audio AUD_CLKID_TDMOUT_A_SCLK>,
+		 <&clkc_audio AUD_CLKID_TDMOUT_A_SCLK_SEL>,
+		 <&clkc_audio AUD_CLKID_TDMOUT_A_LRCLK>,
+		 <&clkc_audio AUD_CLKID_TDMOUT_A_LRCLK>;
+	clock-names = "pclk", "sclk", "sclk_sel",
+		      "lrclk", "lrclk_sel";
+};

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

@@ -0,0 +1,22 @@
+* Amlogic Audio TDM Interfaces
+
+Required properties:
+- compatible: 'amlogic,axg-tdm-iface'
+- clocks: list of clock phandle, one for each entry clock-names.
+- clock-names: should contain the following:
+  * "sclk" : bit clock.
+  * "lrclk": sample clock
+  * "mclk" : master clock
+	     -> optional if the interface is in clock slave mode.
+- #sound-dai-cells: must be 0.
+
+Example of TDM_A on the A113 SoC:
+
+tdmif_a: audio-controller@0 {
+	compatible = "amlogic,axg-tdm-iface";
+	#sound-dai-cells = <0>;
+	clocks = <&clkc_audio AUD_CLKID_MST_A_MCLK>,
+		 <&clkc_audio AUD_CLKID_MST_A_SCLK>,
+		 <&clkc_audio AUD_CLKID_MST_A_LRCLK>;
+	clock-names = "mclk", "sclk", "lrclk";
+};

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

@@ -15,7 +15,6 @@ Required properties:
 - clock-names:    Should be one of each entry matching the clocks phandles list:
                   - "pclk" (peripheral clock) Required.
                   - "gclk" (generated clock) Optional (1).
-                  - "aclk" (Audio PLL clock) Optional (1).
                   - "muxclk" (I2S mux clock) Optional (1).
 
 Optional properties:
@@ -23,9 +22,9 @@ Optional properties:
 - princtrl-names: Should contain only one value - "default".
 
 
-(1) : Only the peripheral clock is required. The generated clock, the Audio
-      PLL clock adn the I2S mux clock are optional and should only be set
-      together, when Master Mode is required.
+(1) : Only the peripheral clock is required. The generated clock and the I2S
+      mux clock are optional and should only be set together, when Master Mode
+      is required.
 
 Example:
 
@@ -40,8 +39,8 @@ Example:
 			(AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1) |
 			 AT91_XDMAC_DT_PERID(32))>;
 		dma-names = "tx", "rx";
-		clocks = <&i2s0_clk>, <&i2s0_gclk>, <&audio_pll_pmc>, <&i2s0muxck>;
-		clock-names = "pclk", "gclk", "aclk", "muxclk";
+		clocks = <&i2s0_clk>, <&i2s0_gclk>, <&i2s0muxck>;
+		clock-names = "pclk", "gclk", "muxclk";
 		pinctrl-names = "default";
 		pinctrl-0 = <&pinctrl_i2s0_default>;
 	};

Documentation/devicetree/bindings/sound/audio-graph-card.txt

@@ -18,6 +18,8 @@ Below are same as Simple-Card.
 - bitclock-inversion
 - frame-inversion
 - mclk-fs
+- hp-det-gpio
+- mic-det-gpio
 - dai-tdm-slot-num
 - dai-tdm-slot-width
 - clocks / system-clock-frequency

Documentation/devicetree/bindings/sound/dioo,dio2125.txt

@@ -1,12 +0,0 @@
-DIO2125 Audio Driver
-
-Required properties:
-- compatible : "dioo,dio2125"
-- enable-gpios : the gpio connected to the enable pin of the dio2125
-
-Example:
-
-amp: analog-amplifier {
-	compatible = "dioo,dio2125";
-	enable-gpios = <&gpio GPIOH_3 0>;
-};

Documentation/devicetree/bindings/sound/everest,es7134.txt

@@ -1,10 +1,15 @@
 ES7134 i2s DA converter
 
 Required properties:
-- compatible : "everest,es7134" or "everest,es7144"
+- compatible : "everest,es7134" or
+               "everest,es7144" or
+	       "everest,es7154"
+- VDD-supply : regulator phandle for the VDD supply
+- PVDD-supply: regulator phandle for the PVDD supply for the es7154
 
 Example:
 
 i2s_codec: external-codec {
 	compatible = "everest,es7134";
+	VDD-supply = <&vcc_5v>;
 };

Documentation/devicetree/bindings/sound/everest,es7241.txt

@@ -0,0 +1,28 @@
+ES7241 i2s AD converter
+
+Required properties:
+- compatible : "everest,es7241"
+- VDDP-supply: regulator phandle for the VDDA supply
+- VDDA-supply: regulator phandle for the VDDP supply
+- VDDD-supply: regulator phandle for the VDDD supply
+
+Optional properties:
+- reset-gpios: gpio connected to the reset pin
+- m0-gpios   : gpio connected to the m0 pin
+- m1-gpios   : gpio connected to the m1 pin
+- everest,sdout-pull-down:
+   Format used by the serial interface is controlled by pulling
+   the sdout. If the sdout is pulled down, leftj format is used.
+   If this property is not provided, sdout is assumed to pulled
+   up and i2s format is used
+
+Example:
+
+linein: audio-codec@2 {
+	#sound-dai-cells = <0>;
+	compatible = "everest,es7241";
+	VDDA-supply = <&vcc_3v3>;
+	VDDP-supply = <&vcc_3v3>;
+	VDDD-supply = <&vcc_3v3>;
+	reset-gpios = <&gpio GPIOH_42>;
+};

Documentation/devicetree/bindings/sound/marvell,pxa2xx-ac97.txt

@@ -0,0 +1,27 @@
+Marvell PXA2xx audio complex
+
+This descriptions matches the AC97 controller found in pxa2xx and pxa3xx series.
+
+Required properties:
+  - compatible: should be one of the following:
+    "marvell,pxa250-ac97"
+    "marvell,pxa270-ac97"
+    "marvell,pxa300-ac97"
+  - reg: device MMIO address space
+  - interrupts: single interrupt generated by AC97 IP
+  - clocks: input clock of the AC97 IP, refer to clock-bindings.txt
+
+Optional properties:
+  - pinctrl-names, pinctrl-0: refer to pinctrl-bindings.txt
+  - reset-gpios: gpio used for AC97 reset, refer to gpio.txt
+
+Example:
+	ac97: sound@40500000 {
+		compatible = "marvell,pxa250-ac97";
+		reg = < 0x40500000 0x1000 >;
+		interrupts = <14>;
+		reset-gpios = <&gpio 113 GPIO_ACTIVE_HIGH>;
+		#sound-dai-cells = <1>;
+		pinctrl-names = "default";
+		pinctrl-0 = < &pmux_ac97_default >;
+	};

Documentation/devicetree/bindings/sound/mrvl,pxa-ssp.txt

@@ -5,6 +5,14 @@ Required properties:
 	compatible	Must be "mrvl,pxa-ssp-dai"
 	port		A phandle reference to a PXA ssp upstream device
 
+Optional properties:
+
+	clock-names
+	clocks		Through "clock-names" and "clocks", external clocks
+			can be configured. If a clock names "extclk" exists,
+			it will be set to the mclk rate of the audio stream
+			and be used as clock provider of the DAI.
+
 Example:
 
 	/* upstream device */

Documentation/devicetree/bindings/sound/mrvl,pxa2xx-pcm.txt

@@ -1,15 +0,0 @@
-DT bindings for ARM PXA2xx PCM platform driver
-
-This is just a dummy driver that registers the PXA ASoC platform driver.
-It does not have any resources assigned.
-
-Required properties:
-
-	- compatible		'mrvl,pxa-pcm-audio'
-
-Example:
-
-	pxa_pcm_audio: snd_soc_pxa_audio {
-		compatible = "mrvl,pxa-pcm-audio";
-	};
-

Documentation/devicetree/bindings/sound/name-prefix.txt

@@ -0,0 +1,24 @@
+Name prefix:
+
+Card implementing the routing property define the connection between
+audio components as list of string pair. Component using the same
+sink/source names may use the name prefix property to prepend the
+name of their sinks/sources with the provided string.
+
+Optional name prefix property:
+- sound-name-prefix : string using as prefix for the sink/source names of
+		      the component.
+
+Example: Two instances of the same component.
+
+amp0: analog-amplifier@0 {
+	compatible = "simple-audio-amplifier";
+	enable-gpios = <&gpio GPIOH_3 0>;
+	sound-name-prefix = "FRONT";
+};
+
+amp1: analog-amplifier@1 {
+	compatible = "simple-audio-amplifier";
+	enable-gpios = <&gpio GPIOH_4 0>;
+	sound-name-prefix = "BACK";
+};

Documentation/devicetree/bindings/sound/qcom,apq8096.txt

@@ -7,7 +7,7 @@ This binding describes the APQ8096 sound card, which uses qdsp for audio.
 	Value type: <stringlist>
 	Definition: must be "qcom,apq8096-sndcard"
 
-- qcom,audio-routing:
+- audio-routing:
 	Usage: Optional
 	Value type: <stringlist>
 	Definition:  A list of the connections between audio components.
@@ -49,6 +49,12 @@ This binding describes the APQ8096 sound card, which uses qdsp for audio.
 			"DMIC1"
 			"DMIC2"
 			"DMIC3"
+
+- model:
+	Usage: required
+	Value type: <stringlist>
+	Definition: The user-visible name of this sound card.
+
 = dailinks
 Each subnode of sndcard represents either a dailink, and subnodes of each
 dailinks would be cpu/codec/platform dais.
@@ -79,11 +85,16 @@ dailinks would be cpu/codec/platform dais.
 	Value type: <phandle with arguments>
 	Definition: dai phandle/s and port of CPU/CODEC/PLATFORM node.
 
+Obsolete:
+	qcom,model: String for soundcard name (Use model instead)
+	qcom,audio-routing: A list of the connections between audio components.
+			    (Use audio-routing instead)
+
 Example:
 
 audio {
 	compatible = "qcom,apq8096-sndcard";
-	qcom,model = "DB820c";
+	model = "DB820c";
 
 	mm1-dai-link {
 		link-name = "MultiMedia1";

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

@@ -18,6 +18,11 @@ used by the apr service device.
 = ADM routing
 "routing" subnode of the ADM node represents adm routing specific configuration
 
+- compatible:
+	Usage: required
+	Value type: <stringlist>
+	Definition: must be "qcom,q6adm-routing".
+
 - #sound-dai-cells
 	Usage: required
 	Value type: <u32>
@@ -28,6 +33,7 @@ q6adm@8 {
 	compatible = "qcom,q6adm";
 	reg = <APR_SVC_ADM>;
 	q6routing: routing {
+		compatible = "qcom,q6adm-routing";
 		#sound-dai-cells = <0>;
 	};
 };

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

@@ -17,6 +17,11 @@ used by all apr services. Must contain the following properties.
 subnode of "dais" representing board specific dai setup.
 "dais" node should have following properties followed by dai children.
 
+- compatible:
+	Usage: required
+	Value type: <stringlist>
+	Definition: must be "qcom,q6afe-dais"
+
 - #sound-dai-cells
 	Usage: required
 	Value type: <u32>
@@ -100,6 +105,7 @@ q6afe@4 {
 	reg = <APR_SVC_AFE>;
 
 	dais {
+		compatible = "qcom,q6afe-dais";
 		#sound-dai-cells = <1>;
 		#address-cells = <1>;
 		#size-cells = <0>;

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

@@ -17,6 +17,11 @@ used by the apr service device.
 = ASM DAIs (Digial Audio Interface)
 "dais" subnode of the ASM node represents dai specific configuration
 
+- compatible:
+	Usage: required
+	Value type: <stringlist>
+	Definition: must be "qcom,q6asm-dais".
+
 - #sound-dai-cells
 	Usage: required
 	Value type: <u32>
@@ -28,6 +33,7 @@ q6asm@7 {
 	compatible = "qcom,q6asm";
 	reg = <APR_SVC_ASM>;
 	q6asmdai: dais {
+		compatible = "qcom,q6asm-dais";
 		#sound-dai-cells = <1>;
 	};
 };

Documentation/devicetree/bindings/sound/qcom,sdm845.txt

@@ -0,0 +1,80 @@
+* Qualcomm Technologies Inc. SDM845 ASoC sound card driver
+
+This binding describes the SDM845 sound card, which uses qdsp for audio.
+
+- compatible:
+	Usage: required
+	Value type: <stringlist>
+	Definition: must be "qcom,sdm845-sndcard"
+
+- audio-routing:
+	Usage: Optional
+	Value type: <stringlist>
+	Definition:  A list of the connections between audio components.
+		  Each entry is a pair of strings, the first being the
+		  connection's sink, the second being the connection's
+		  source. Valid names could be power supplies, MicBias
+		  of codec and the jacks on the board.
+
+- model:
+	Usage: required
+	Value type: <stringlist>
+	Definition: The user-visible name of this sound card.
+
+= dailinks
+Each subnode of sndcard represents either a dailink, and subnodes of each
+dailinks would be cpu/codec/platform dais.
+
+- link-name:
+	Usage: required
+	Value type: <string>
+	Definition: User friendly name for dai link
+
+= CPU, PLATFORM, CODEC dais subnodes
+- cpu:
+	Usage: required
+	Value type: <subnode>
+	Definition: cpu dai sub-node
+
+- codec:
+	Usage: required
+	Value type: <subnode>
+	Definition: codec dai sub-node
+
+- platform:
+	Usage: Optional
+	Value type: <subnode>
+	Definition: platform dai sub-node
+
+- sound-dai:
+	Usage: required
+	Value type: <phandle>
+	Definition: dai phandle/s and port of CPU/CODEC/PLATFORM node.
+
+Example:
+
+audio {
+	compatible = "qcom,sdm845-sndcard";
+	model = "sdm845-snd-card";
+	pinctrl-names = "default", "sleep";
+	pinctrl-0 = <&pri_mi2s_active &pri_mi2s_ws_active>;
+	pinctrl-1 = <&pri_mi2s_sleep &pri_mi2s_ws_sleep>;
+
+	mm1-dai-link {
+		link-name = "MultiMedia1";
+		cpu {
+			sound-dai = <&q6asmdai MSM_FRONTEND_DAI_MULTIMEDIA1>;
+		};
+	};
+
+	pri-mi2s-dai-link {
+		link-name = "PRI MI2S Playback";
+		cpu {
+			sound-dai = <&q6afedai PRIMARY_MI2S_RX>;
+		};
+
+		platform {
+			sound-dai = <&q6routing>;
+		};
+	};
+};

Documentation/devicetree/bindings/sound/qcom,wcd9335.txt

@@ -0,0 +1,123 @@
+QCOM WCD9335 Codec
+
+Qualcomm WCD9335 Codec is a standalone Hi-Fi audio codec IC, supports
+Qualcomm Technologies, Inc. (QTI) multimedia solutions, including
+the MSM8996, MSM8976, and MSM8956 chipsets. It has in-built
+Soundwire controller, interrupt mux. It supports both I2S/I2C and
+SLIMbus audio interfaces.
+
+Required properties with SLIMbus Interface:
+
+- compatible:
+	Usage: required
+	Value type: <stringlist>
+	Definition: For SLIMbus interface it should be "slimMID,PID",
+		    textual representation of Manufacturer ID, Product Code,
+		    shall be in lower case hexadecimal with leading zeroes
+		    suppressed.  Refer to slimbus/bus.txt for details.
+		    Should be:
+		    "slim217,1a0" for MSM8996 and APQ8096 SoCs with SLIMbus.
+
+- reg
+	Usage: required
+	Value type: <u32 u32>
+	Definition: Should be ('Device index', 'Instance ID')
+
+- interrupts
+	Usage: required
+	Value type: <prop-encoded-array>
+	Definition: Interrupts via WCD INTR1 and INTR2 pins
+
+- interrupt-names:
+	Usage: required
+	Value type: <String array>
+	Definition: Interrupt names of WCD INTR1 and INTR2
+	Should be: "intr1", "intr2"
+
+- reset-gpio:
+	Usage: required
+	Value type: <String Array>
+	Definition: Reset gpio line
+
+- qcom,ifd:
+	Usage: required
+	Value type: <phandle>
+	Definition: SLIM interface device
+
+- clocks:
+	Usage: required
+	Value type: <prop-encoded-array>
+	Definition: See clock-bindings.txt section "consumers". List of
+                three clock specifiers for mclk, mclk2 and slimbus clock.
+
+- clock-names:
+	Usage: required
+	Value type: <string>
+	Definition: Must contain "mclk", "mclk2" and "slimbus" strings.
+
+- vdd-buck-supply:
+	Usage: required
+	Value type: <phandle>
+	Definition: Should contain a reference to the 1.8V buck supply
+
+- vdd-buck-sido-supply:
+	Usage: required
+	Value type: <phandle>
+	Definition: Should contain a reference to the 1.8V SIDO buck supply
+
+- vdd-rx-supply:
+	Usage: required
+	Value type: <phandle>
+	Definition: Should contain a reference to the 1.8V rx supply
+
+- vdd-tx-supply:
+	Usage: required
+	Value type: <phandle>
+	Definition: Should contain a reference to the 1.8V tx supply
+
+- vdd-vbat-supply:
+	Usage: Optional
+	Value type: <phandle>
+	Definition: Should contain a reference to the vbat supply
+
+- vdd-micbias-supply:
+	Usage: required
+	Value type: <phandle>
+	Definition: Should contain a reference to the micbias supply
+
+- vdd-io-supply:
+	Usage: required
+	Value type: <phandle>
+	Definition: Should contain a reference to the 1.8V io supply
+
+- interrupt-controller:
+	Usage: required
+	Definition: Indicating that this is a interrupt controller
+
+- #interrupt-cells:
+	Usage: required
+	Value type: <int>
+	Definition: should be 1
+
+#sound-dai-cells
+	Usage: required
+	Value type: <u32>
+	Definition: Must be 1
+
+codec@1{
+	compatible = "slim217,1a0";
+	reg  = <1 0>;
+	interrupts = <&msmgpio 54 IRQ_TYPE_LEVEL_HIGH>;
+	interrupt-names = "intr2"
+	reset-gpio = <&msmgpio 64 0>;
+	qcom,ifd  = <&wc9335_ifd>;
+	clock-names = "mclk", "native";
+	clocks = <&rpmcc RPM_SMD_DIV_CLK1>,
+		 <&rpmcc RPM_SMD_BB_CLK1>;
+	vdd-buck-supply = <&pm8994_s4>;
+	vdd-rx-supply = <&pm8994_s4>;
+	vdd-buck-sido-supply = <&pm8994_s4>;
+	vdd-tx-supply = <&pm8994_s4>;
+	vdd-io-supply = <&pm8994_s4>;
+	#sound-dai-cells = <1>;
+}

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

@@ -352,6 +352,7 @@ Required properties:
 				    - "renesas,rcar_sound-r8a7794" (R-Car E2)
 				    - "renesas,rcar_sound-r8a7795" (R-Car H3)
 				    - "renesas,rcar_sound-r8a7796" (R-Car M3-W)
+				    - "renesas,rcar_sound-r8a77965" (R-Car M3-N)
 - reg				: Should contain the register physical address.
 				  required register is
 				   SRU/ADG/SSI      if generation1

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

@@ -7,6 +7,7 @@ Required properties:
 
 - compatible: should be one of the following:
    - "rockchip,rk3066-i2s": for rk3066
+   - "rockchip,px30-i2s", "rockchip,rk3066-i2s": for px30
    - "rockchip,rk3036-i2s", "rockchip,rk3066-i2s": for rk3036
    - "rockchip,rk3188-i2s", "rockchip,rk3066-i2s": for rk3188
    - "rockchip,rk3228-i2s", "rockchip,rk3066-i2s": for rk3228

Documentation/devicetree/bindings/sound/rt5682.txt

@@ -0,0 +1,50 @@
+RT5682 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+- compatible : "realtek,rt5682" or "realtek,rt5682i"
+
+- reg : The I2C address of the device.
+
+Optional properties:
+
+- interrupts : The CODEC's interrupt output.
+
+- realtek,dmic1-data-pin
+  0: dmic1 is not used
+  1: using GPIO2 pin as dmic1 data pin
+  2: using GPIO5 pin as dmic1 data pin
+
+- realtek,dmic1-clk-pin
+  0: using GPIO1 pin as dmic1 clock pin
+  1: using GPIO3 pin as dmic1 clock pin
+
+- realtek,jd-src
+  0: No JD is used
+  1: using JD1 as JD source
+
+- realtek,ldo1-en-gpios : The GPIO that controls the CODEC's LDO1_EN pin.
+
+Pins on the device (for linking into audio routes) for RT5682:
+
+  * DMIC L1
+  * DMIC R1
+  * IN1P
+  * HPOL
+  * HPOR
+
+Example:
+
+rt5682 {
+	compatible = "realtek,rt5682i";
+	reg = <0x1a>;
+	interrupt-parent = <&gpio>;
+	interrupts = <TEGRA_GPIO(U, 6) GPIO_ACTIVE_HIGH>;
+	realtek,ldo1-en-gpios =
+		<&gpio TEGRA_GPIO(R, 2) GPIO_ACTIVE_HIGH>;
+	realtek,dmic1-data-pin = <1>;
+	realtek,dmic1-clk-pin = <1>;
+	realtek,jd-src = <1>;
+};

Documentation/devicetree/bindings/sound/sgtl5000.txt

@@ -17,7 +17,7 @@ Optional properties:
 
 - VDDD-supply : the regulator provider of VDDD
 
-- micbias-resistor-k-ohms : the bias resistor to be used in kOmhs
+- micbias-resistor-k-ohms : the bias resistor to be used in kOhms
 	The resistor can take values of 2k, 4k or 8k.
 	If set to 0 it will be off.
 	If this node is not mentioned or if the value is unknown, then

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

@@ -0,0 +1,12 @@
+Simple Amplifier Audio Driver
+
+Required properties:
+- compatible : "dioo,dio2125" or "simple-audio-amplifier"
+- enable-gpios : the gpio connected to the enable pin of the simple amplifier
+
+Example:
+
+amp: analog-amplifier {
+	compatible = "simple-audio-amplifier";
+	enable-gpios = <&gpio GPIOH_3 0>;
+};

Documentation/devicetree/bindings/sound/tas571x.txt

@@ -7,6 +7,7 @@ powerdown (optional).
 Required properties:
 
 - compatible: should be one of the following:
+  - "ti,tas5707"
   - "ti,tas5711",
   - "ti,tas5717",
   - "ti,tas5719",

Documentation/sound/alsa-configuration.rst

@@ -1568,7 +1568,7 @@ joystick_io
 The driver requires firmware files ``turtlebeach/msndinit.bin`` and
 ``turtlebeach/msndperm.bin`` in the proper firmware directory.
 
-See Documentation/sound/oss/MultiSound for important information
+See Documentation/sound/cards/multisound.sh for important information
 about this driver.  Note that it has been discontinued, but the 
 Voyetra Turtle Beach knowledge base entry for it is still available
 at

Documentation/sound/cards/multisound.sh

@@ -0,0 +1,1139 @@
+#! /bin/sh
+#
+#  Turtle Beach MultiSound Driver Notes
+#  -- Andrew Veliath <andrewtv@usa.net>
+#
+#  Last update:                      September 10, 1998
+#  Corresponding msnd driver:        0.8.3
+#
+# ** This file is a README (top part) and shell archive (bottom part).
+#    The corresponding archived utility sources can be unpacked by
+#    running `sh MultiSound' (the utilities are only needed for the
+#    Pinnacle and Fiji cards). **
+#
+#
+#  -=-=- Getting Firmware -=-=-
+#  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+#
+#  See the section `Obtaining and Creating Firmware Files' in this
+#  document for instructions on obtaining the necessary firmware
+#  files.
+#
+#
+#  Supported Features
+#  ~~~~~~~~~~~~~~~~~~
+#
+#  Currently, full-duplex digital audio (/dev/dsp only, /dev/audio is
+#  not currently available) and mixer functionality (/dev/mixer) are
+#  supported (memory mapped digital audio is not yet supported).
+#  Digital transfers and monitoring can be done as well if you have
+#  the digital daughterboard (see the section on using the S/PDIF port
+#  for more information).
+#
+#  Support for the Turtle Beach MultiSound Hurricane architecture is
+#  composed of the following modules (these can also operate compiled
+#  into the kernel):
+#
+#  snd-msnd-lib           - MultiSound base (requires snd)
+#
+#  snd-msnd-classic       - Base audio/mixer support for Classic, Monetery and
+#                           Tahiti cards
+#
+#  snd-msnd-pinnacle      - Base audio/mixer support for Pinnacle and Fiji cards
+#
+#
+#  Important Notes - Read Before Using
+#  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+#
+#  The firmware files are not included (may change in future).  You
+#  must obtain these images from Turtle Beach (they are included in
+#  the MultiSound Development Kits), and place them in /etc/sound for
+#  example, and give the full paths in the Linux configuration.  If
+#  you are compiling in support for the MultiSound driver rather than
+#  using it as a module, these firmware files must be accessible
+#  during kernel compilation.
+#
+#  Please note these files must be binary files, not assembler.  See
+#  the section later in this document for instructions to obtain these
+#  files.
+#
+#
+#  Configuring Card Resources
+#  ~~~~~~~~~~~~~~~~~~~~~~~~~~
+#
+#  ** This section is very important, as your card may not work at all
+#     or your machine may crash if you do not do this correctly. **
+#
+#  * Classic/Monterey/Tahiti
+#
+#  These cards are configured through the driver snd-msnd-classic.  You must
+#  know the io port, then the driver will select the irq and memory resources
+#  on the card.  It is up to you to know if these are free locations or now,
+#  a conflict can lock the machine up.
+#
+#  * Pinnacle/Fiji
+#
+#  The Pinnacle and Fiji cards have an extra config port, either
+#  0x250, 0x260 or 0x270.  This port can be disabled to have the card
+#  configured strictly through PnP, however you lose the ability to
+#  access the IDE controller and joystick devices on this card when
+#  using PnP.  The included pinnaclecfg program in this shell archive
+#  can be used to configure the card in non-PnP mode, and in PnP mode
+#  you can use isapnptools.  These are described briefly here.
+#
+#  pinnaclecfg is not required; you can use the snd-msnd-pinnacle module
+#  to fully configure the card as well.  However, pinnaclecfg can be
+#  used to change the resource values of a particular device after the
+#  snd-msnd-pinnacle module has been loaded.  If you are compiling the
+#  driver into the kernel, you must set these values during compile
+#  time, however other peripheral resource values can be changed with
+#  the pinnaclecfg program after the kernel is loaded.
+#
+#
+#  *** PnP mode
+#
+#  Use pnpdump to obtain a sample configuration if you can; I was able
+#  to obtain one with the command `pnpdump 1 0x203' -- this may vary
+#  for you (running pnpdump by itself did not work for me).  Then,
+#  edit this file and use isapnp to uncomment and set the card values.
+#  Use these values when inserting the snd-msnd-pinnacle module.  Using
+#  this method, you can set the resources for the DSP and the Kurzweil
+#  synth (Pinnacle).  Since Linux does not directly support PnP
+#  devices, you may have difficulty when using the card in PnP mode
+#  when it the driver is compiled into the kernel.  Using non-PnP mode
+#  is preferable in this case.
+#
+#  Here is an example mypinnacle.conf for isapnp that sets the card to
+#  io base 0x210, irq 5 and mem 0xd8000, and also sets the Kurzweil
+#  synth to 0x330 and irq 9 (may need editing for your system):
+#
+#  (READPORT 0x0203)
+#  (CSN 2)
+#  (IDENTIFY *)
+#
+#  # DSP
+#  (CONFIGURE BVJ0440/-1 (LD 0
+#          (INT 0 (IRQ 5 (MODE +E))) (IO 0 (BASE 0x0210)) (MEM 0 (BASE 0x0d8000))
+#          (ACT Y)))
+#
+#  # Kurzweil Synth (Pinnacle Only)
+#  (CONFIGURE BVJ0440/-1 (LD 1
+#          (IO 0 (BASE 0x0330)) (INT 0 (IRQ 9 (MODE +E)))
+#          (ACT Y)))
+#
+#  (WAITFORKEY)
+#
+#
+#  *** Non-PnP mode
+#
+#  The second way is by running the card in non-PnP mode.  This
+#  actually has some advantages in that you can access some other
+#  devices on the card, such as the joystick and IDE controller.  To
+#  configure the card, unpack this shell archive and build the
+#  pinnaclecfg program.  Using this program, you can assign the
+#  resource values to the card's devices, or disable the devices.  As
+#  an alternative to using pinnaclecfg, you can specify many of the
+#  configuration values when loading the snd-msnd-pinnacle module (or
+#  during kernel configuration when compiling the driver into the
+#  kernel).
+#
+#  If you specify cfg=0x250 for the snd-msnd-pinnacle module, it
+#  automatically configure the card to the given io, irq and memory
+#  values using that config port (the config port is jumper selectable
+#  on the card to 0x250, 0x260 or 0x270).
+#
+#  See the `snd-msnd-pinnacle Additional Options' section below for more
+#  information on these parameters (also, if you compile the driver
+#  directly into the kernel, these extra parameters can be useful
+#  here).
+#
+#
+# ** It is very easy to cause problems in your machine if you choose a
+#    resource value which is incorrect. **
+#
+#
+#  Examples
+#  ~~~~~~~~
+#
+#  * MultiSound Classic/Monterey/Tahiti:
+#
+#  modprobe snd
+#  insmod snd-msnd-lib
+#  insmod snd-msnd-classic io=0x290 irq=7 mem=0xd0000
+#
+#  * MultiSound Pinnacle in PnP mode:
+#
+#  modprobe snd
+#  insmod snd-msnd-lib
+#  isapnp mypinnacle.conf
+#  insmod snd-msnd-pinnacle io=0x210 irq=5 mem=0xd8000 <-- match mypinnacle.conf values
+#
+#  * MultiSound Pinnacle in non-PnP mode (replace 0x250 with your configuration port,
+#    one of 0x250, 0x260 or 0x270):
+#
+#  modprobe snd
+#  insmod snd-msnd-lib
+#  insmod snd-msnd-pinnacle cfg=0x250 io=0x290 irq=5 mem=0xd0000
+#
+# * To use the MPU-compatible Kurzweil synth on the Pinnacle in PnP
+#   mode, add the following (assumes you did `isapnp mypinnacle.conf'):
+#
+#  insmod snd
+#  insmod mpu401 io=0x330 irq=9                    <-- match mypinnacle.conf values
+#
+# * To use the MPU-compatible Kurzweil synth on the Pinnacle in non-PnP
+#   mode, add the following.  Note how we first configure the peripheral's
+#   resources, _then_ install a Linux driver for it:
+#
+#  insmod snd
+#  pinnaclecfg 0x250 mpu 0x330 9
+#  insmod mpu401 io=0x330 irq=9
+#
+#  -- OR you can use the following sequence without pinnaclecfg in non-PnP mode:
+#
+#  modprobe snd
+#  insmod snd-msnd-lib
+#  insmod snd-msnd-pinnacle cfg=0x250 io=0x290 irq=5 mem=0xd0000 mpu_io=0x330 mpu_irq=9
+#  insmod snd
+#  insmod mpu401 io=0x330 irq=9
+#
+# * To setup the joystick port on the Pinnacle in non-PnP mode (though
+#   you have to find the actual Linux joystick driver elsewhere), you
+#   can use pinnaclecfg:
+#
+#   pinnaclecfg 0x250 joystick 0x200
+#
+#  -- OR you can configure this using snd-msnd-pinnacle with the following:
+#
+#  modprobe snd
+#  insmod snd-msnd-lib
+#  insmod snd-msnd-pinnacle cfg=0x250 io=0x290 irq=5 mem=0xd0000 joystick_io=0x200
+#
+#
+#  snd-msnd-classic, snd-msnd-pinnacle Required Options
+#  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+#
+#  If the following options are not given, the module will not load.
+#  Examine the kernel message log for informative error messages.
+#  WARNING--probing isn't supported so try to make sure you have the
+#  correct shared memory area, otherwise you may experience problems.
+#
+#  io                   I/O base of DSP, e.g. io=0x210
+#  irq                  IRQ number, e.g. irq=5
+#  mem                  Shared memory area, e.g. mem=0xd8000
+#
+#
+#  snd-msnd-classic, snd-msnd-pinnacle Additional Options
+#  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+#
+#  fifosize             The digital audio FIFOs, in kilobytes.  If not
+#                       specified, the default will be used.  Increasing
+#                       this value will reduce the chance of a FIFO
+#                       underflow at the expense of increasing overall
+#                       latency.  For example, fifosize=512 will
+#                       allocate 512kB read and write FIFOs (1MB total).
+#                       While this may reduce dropouts, a heavy machine
+#                       load will undoubtedly starve the FIFO of data
+#                       and you will eventually get dropouts.  One
+#                       option is to alter the scheduling priority of
+#                       the playback process, using `nice' or some form
+#                       of POSIX soft real-time scheduling.
+#
+#  calibrate_signal     Setting this to one calibrates the ADCs to the
+#                       signal, zero calibrates to the card (defaults
+#                       to zero).
+#
+#
+#  snd-msnd-pinnacle Additional Options
+#  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+#
+#  digital              Specify digital=1 to enable the S/PDIF input
+#                       if you have the digital daughterboard
+#                       adapter. This will enable access to the
+#                       DIGITAL1 input for the soundcard in the mixer.
+#                       Some mixer programs might have trouble setting
+#                       the DIGITAL1 source as an input.  If you have
+#                       trouble, you can try the setdigital.c program
+#                       at the bottom of this document.
+#
+#  cfg                  Non-PnP configuration port for the Pinnacle
+#                       and Fiji (typically 0x250, 0x260 or 0x270,
+#                       depending on the jumper configuration).  If
+#                       this option is omitted, then it is assumed
+#                       that the card is in PnP mode, and that the
+#                       specified DSP resource values are already
+#                       configured with PnP (i.e. it won't attempt to
+#                       do any sort of configuration).
+#
+#  When the Pinnacle is in non-PnP mode, you can use the following
+#  options to configure particular devices.  If a full specification
+#  for a device is not given, then the device is not configured.  Note
+#  that you still must use a Linux driver for any of these devices
+#  once their resources are setup (such as the Linux joystick driver,
+#  or the MPU401 driver from OSS for the Kurzweil synth).
+#
+#  mpu_io               I/O port of MPU (on-board Kurzweil synth)
+#  mpu_irq              IRQ of MPU (on-board Kurzweil synth)
+#  ide_io0		First I/O port of IDE controller
+#  ide_io1		Second I/O port of IDE controller
+#  ide_irq		IRQ IDE controller
+#  joystick_io          I/O port of joystick
+#
+#
+#  Obtaining and Creating Firmware Files
+#  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+#
+#       For the Classic/Tahiti/Monterey
+#       ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+#
+#  Download to /tmp and unzip the following file from Turtle Beach:
+#
+#       ftp://ftp.voyetra.com/pub/tbs/msndcl/msndvkit.zip
+#
+#  When unzipped, unzip the file named MsndFiles.zip.  Then copy the
+#  following firmware files to /etc/sound (note the file renaming):
+#
+#    cp DSPCODE/MSNDINIT.BIN /etc/sound/msndinit.bin
+#    cp DSPCODE/MSNDPERM.REB /etc/sound/msndperm.bin
+#
+#  When configuring the Linux kernel, specify /etc/sound/msndinit.bin and
+#  /etc/sound/msndperm.bin for the two firmware files (Linux kernel
+#  versions older than 2.2 do not ask for firmware paths, and are
+#  hardcoded to /etc/sound).
+#
+#  If you are compiling the driver into the kernel, these files must
+#  be accessible during compilation, but will not be needed later.
+#  The files must remain, however, if the driver is used as a module.
+#
+#
+#       For the Pinnacle/Fiji
+#       ~~~~~~~~~~~~~~~~~~~~~
+#
+#  Download to /tmp and unzip the following file from Turtle Beach (be
+#  sure to use the entire URL; some have had trouble navigating to the
+#  URL):
+#
+#       ftp://ftp.voyetra.com/pub/tbs/pinn/pnddk100.zip
+#
+#  Unpack this shell archive, and run make in the created directory
+#  (you need a C compiler and flex to build the utilities).  This
+#  should give you the executables conv, pinnaclecfg and setdigital.
+#  conv is only used temporarily here to create the firmware files,
+#  while pinnaclecfg is used to configure the Pinnacle or Fiji card in
+#  non-PnP mode, and setdigital can be used to set the S/PDIF input on
+#  the mixer (pinnaclecfg and setdigital should be copied to a
+#  convenient place, possibly run during system initialization).
+#
+#  To generating the firmware files with the `conv' program, we create
+#  the binary firmware files by doing the following conversion
+#  (assuming the archive unpacked into a directory named PINNDDK):
+#
+#    ./conv < PINNDDK/dspcode/pndspini.asm > /etc/sound/pndspini.bin
+#    ./conv < PINNDDK/dspcode/pndsperm.asm > /etc/sound/pndsperm.bin
+#
+#  The conv (and conv.l) program is not needed after conversion and can
+#  be safely deleted.  Then, when configuring the Linux kernel, specify
+#  /etc/sound/pndspini.bin and /etc/sound/pndsperm.bin for the two
+#  firmware files (Linux kernel versions older than 2.2 do not ask for
+#  firmware paths, and are hardcoded to /etc/sound).
+#
+#  If you are compiling the driver into the kernel, these files must
+#  be accessible during compilation, but will not be needed later.
+#  The files must remain, however, if the driver is used as a module.
+#
+#
+#  Using Digital I/O with the S/PDIF Port
+#  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+#
+#  If you have a Pinnacle or Fiji with the digital daughterboard and
+#  want to set it as the input source, you can use this program if you
+#  have trouble trying to do it with a mixer program (be sure to
+#  insert the module with the digital=1 option, or say Y to the option
+#  during compiled-in kernel operation).  Upon selection of the S/PDIF
+#  port, you should be able monitor and record from it.
+#
+#  There is something to note about using the S/PDIF port.  Digital
+#  timing is taken from the digital signal, so if a signal is not
+#  connected to the port and it is selected as recording input, you
+#  will find PCM playback to be distorted in playback rate.  Also,
+#  attempting to record at a sampling rate other than the DAT rate may
+#  be problematic (i.e. trying to record at 8000Hz when the DAT signal
+#  is 44100Hz).  If you have a problem with this, set the recording
+#  input to analog if you need to record at a rate other than that of
+#  the DAT rate.
+#
+#
+#  -- Shell archive attached below, just run `sh MultiSound' to extract.
+#     Contains Pinnacle/Fiji utilities to convert firmware, configure
+#     in non-PnP mode, and select the DIGITAL1 input for the mixer.
+#
+#
+#!/bin/sh
+# This is a shell archive (produced by GNU sharutils 4.2).
+# To extract the files from this archive, save it to some FILE, remove
+# everything before the `!/bin/sh' line above, then type `sh FILE'.
+#
+# Made on 1998-12-04 10:07 EST by <andrewtv@ztransform.velsoft.com>.
+# Source directory was `/home/andrewtv/programming/pinnacle/pinnacle'.
+#
+# Existing files will *not* be overwritten unless `-c' is specified.
+#
+# This shar contains:
+# length mode       name
+# ------ ---------- ------------------------------------------
+#   2064 -rw-rw-r-- MultiSound.d/setdigital.c
+#  10224 -rw-rw-r-- MultiSound.d/pinnaclecfg.c
+#    106 -rw-rw-r-- MultiSound.d/Makefile
+#    146 -rw-rw-r-- MultiSound.d/conv.l
+#   1491 -rw-rw-r-- MultiSound.d/msndreset.c
+#
+save_IFS="${IFS}"
+IFS="${IFS}:"
+gettext_dir=FAILED
+locale_dir=FAILED
+first_param="$1"
+for dir in $PATH
+do
+  if test "$gettext_dir" = FAILED && test -f $dir/gettext \
+     && ($dir/gettext --version >/dev/null 2>&1)
+  then
+    set `$dir/gettext --version 2>&1`
+    if test "$3" = GNU
+    then
+      gettext_dir=$dir
+    fi
+  fi
+  if test "$locale_dir" = FAILED && test -f $dir/shar \
+     && ($dir/shar --print-text-domain-dir >/dev/null 2>&1)
+  then
+    locale_dir=`$dir/shar --print-text-domain-dir`
+  fi
+done
+IFS="$save_IFS"
+if test "$locale_dir" = FAILED || test "$gettext_dir" = FAILED
+then
+  echo=echo
+else
+  TEXTDOMAINDIR=$locale_dir
+  export TEXTDOMAINDIR
+  TEXTDOMAIN=sharutils
+  export TEXTDOMAIN
+  echo="$gettext_dir/gettext -s"
+fi
+touch -am 1231235999 $$.touch >/dev/null 2>&1
+if test ! -f 1231235999 && test -f $$.touch; then
+  shar_touch=touch
+else
+  shar_touch=:
+  echo
+  $echo 'WARNING: not restoring timestamps.  Consider getting and'
+  $echo "installing GNU \`touch', distributed in GNU File Utilities..."
+  echo
+fi
+rm -f 1231235999 $$.touch
+#
+if mkdir _sh01426; then
+  $echo 'x -' 'creating lock directory'
+else
+  $echo 'failed to create lock directory'
+  exit 1
+fi
+# ============= MultiSound.d/setdigital.c ==============
+if test ! -d 'MultiSound.d'; then
+  $echo 'x -' 'creating directory' 'MultiSound.d'
+  mkdir 'MultiSound.d'
+fi
+if test -f 'MultiSound.d/setdigital.c' && test "$first_param" != -c; then
+  $echo 'x -' SKIPPING 'MultiSound.d/setdigital.c' '(file already exists)'
+else
+  $echo 'x -' extracting 'MultiSound.d/setdigital.c' '(text)'
+  sed 's/^X//' << 'SHAR_EOF' > 'MultiSound.d/setdigital.c' &&
+/*********************************************************************
+X *
+X * setdigital.c - sets the DIGITAL1 input for a mixer
+X *
+X * Copyright (C) 1998 Andrew Veliath
+X *
+X * This program is free software; you can redistribute it and/or modify
+X * it under the terms of the GNU General Public License as published by
+X * the Free Software Foundation; either version 2 of the License, or
+X * (at your option) any later version.
+X *
+X * This program is distributed in the hope that it will be useful,
+X * but WITHOUT ANY WARRANTY; without even the implied warranty of
+X * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+X * GNU General Public License for more details.
+X *
+X * You should have received a copy of the GNU General Public License
+X * along with this program; if not, write to the Free Software
+X * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+X *
+X ********************************************************************/
+X
+#include <stdio.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <fcntl.h>
+#include <sys/types.h>
+#include <sys/stat.h>
+#include <sys/ioctl.h>
+#include <sys/soundcard.h>
+X
+int main(int argc, char *argv[])
+{
+X	int fd;
+X	unsigned long recmask, recsrc;
+X
+X	if (argc != 2) {
+X		fprintf(stderr, "usage: setdigital <mixer device>\n");
+X		exit(1);
+X	}
+X
+X	if ((fd = open(argv[1], O_RDWR)) < 0) {
+X		perror(argv[1]);
+X		exit(1);
+X	}
+X
+X	if (ioctl(fd, SOUND_MIXER_READ_RECMASK, &recmask) < 0) {
+X		fprintf(stderr, "error: ioctl read recording mask failed\n");
+X		perror("ioctl");
+X		close(fd);
+X		exit(1);
+X	}
+X
+X	if (!(recmask & SOUND_MASK_DIGITAL1)) {
+X		fprintf(stderr, "error: cannot find DIGITAL1 device in mixer\n");
+X		close(fd);
+X		exit(1);
+X	}
+X
+X	if (ioctl(fd, SOUND_MIXER_READ_RECSRC, &recsrc) < 0) {
+X		fprintf(stderr, "error: ioctl read recording source failed\n");
+X		perror("ioctl");
+X		close(fd);
+X		exit(1);
+X	}
+X
+X	recsrc |= SOUND_MASK_DIGITAL1;
+X
+X	if (ioctl(fd, SOUND_MIXER_WRITE_RECSRC, &recsrc) < 0) {
+X		fprintf(stderr, "error: ioctl write recording source failed\n");
+X		perror("ioctl");
+X		close(fd);
+X		exit(1);
+X	}
+X
+X	close(fd);
+X
+X	return 0;
+}
+SHAR_EOF
+  $shar_touch -am 1204092598 'MultiSound.d/setdigital.c' &&
+  chmod 0664 'MultiSound.d/setdigital.c' ||
+  $echo 'restore of' 'MultiSound.d/setdigital.c' 'failed'
+  if ( md5sum --help 2>&1 | grep 'sage: md5sum \[' ) >/dev/null 2>&1 \
+  && ( md5sum --version 2>&1 | grep -v 'textutils 1.12' ) >/dev/null; then
+    md5sum -c << SHAR_EOF >/dev/null 2>&1 \
+    || $echo 'MultiSound.d/setdigital.c:' 'MD5 check failed'
+e87217fc3e71288102ba41fd81f71ec4  MultiSound.d/setdigital.c
+SHAR_EOF
+  else
+    shar_count="`LC_ALL= LC_CTYPE= LANG= wc -c < 'MultiSound.d/setdigital.c'`"
+    test 2064 -eq "$shar_count" ||
+    $echo 'MultiSound.d/setdigital.c:' 'original size' '2064,' 'current size' "$shar_count!"
+  fi
+fi
+# ============= MultiSound.d/pinnaclecfg.c ==============
+if test -f 'MultiSound.d/pinnaclecfg.c' && test "$first_param" != -c; then
+  $echo 'x -' SKIPPING 'MultiSound.d/pinnaclecfg.c' '(file already exists)'
+else
+  $echo 'x -' extracting 'MultiSound.d/pinnaclecfg.c' '(text)'
+  sed 's/^X//' << 'SHAR_EOF' > 'MultiSound.d/pinnaclecfg.c' &&
+/*********************************************************************
+X *
+X * pinnaclecfg.c - Pinnacle/Fiji Device Configuration Program
+X *
+X * This is for NON-PnP mode only.  For PnP mode, use isapnptools.
+X *
+X * This is Linux-specific, and must be run with root permissions.
+X *
+X * Part of the Turtle Beach MultiSound Sound Card Driver for Linux
+X *
+X * Copyright (C) 1998 Andrew Veliath
+X *
+X * This program is free software; you can redistribute it and/or modify
+X * it under the terms of the GNU General Public License as published by
+X * the Free Software Foundation; either version 2 of the License, or
+X * (at your option) any later version.
+X *
+X * This program is distributed in the hope that it will be useful,
+X * but WITHOUT ANY WARRANTY; without even the implied warranty of
+X * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+X * GNU General Public License for more details.
+X *
+X * You should have received a copy of the GNU General Public License
+X * along with this program; if not, write to the Free Software
+X * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+X *
+X ********************************************************************/
+X
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <errno.h>
+#include <unistd.h>
+#include <asm/types.h>
+#include <sys/io.h>
+X
+#define IREG_LOGDEVICE		0x07
+#define IREG_ACTIVATE		0x30
+#define LD_ACTIVATE		0x01
+#define LD_DISACTIVATE		0x00
+#define IREG_EECONTROL		0x3F
+#define IREG_MEMBASEHI		0x40
+#define IREG_MEMBASELO		0x41
+#define IREG_MEMCONTROL		0x42
+#define IREG_MEMRANGEHI		0x43
+#define IREG_MEMRANGELO		0x44
+#define MEMTYPE_8BIT		0x00
+#define MEMTYPE_16BIT		0x02
+#define MEMTYPE_RANGE		0x00
+#define MEMTYPE_HIADDR		0x01
+#define IREG_IO0_BASEHI		0x60
+#define IREG_IO0_BASELO		0x61
+#define IREG_IO1_BASEHI		0x62
+#define IREG_IO1_BASELO		0x63
+#define IREG_IRQ_NUMBER		0x70
+#define IREG_IRQ_TYPE		0x71
+#define IRQTYPE_HIGH		0x02
+#define IRQTYPE_LOW		0x00
+#define IRQTYPE_LEVEL		0x01
+#define IRQTYPE_EDGE		0x00
+X
+#define HIBYTE(w)		((BYTE)(((WORD)(w) >> 8) & 0xFF))
+#define LOBYTE(w)		((BYTE)(w))
+#define MAKEWORD(low,hi)	((WORD)(((BYTE)(low))|(((WORD)((BYTE)(hi)))<<8)))
+X
+typedef __u8			BYTE;
+typedef __u16			USHORT;
+typedef __u16			WORD;
+X
+static int config_port = -1;
+X
+static int msnd_write_cfg(int cfg, int reg, int value)
+{
+X	outb(reg, cfg);
+X	outb(value, cfg + 1);
+X	if (value != inb(cfg + 1)) {
+X		fprintf(stderr, "error: msnd_write_cfg: I/O error\n");
+X		return -EIO;
+X	}
+X	return 0;
+}
+X
+static int msnd_read_cfg(int cfg, int reg)
+{
+X	outb(reg, cfg);
+X	return inb(cfg + 1);
+}
+X
+static int msnd_write_cfg_io0(int cfg, int num, WORD io)
+{
+X	if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+X		return -EIO;
+X	if (msnd_write_cfg(cfg, IREG_IO0_BASEHI, HIBYTE(io)))
+X		return -EIO;
+X	if (msnd_write_cfg(cfg, IREG_IO0_BASELO, LOBYTE(io)))
+X		return -EIO;
+X	return 0;
+}
+X
+static int msnd_read_cfg_io0(int cfg, int num, WORD *io)
+{
+X	if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+X		return -EIO;
+X
+X	*io = MAKEWORD(msnd_read_cfg(cfg, IREG_IO0_BASELO),
+X		       msnd_read_cfg(cfg, IREG_IO0_BASEHI));
+X
+X	return 0;
+}
+X
+static int msnd_write_cfg_io1(int cfg, int num, WORD io)
+{
+X	if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+X		return -EIO;
+X	if (msnd_write_cfg(cfg, IREG_IO1_BASEHI, HIBYTE(io)))
+X		return -EIO;
+X	if (msnd_write_cfg(cfg, IREG_IO1_BASELO, LOBYTE(io)))
+X		return -EIO;
+X	return 0;
+}
+X
+static int msnd_read_cfg_io1(int cfg, int num, WORD *io)
+{
+X	if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+X		return -EIO;
+X
+X	*io = MAKEWORD(msnd_read_cfg(cfg, IREG_IO1_BASELO),
+X		       msnd_read_cfg(cfg, IREG_IO1_BASEHI));
+X
+X	return 0;
+}
+X
+static int msnd_write_cfg_irq(int cfg, int num, WORD irq)
+{
+X	if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+X		return -EIO;
+X	if (msnd_write_cfg(cfg, IREG_IRQ_NUMBER, LOBYTE(irq)))
+X		return -EIO;
+X	if (msnd_write_cfg(cfg, IREG_IRQ_TYPE, IRQTYPE_EDGE))
+X		return -EIO;
+X	return 0;
+}
+X
+static int msnd_read_cfg_irq(int cfg, int num, WORD *irq)
+{
+X	if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+X		return -EIO;
+X
+X	*irq = msnd_read_cfg(cfg, IREG_IRQ_NUMBER);
+X
+X	return 0;
+}
+X
+static int msnd_write_cfg_mem(int cfg, int num, int mem)
+{
+X	WORD wmem;
+X
+X	mem >>= 8;
+X	mem &= 0xfff;
+X	wmem = (WORD)mem;
+X	if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+X		return -EIO;
+X	if (msnd_write_cfg(cfg, IREG_MEMBASEHI, HIBYTE(wmem)))
+X		return -EIO;
+X	if (msnd_write_cfg(cfg, IREG_MEMBASELO, LOBYTE(wmem)))
+X		return -EIO;
+X	if (wmem && msnd_write_cfg(cfg, IREG_MEMCONTROL, (MEMTYPE_HIADDR | MEMTYPE_16BIT)))
+X		return -EIO;
+X	return 0;
+}
+X
+static int msnd_read_cfg_mem(int cfg, int num, int *mem)
+{
+X	if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+X		return -EIO;
+X
+X	*mem = MAKEWORD(msnd_read_cfg(cfg, IREG_MEMBASELO),
+X			msnd_read_cfg(cfg, IREG_MEMBASEHI));
+X	*mem <<= 8;
+X
+X	return 0;
+}
+X
+static int msnd_activate_logical(int cfg, int num)
+{
+X	if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+X		return -EIO;
+X	if (msnd_write_cfg(cfg, IREG_ACTIVATE, LD_ACTIVATE))
+X		return -EIO;
+X	return 0;
+}
+X
+static int msnd_write_cfg_logical(int cfg, int num, WORD io0, WORD io1, WORD irq, int mem)
+{
+X	if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+X		return -EIO;
+X	if (msnd_write_cfg_io0(cfg, num, io0))
+X		return -EIO;
+X	if (msnd_write_cfg_io1(cfg, num, io1))
+X		return -EIO;
+X	if (msnd_write_cfg_irq(cfg, num, irq))
+X		return -EIO;
+X	if (msnd_write_cfg_mem(cfg, num, mem))
+X		return -EIO;
+X	if (msnd_activate_logical(cfg, num))
+X		return -EIO;
+X	return 0;
+}
+X
+static int msnd_read_cfg_logical(int cfg, int num, WORD *io0, WORD *io1, WORD *irq, int *mem)
+{
+X	if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+X		return -EIO;
+X	if (msnd_read_cfg_io0(cfg, num, io0))
+X		return -EIO;
+X	if (msnd_read_cfg_io1(cfg, num, io1))
+X		return -EIO;
+X	if (msnd_read_cfg_irq(cfg, num, irq))
+X		return -EIO;
+X	if (msnd_read_cfg_mem(cfg, num, mem))
+X		return -EIO;
+X	return 0;
+}
+X
+static void usage(void)
+{
+X	fprintf(stderr,
+X		"\n"
+X		"pinnaclecfg 1.0\n"
+X		"\n"
+X		"usage: pinnaclecfg <config port> [device config]\n"
+X		"\n"
+X		"This is for use with the card in NON-PnP mode only.\n"
+X		"\n"
+X		"Available devices (not all available for Fiji):\n"
+X		"\n"
+X		"        Device                       Description\n"
+X		"        -------------------------------------------------------------------\n"
+X		"        reset                        Reset all devices (i.e. disable)\n"
+X		"        show                         Display current device configurations\n"
+X		"\n"
+X		"        dsp <io> <irq> <mem>         Audio device\n"
+X		"        mpu <io> <irq>               Internal Kurzweil synth\n"
+X		"        ide <io0> <io1> <irq>        On-board IDE controller\n"
+X		"        joystick <io>                Joystick port\n"
+X		"\n");
+X	exit(1);
+}
+X
+static int cfg_reset(void)
+{
+X	int i;
+X
+X	for (i = 0; i < 4; ++i)
+X		msnd_write_cfg_logical(config_port, i, 0, 0, 0, 0);
+X
+X	return 0;
+}
+X
+static int cfg_show(void)
+{
+X	int i;
+X	int count = 0;
+X
+X	for (i = 0; i < 4; ++i) {
+X		WORD io0, io1, irq;
+X		int mem;
+X		msnd_read_cfg_logical(config_port, i, &io0, &io1, &irq, &mem);
+X		switch (i) {
+X		case 0:
+X			if (io0 || irq || mem) {
+X				printf("dsp 0x%x %d 0x%x\n", io0, irq, mem);
+X				++count;
+X			}
+X			break;
+X		case 1:
+X			if (io0 || irq) {
+X				printf("mpu 0x%x %d\n", io0, irq);
+X				++count;
+X			}
+X			break;
+X		case 2:
+X			if (io0 || io1 || irq) {
+X				printf("ide 0x%x 0x%x %d\n", io0, io1, irq);
+X				++count;
+X			}
+X			break;
+X		case 3:
+X			if (io0) {
+X				printf("joystick 0x%x\n", io0);
+X				++count;
+X			}
+X			break;
+X		}
+X	}
+X
+X	if (count == 0)
+X		fprintf(stderr, "no devices configured\n");
+X
+X	return 0;
+}
+X
+static int cfg_dsp(int argc, char *argv[])
+{
+X	int io, irq, mem;
+X
+X	if (argc < 3 ||
+X	    sscanf(argv[0], "0x%x", &io) != 1 ||
+X	    sscanf(argv[1], "%d", &irq) != 1 ||
+X	    sscanf(argv[2], "0x%x", &mem) != 1)
+X		usage();
+X
+X	if (!(io == 0x290 ||
+X	      io == 0x260 ||
+X	      io == 0x250 ||
+X	      io == 0x240 ||
+X	      io == 0x230 ||
+X	      io == 0x220 ||
+X	      io == 0x210 ||
+X	      io == 0x3e0)) {
+X		fprintf(stderr, "error: io must be one of "
+X			"210, 220, 230, 240, 250, 260, 290, or 3E0\n");
+X		usage();
+X	}
+X
+X	if (!(irq == 5 ||
+X	      irq == 7 ||
+X	      irq == 9 ||
+X	      irq == 10 ||
+X	      irq == 11 ||
+X	      irq == 12)) {
+X		fprintf(stderr, "error: irq must be one of "
+X			"5, 7, 9, 10, 11 or 12\n");
+X		usage();
+X	}
+X
+X	if (!(mem == 0xb0000 ||
+X	      mem == 0xc8000 ||
+X	      mem == 0xd0000 ||
+X	      mem == 0xd8000 ||
+X	      mem == 0xe0000 ||
+X	      mem == 0xe8000)) {
+X		fprintf(stderr, "error: mem must be one of "
+X			"0xb0000, 0xc8000, 0xd0000, 0xd8000, 0xe0000 or 0xe8000\n");
+X		usage();
+X	}
+X
+X	return msnd_write_cfg_logical(config_port, 0, io, 0, irq, mem);
+}
+X
+static int cfg_mpu(int argc, char *argv[])
+{
+X	int io, irq;
+X
+X	if (argc < 2 ||
+X	    sscanf(argv[0], "0x%x", &io) != 1 ||
+X	    sscanf(argv[1], "%d", &irq) != 1)
+X		usage();
+X
+X	return msnd_write_cfg_logical(config_port, 1, io, 0, irq, 0);
+}
+X
+static int cfg_ide(int argc, char *argv[])
+{
+X	int io0, io1, irq;
+X
+X	if (argc < 3 ||
+X	    sscanf(argv[0], "0x%x", &io0) != 1 ||
+X	    sscanf(argv[0], "0x%x", &io1) != 1 ||
+X	    sscanf(argv[1], "%d", &irq) != 1)
+X		usage();
+X
+X	return msnd_write_cfg_logical(config_port, 2, io0, io1, irq, 0);
+}
+X
+static int cfg_joystick(int argc, char *argv[])
+{
+X	int io;
+X
+X	if (argc < 1 ||
+X	    sscanf(argv[0], "0x%x", &io) != 1)
+X		usage();
+X
+X	return msnd_write_cfg_logical(config_port, 3, io, 0, 0, 0);
+}
+X
+int main(int argc, char *argv[])
+{
+X	char *device;
+X	int rv = 0;
+X
+X	--argc; ++argv;
+X
+X	if (argc < 2)
+X		usage();
+X
+X	sscanf(argv[0], "0x%x", &config_port);
+X	if (config_port != 0x250 && config_port != 0x260 && config_port != 0x270) {
+X		fprintf(stderr, "error: <config port> must be 0x250, 0x260 or 0x270\n");
+X		exit(1);
+X	}
+X	if (ioperm(config_port, 2, 1)) {
+X		perror("ioperm");
+X		fprintf(stderr, "note: pinnaclecfg must be run as root\n");
+X		exit(1);
+X	}
+X	device = argv[1];
+X
+X	argc -= 2; argv += 2;
+X
+X	if (strcmp(device, "reset") == 0)
+X		rv = cfg_reset();
+X	else if (strcmp(device, "show") == 0)
+X		rv = cfg_show();
+X	else if (strcmp(device, "dsp") == 0)
+X		rv = cfg_dsp(argc, argv);
+X	else if (strcmp(device, "mpu") == 0)
+X		rv = cfg_mpu(argc, argv);
+X	else if (strcmp(device, "ide") == 0)
+X		rv = cfg_ide(argc, argv);
+X	else if (strcmp(device, "joystick") == 0)
+X		rv = cfg_joystick(argc, argv);
+X	else {
+X		fprintf(stderr, "error: unknown device %s\n", device);
+X		usage();
+X	}
+X
+X	if (rv)
+X		fprintf(stderr, "error: device configuration failed\n");
+X
+X	return 0;
+}
+SHAR_EOF
+  $shar_touch -am 1204092598 'MultiSound.d/pinnaclecfg.c' &&
+  chmod 0664 'MultiSound.d/pinnaclecfg.c' ||
+  $echo 'restore of' 'MultiSound.d/pinnaclecfg.c' 'failed'
+  if ( md5sum --help 2>&1 | grep 'sage: md5sum \[' ) >/dev/null 2>&1 \
+  && ( md5sum --version 2>&1 | grep -v 'textutils 1.12' ) >/dev/null; then
+    md5sum -c << SHAR_EOF >/dev/null 2>&1 \
+    || $echo 'MultiSound.d/pinnaclecfg.c:' 'MD5 check failed'
+366bdf27f0db767a3c7921d0a6db20fe  MultiSound.d/pinnaclecfg.c
+SHAR_EOF
+  else
+    shar_count="`LC_ALL= LC_CTYPE= LANG= wc -c < 'MultiSound.d/pinnaclecfg.c'`"
+    test 10224 -eq "$shar_count" ||
+    $echo 'MultiSound.d/pinnaclecfg.c:' 'original size' '10224,' 'current size' "$shar_count!"
+  fi
+fi
+# ============= MultiSound.d/Makefile ==============
+if test -f 'MultiSound.d/Makefile' && test "$first_param" != -c; then
+  $echo 'x -' SKIPPING 'MultiSound.d/Makefile' '(file already exists)'
+else
+  $echo 'x -' extracting 'MultiSound.d/Makefile' '(text)'
+  sed 's/^X//' << 'SHAR_EOF' > 'MultiSound.d/Makefile' &&
+CC	= gcc
+CFLAGS	= -O
+PROGS	= setdigital msndreset pinnaclecfg conv
+X
+all: $(PROGS)
+X
+clean:
+X	rm -f $(PROGS)
+SHAR_EOF
+  $shar_touch -am 1204092398 'MultiSound.d/Makefile' &&
+  chmod 0664 'MultiSound.d/Makefile' ||
+  $echo 'restore of' 'MultiSound.d/Makefile' 'failed'
+  if ( md5sum --help 2>&1 | grep 'sage: md5sum \[' ) >/dev/null 2>&1 \
+  && ( md5sum --version 2>&1 | grep -v 'textutils 1.12' ) >/dev/null; then
+    md5sum -c << SHAR_EOF >/dev/null 2>&1 \
+    || $echo 'MultiSound.d/Makefile:' 'MD5 check failed'
+76ca8bb44e3882edcf79c97df6c81845  MultiSound.d/Makefile
+SHAR_EOF
+  else
+    shar_count="`LC_ALL= LC_CTYPE= LANG= wc -c < 'MultiSound.d/Makefile'`"
+    test 106 -eq "$shar_count" ||
+    $echo 'MultiSound.d/Makefile:' 'original size' '106,' 'current size' "$shar_count!"
+  fi
+fi
+# ============= MultiSound.d/conv.l ==============
+if test -f 'MultiSound.d/conv.l' && test "$first_param" != -c; then
+  $echo 'x -' SKIPPING 'MultiSound.d/conv.l' '(file already exists)'
+else
+  $echo 'x -' extracting 'MultiSound.d/conv.l' '(text)'
+  sed 's/^X//' << 'SHAR_EOF' > 'MultiSound.d/conv.l' &&
+%%
+[ \n\t,\r]
+\;.*
+DB
+[0-9A-Fa-f]+H	{ int n; sscanf(yytext, "%xH", &n); printf("%c", n); }
+%%
+int yywrap() { return 1; }
+void main() { yylex(); }
+SHAR_EOF
+  $shar_touch -am 0828231798 'MultiSound.d/conv.l' &&
+  chmod 0664 'MultiSound.d/conv.l' ||
+  $echo 'restore of' 'MultiSound.d/conv.l' 'failed'
+  if ( md5sum --help 2>&1 | grep 'sage: md5sum \[' ) >/dev/null 2>&1 \
+  && ( md5sum --version 2>&1 | grep -v 'textutils 1.12' ) >/dev/null; then
+    md5sum -c << SHAR_EOF >/dev/null 2>&1 \
+    || $echo 'MultiSound.d/conv.l:' 'MD5 check failed'
+d2411fc32cd71a00dcdc1f009e858dd2  MultiSound.d/conv.l
+SHAR_EOF
+  else
+    shar_count="`LC_ALL= LC_CTYPE= LANG= wc -c < 'MultiSound.d/conv.l'`"
+    test 146 -eq "$shar_count" ||
+    $echo 'MultiSound.d/conv.l:' 'original size' '146,' 'current size' "$shar_count!"
+  fi
+fi
+# ============= MultiSound.d/msndreset.c ==============
+if test -f 'MultiSound.d/msndreset.c' && test "$first_param" != -c; then
+  $echo 'x -' SKIPPING 'MultiSound.d/msndreset.c' '(file already exists)'
+else
+  $echo 'x -' extracting 'MultiSound.d/msndreset.c' '(text)'
+  sed 's/^X//' << 'SHAR_EOF' > 'MultiSound.d/msndreset.c' &&
+/*********************************************************************
+X *
+X * msndreset.c - resets the MultiSound card
+X *
+X * Copyright (C) 1998 Andrew Veliath
+X *
+X * This program is free software; you can redistribute it and/or modify
+X * it under the terms of the GNU General Public License as published by
+X * the Free Software Foundation; either version 2 of the License, or
+X * (at your option) any later version.
+X *
+X * This program is distributed in the hope that it will be useful,
+X * but WITHOUT ANY WARRANTY; without even the implied warranty of
+X * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+X * GNU General Public License for more details.
+X *
+X * You should have received a copy of the GNU General Public License
+X * along with this program; if not, write to the Free Software
+X * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+X *
+X ********************************************************************/
+X
+#include <stdio.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <fcntl.h>
+#include <sys/types.h>
+#include <sys/stat.h>
+#include <sys/ioctl.h>
+#include <sys/soundcard.h>
+X
+int main(int argc, char *argv[])
+{
+X	int fd;
+X
+X	if (argc != 2) {
+X		fprintf(stderr, "usage: msndreset <mixer device>\n");
+X		exit(1);
+X	}
+X
+X	if ((fd = open(argv[1], O_RDWR)) < 0) {
+X		perror(argv[1]);
+X		exit(1);
+X	}
+X
+X	if (ioctl(fd, SOUND_MIXER_PRIVATE1, 0) < 0) {
+X		fprintf(stderr, "error: msnd ioctl reset failed\n");
+X		perror("ioctl");
+X		close(fd);
+X		exit(1);
+X	}
+X
+X	close(fd);
+X
+X	return 0;
+}
+SHAR_EOF
+  $shar_touch -am 1204100698 'MultiSound.d/msndreset.c' &&
+  chmod 0664 'MultiSound.d/msndreset.c' ||
+  $echo 'restore of' 'MultiSound.d/msndreset.c' 'failed'
+  if ( md5sum --help 2>&1 | grep 'sage: md5sum \[' ) >/dev/null 2>&1 \
+  && ( md5sum --version 2>&1 | grep -v 'textutils 1.12' ) >/dev/null; then
+    md5sum -c << SHAR_EOF >/dev/null 2>&1 \
+    || $echo 'MultiSound.d/msndreset.c:' 'MD5 check failed'
+c52f876521084e8eb25e12e01dcccb8a  MultiSound.d/msndreset.c
+SHAR_EOF
+  else
+    shar_count="`LC_ALL= LC_CTYPE= LANG= wc -c < 'MultiSound.d/msndreset.c'`"
+    test 1491 -eq "$shar_count" ||
+    $echo 'MultiSound.d/msndreset.c:' 'original size' '1491,' 'current size' "$shar_count!"
+  fi
+fi
+rm -fr _sh01426
+exit 0

Documentation/sound/hd-audio/models.rst

@@ -34,6 +34,22 @@ ALC262
 ======
 inv-dmic
     Inverted internal mic workaround
+fsc-h270
+    Fixups for Fujitsu-Siemens Celsius H270
+fsc-s7110
+    Fixups for Fujitsu-Siemens Lifebook S7110
+hp-z200
+    Fixups for HP Z200
+tyan
+    Fixups for Tyan Thunder n6650W
+lenovo-3000
+    Fixups for Lenovo 3000
+benq
+    Fixups for Benq ED8
+benq-t31
+    Fixups for Benq T31
+bayleybay
+    Fixups for Intel BayleyBay
 
 ALC267/268
 ==========
@@ -41,6 +57,8 @@ inv-dmic
     Inverted internal mic workaround
 hp-eapd
     Disable HP EAPD on NID 0x15
+spdif
+    Enable SPDIF output on NID 0x1e
 
 ALC22x/23x/25x/269/27x/28x/29x (and vendor-specific ALC3xxx models)
 ===================================================================
@@ -70,6 +88,10 @@ dell-headset-multi
     Headset jack, which can also be used as mic-in
 dell-headset-dock
     Headset jack (without mic-in), and also dock I/O
+dell-headset3
+    Headset jack (without mic-in), and also dock I/O, variant 3
+dell-headset4
+    Headset jack (without mic-in), and also dock I/O, variant 4
 alc283-dac-wcaps
     Fixups for Chromebook with ALC283
 alc283-sense-combo
@@ -80,15 +102,173 @@ tpt440
     Lenovo Thinkpad T440s setup
 tpt460
     Lenovo Thinkpad T460/560 setup
+tpt470-dock
+    Lenovo Thinkpad T470 dock setup
 dual-codecs
     Lenovo laptops with dual codecs
 alc700-ref
     Intel reference board with ALC700 codec
+vaio
+    Pin fixups for Sony VAIO laptops
+dell-m101z
+    COEF setup for Dell M101z
+asus-g73jw
+    Subwoofer pin fixup for ASUS G73JW
+lenovo-eapd
+    Inversed EAPD setup for Lenovo laptops
+sony-hweq
+    H/W EQ COEF setup for Sony laptops
+pcm44k
+    Fixed PCM 44kHz constraints (for buggy devices)
+lifebook
+    Dock pin fixups for Fujitsu Lifebook
+lifebook-extmic
+    Headset mic fixup for Fujitsu Lifebook
+lifebook-hp-pin
+    Headphone pin fixup for Fujitsu Lifebook
+lifebook-u7x7
+    Lifebook U7x7 fixups
+alc269vb-amic
+    ALC269VB analog mic pin fixups
+alc269vb-dmic
+    ALC269VB digital mic pin fixups
+hp-mute-led-mic1
+    Mute LED via Mic1 pin on HP
+hp-mute-led-mic2
+    Mute LED via Mic2 pin on HP
+hp-mute-led-mic3
+    Mute LED via Mic3 pin on HP
+hp-gpio-mic1
+    GPIO + Mic1 pin LED on HP
+hp-line1-mic1
+    Mute LED via Line1 + Mic1 pins on HP
+noshutup
+    Skip shutup callback
+sony-nomic
+    Headset mic fixup for Sony laptops
+aspire-headset-mic
+    Headset pin fixup for Acer Aspire
+asus-x101
+    ASUS X101 fixups
+acer-ao7xx
+    Acer AO7xx fixups
+acer-aspire-e1
+    Acer Aspire E1 fixups
+acer-ac700
+    Acer AC700 fixups
+limit-mic-boost
+    Limit internal mic boost on Lenovo machines
+asus-zenbook
+    ASUS Zenbook fixups
+asus-zenbook-ux31a
+    ASUS Zenbook UX31A fixups
+ordissimo
+    Ordissimo EVE2 (or Malata PC-B1303) fixups
+asus-tx300
+    ASUS TX300 fixups
+alc283-int-mic
+    ALC283 COEF setup for Lenovo machines
+mono-speakers
+    Subwoofer and headset fixupes for Dell Inspiron
+alc290-subwoofer
+    Subwoofer fixups for Dell Vostro
+thinkpad
+    Binding with thinkpad_acpi driver for Lenovo machines
+dmic-thinkpad
+    thinkpad_acpi binding + digital mic support
+alc255-acer
+    ALC255 fixups on Acer machines
+alc255-asus
+    ALC255 fixups on ASUS machines
+alc255-dell1
+    ALC255 fixups on Dell machines
+alc255-dell2
+    ALC255 fixups on Dell machines, variant 2
+alc293-dell1
+    ALC293 fixups on Dell machines
+alc283-headset
+    Headset pin fixups on ALC283
+aspire-v5
+    Acer Aspire V5 fixups
+hp-gpio4
+    GPIO and Mic1 pin mute LED fixups for HP
+hp-gpio-led
+    GPIO mute LEDs on HP
+hp-gpio2-hotkey
+    GPIO mute LED with hot key handling on HP
+hp-dock-pins
+    GPIO mute LEDs and dock support on HP
+hp-dock-gpio-mic
+    GPIO, Mic mute LED and dock support on HP
+hp-9480m
+    HP 9480m fixups
+alc288-dell1
+    ALC288 fixups on Dell machines
+alc288-dell-xps13
+    ALC288 fixups on Dell XPS13
+dell-e7x
+    Dell E7x fixups
+alc293-dell
+    ALC293 fixups on Dell machines
+alc298-dell1
+    ALC298 fixups on Dell machines
+alc298-dell-aio
+    ALC298 fixups on Dell AIO machines
+alc275-dell-xps
+    ALC275 fixups on Dell XPS models
+alc256-dell-xps13
+    ALC256 fixups on Dell XPS13
+lenovo-spk-noise
+    Workaround for speaker noise on Lenovo machines
+lenovo-hotkey
+    Hot-key support via Mic2 pin on Lenovo machines
+dell-spk-noise
+    Workaround for speaker noise on Dell machines
+alc255-dell1
+    ALC255 fixups on Dell machines
+alc295-disable-dac3
+    Disable DAC3 routing on ALC295
+alc280-hp-headset
+    HP Elitebook fixups
+alc221-hp-mic
+    Front mic pin fixup on HP machines
+alc298-spk-volume
+    Speaker pin routing workaround on ALC298
+dell-inspiron-7559
+    Dell Inspiron 7559 fixups
+ativ-book
+    Samsung Ativ book 8 fixups
+alc221-hp-mic
+    ALC221 headset fixups on HP machines
+alc256-asus-mic
+    ALC256 fixups on ASUS machines
+alc256-asus-aio
+    ALC256 fixups on ASUS AIO machines
+alc233-eapd
+    ALC233 fixups on ASUS machines
+alc294-lenovo-mic
+    ALC294 Mic pin fixup for Lenovo AIO machines
+alc225-wyse
+    Dell Wyse fixups
+alc274-dell-aio
+    ALC274 fixups on Dell AIO machines
+alc255-dummy-lineout
+    Dell Precision 3930 fixups
+alc255-dell-headset"},
+    Dell Precision 3630 fixups
+alc295-hp-x360
+    HP Spectre X360 fixups
 
 ALC66x/67x/892
 ==============
+aspire
+    Subwoofer pin fixup for Aspire laptops
+ideapad
+    Subwoofer pin fixup for Ideapad laptops
 mario
     Chromebook mario model fixup
+hp-rp5800
+    Headphone pin fixup for HP RP5800
 asus-mode1
     ASUS
 asus-mode2
@@ -105,10 +285,40 @@ asus-mode7
     ASUS
 asus-mode8
     ASUS
+zotac-z68
+    Front HP fixup for Zotac Z68
 inv-dmic
     Inverted internal mic workaround
+alc662-headset-multi
+    Dell headset jack, which can also be used as mic-in (ALC662)
 dell-headset-multi
     Headset jack, which can also be used as mic-in
+alc662-headset
+    Headset mode support on ALC662
+alc668-headset
+    Headset mode support on ALC668
+bass16
+    Bass speaker fixup on pin 0x16
+bass1a
+    Bass speaker fixup on pin 0x1a
+automute
+    Auto-mute fixups for ALC668
+dell-xps13
+    Dell XPS13 fixups
+asus-nx50
+    ASUS Nx50 fixups
+asus-nx51
+    ASUS Nx51 fixups
+alc891-headset
+    Headset mode support on ALC891
+alc891-headset-multi
+    Dell headset jack, which can also be used as mic-in (ALC891)
+acer-veriton
+    Acer Veriton speaker pin fixup
+asrock-mobo
+    Fix invalid 0x15 / 0x16 pins
+usi-headset
+    Headset support on USI machines
 dual-codecs
     Lenovo laptops with dual codecs
 
@@ -116,20 +326,70 @@ ALC680
 ======
 N/A
 
-ALC88x/898/1150
-======================
+ALC88x/898/1150/1220
+====================
+abit-aw9d
+    Pin fixups for Abit AW9D-MAX
+lenovo-y530
+    Pin fixups for Lenovo Y530
+acer-aspire-7736
+    Fixup for Acer Aspire 7736
+asus-w90v
+    Pin fixup for ASUS W90V
+cd
+    Enable audio CD pin NID 0x1c
+no-front-hp
+    Disable front HP pin NID 0x1b
+vaio-tt
+    Pin fixup for VAIO TT
+eee1601
+    COEF setups for ASUS Eee 1601
+alc882-eapd
+    Change EAPD COEF mode on ALC882
+alc883-eapd
+    Change EAPD COEF mode on ALC883
+gpio1
+    Enable GPIO1
+gpio2
+    Enable GPIO2
+gpio3
+    Enable GPIO3
+alc889-coef
+    Setup ALC889 COEF
+asus-w2jc
+    Fixups for ASUS W2JC
 acer-aspire-4930g
     Acer Aspire 4930G/5930G/6530G/6930G/7730G
 acer-aspire-8930g
     Acer Aspire 8330G/6935G
 acer-aspire
     Acer Aspire others
+macpro-gpio
+    GPIO setup for Mac Pro
+dac-route
+    Workaround for DAC routing on Acer Aspire
+mbp-vref
+    Vref setup for Macbook Pro
+imac91-vref
+    Vref setup for iMac 9,1
+mba11-vref
+    Vref setup for MacBook Air 1,1
+mba21-vref
+    Vref setup for MacBook Air 2,1
+mp11-vref
+    Vref setup for Mac Pro 1,1
+mp41-vref
+    Vref setup for Mac Pro 4,1
 inv-dmic
     Inverted internal mic workaround
 no-primary-hp
     VAIO Z/VGC-LN51JGB workaround (for fixed speaker DAC)
+asus-bass
+    Bass speaker setup for ASUS ET2700
 dual-codecs
     ALC1220 dual codecs for Gaming mobos
+clevo-p950
+    Fixups for Clevo P950
 
 ALC861/660
 ==========

Documentation/sound/soc/dpcm.rst

@@ -254,9 +254,7 @@ configuration.
 	channels->min = channels->max = 2;
 
 	/* set DAI0 to 16 bit */
-	snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT -
-				    SNDRV_PCM_HW_PARAM_FIRST_MASK],
-				    SNDRV_PCM_FORMAT_S16_LE);
+	params_set_format(params, SNDRV_PCM_FORMAT_S16_LE);
 	return 0;
   }
 

MAINTAINERS

@@ -13585,6 +13585,13 @@ L:	linux-block@vger.kernel.org
 S:	Maintained
 F:	drivers/block/skd*[ch]
 
+STI AUDIO (ASoC) DRIVERS
+M:	Arnaud Pouliquen <arnaud.pouliquen@st.com>
+L:	alsa-devel@alsa-project.org (moderated for non-subscribers)
+S:	Maintained
+F:	Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt
+F:	sound/soc/sti/
+
 STI CEC DRIVER
 M:	Benjamin Gaignard <benjamin.gaignard@linaro.org>
 S:	Maintained
@@ -13598,6 +13605,14 @@ T:	git git://linuxtv.org/media_tree.git
 S:	Maintained
 F:	drivers/media/usb/stk1160/
 
+STM32 AUDIO (ASoC) DRIVERS
+M:	Olivier Moysan <olivier.moysan@st.com>
+M:	Arnaud Pouliquen <arnaud.pouliquen@st.com>
+L:	alsa-devel@alsa-project.org (moderated for non-subscribers)
+S:	Maintained
+F:	Documentation/devicetree/bindings/sound/st,stm32-*.txt
+F:	sound/soc/stm/
+
 STM32 TIMER/LPTIMER DRIVERS
 M:	Fabrice Gasnier <fabrice.gasnier@st.com>
 S:	Maintained

arch/arm/mach-pxa/devices.c

@@ -4,6 +4,7 @@
 #include <linux/init.h>
 #include <linux/platform_device.h>
 #include <linux/dma-mapping.h>
+#include <linux/dmaengine.h>
 #include <linux/spi/pxa2xx_spi.h>
 #include <linux/platform_data/i2c-pxa.h>
 
@@ -59,16 +60,6 @@ static struct resource pxamci_resources[] = {
 		.end	= IRQ_MMC,
 		.flags	= IORESOURCE_IRQ,
 	},
-	[2] = {
-		.start	= 21,
-		.end	= 21,
-		.flags	= IORESOURCE_DMA,
-	},
-	[3] = {
-		.start	= 22,
-		.end	= 22,
-		.flags	= IORESOURCE_DMA,
-	},
 };
 
 static u64 pxamci_dmamask = 0xffffffffUL;
@@ -406,16 +397,6 @@ static struct resource pxa_ir_resources[] = {
 		.end	= 0x40700023,
 		.flags  = IORESOURCE_MEM,
 	},
-	[5] = {
-		.start  = 17,
-		.end	= 17,
-		.flags  = IORESOURCE_DMA,
-	},
-	[6] = {
-		.start  = 18,
-		.end	= 18,
-		.flags  = IORESOURCE_DMA,
-	},
 };
 
 struct platform_device pxa_device_ficp = {
@@ -544,18 +525,6 @@ static struct resource pxa25x_resource_ssp[] = {
 		.end	= IRQ_SSP,
 		.flags	= IORESOURCE_IRQ,
 	},
-	[2] = {
-		/* DRCMR for RX */
-		.start	= 13,
-		.end	= 13,
-		.flags	= IORESOURCE_DMA,
-	},
-	[3] = {
-		/* DRCMR for TX */
-		.start	= 14,
-		.end	= 14,
-		.flags	= IORESOURCE_DMA,
-	},
 };
 
 struct platform_device pxa25x_device_ssp = {
@@ -582,18 +551,6 @@ static struct resource pxa25x_resource_nssp[] = {
 		.end	= IRQ_NSSP,
 		.flags	= IORESOURCE_IRQ,
 	},
-	[2] = {
-		/* DRCMR for RX */
-		.start	= 15,
-		.end	= 15,
-		.flags	= IORESOURCE_DMA,
-	},
-	[3] = {
-		/* DRCMR for TX */
-		.start	= 16,
-		.end	= 16,
-		.flags	= IORESOURCE_DMA,
-	},
 };
 
 struct platform_device pxa25x_device_nssp = {
@@ -620,18 +577,6 @@ static struct resource pxa25x_resource_assp[] = {
 		.end	= IRQ_ASSP,
 		.flags	= IORESOURCE_IRQ,
 	},
-	[2] = {
-		/* DRCMR for RX */
-		.start	= 23,
-		.end	= 23,
-		.flags	= IORESOURCE_DMA,
-	},
-	[3] = {
-		/* DRCMR for TX */
-		.start	= 24,
-		.end	= 24,
-		.flags	= IORESOURCE_DMA,
-	},
 };
 
 struct platform_device pxa25x_device_assp = {
@@ -750,18 +695,6 @@ static struct resource pxa27x_resource_ssp1[] = {
 		.end	= IRQ_SSP,
 		.flags	= IORESOURCE_IRQ,
 	},
-	[2] = {
-		/* DRCMR for RX */
-		.start	= 13,
-		.end	= 13,
-		.flags	= IORESOURCE_DMA,
-	},
-	[3] = {
-		/* DRCMR for TX */
-		.start	= 14,
-		.end	= 14,
-		.flags	= IORESOURCE_DMA,
-	},
 };
 
 struct platform_device pxa27x_device_ssp1 = {
@@ -788,18 +721,6 @@ static struct resource pxa27x_resource_ssp2[] = {
 		.end	= IRQ_SSP2,
 		.flags	= IORESOURCE_IRQ,
 	},
-	[2] = {
-		/* DRCMR for RX */
-		.start	= 15,
-		.end	= 15,
-		.flags	= IORESOURCE_DMA,
-	},
-	[3] = {
-		/* DRCMR for TX */
-		.start	= 16,
-		.end	= 16,
-		.flags	= IORESOURCE_DMA,
-	},
 };
 
 struct platform_device pxa27x_device_ssp2 = {
@@ -826,18 +747,6 @@ static struct resource pxa27x_resource_ssp3[] = {
 		.end	= IRQ_SSP3,
 		.flags	= IORESOURCE_IRQ,
 	},
-	[2] = {
-		/* DRCMR for RX */
-		.start	= 66,
-		.end	= 66,
-		.flags	= IORESOURCE_DMA,
-	},
-	[3] = {
-		/* DRCMR for TX */
-		.start	= 67,
-		.end	= 67,
-		.flags	= IORESOURCE_DMA,
-	},
 };
 
 struct platform_device pxa27x_device_ssp3 = {
@@ -894,16 +803,6 @@ static struct resource pxa3xx_resources_mci2[] = {
 		.end	= IRQ_MMC2,
 		.flags	= IORESOURCE_IRQ,
 	},
-	[2] = {
-		.start	= 93,
-		.end	= 93,
-		.flags	= IORESOURCE_DMA,
-	},
-	[3] = {
-		.start	= 94,
-		.end	= 94,
-		.flags	= IORESOURCE_DMA,
-	},
 };
 
 struct platform_device pxa3xx_device_mci2 = {
@@ -933,16 +832,6 @@ static struct resource pxa3xx_resources_mci3[] = {
 		.end	= IRQ_MMC3,
 		.flags	= IORESOURCE_IRQ,
 	},
-	[2] = {
-		.start	= 100,
-		.end	= 100,
-		.flags	= IORESOURCE_DMA,
-	},
-	[3] = {
-		.start	= 101,
-		.end	= 101,
-		.flags	= IORESOURCE_DMA,
-	},
 };
 
 struct platform_device pxa3xx_device_mci3 = {
@@ -1020,18 +909,6 @@ static struct resource pxa3xx_resources_nand[] = {
 		.end	= IRQ_NAND,
 		.flags	= IORESOURCE_IRQ,
 	},
-	[2] = {
-		/* DRCMR for Data DMA */
-		.start	= 97,
-		.end	= 97,
-		.flags	= IORESOURCE_DMA,
-	},
-	[3] = {
-		/* DRCMR for Command DMA */
-		.start	= 99,
-		.end	= 99,
-		.flags	= IORESOURCE_DMA,
-	},
 };
 
 static u64 pxa3xx_nand_dma_mask = DMA_BIT_MASK(32);
@@ -1065,18 +942,6 @@ static struct resource pxa3xx_resource_ssp4[] = {
 		.end	= IRQ_SSP4,
 		.flags	= IORESOURCE_IRQ,
 	},
-	[2] = {
-		/* DRCMR for RX */
-		.start	= 2,
-		.end	= 2,
-		.flags	= IORESOURCE_DMA,
-	},
-	[3] = {
-		/* DRCMR for TX */
-		.start	= 3,
-		.end	= 3,
-		.flags	= IORESOURCE_DMA,
-	},
 };
 
 /*
@@ -1202,11 +1067,6 @@ void __init pxa2xx_set_spi_info(unsigned id, struct pxa2xx_spi_master *info)
 	platform_device_add(pd);
 }
 
-static struct mmp_dma_platdata pxa_dma_pdata = {
-	.dma_channels	= 0,
-	.nb_requestors	= 0,
-};
-
 static struct resource pxa_dma_resource[] = {
 	[0] = {
 		.start	= 0x40000000,
@@ -1233,9 +1093,7 @@ static struct platform_device pxa2xx_pxa_dma = {
 	.resource	= pxa_dma_resource,
 };
 
-void __init pxa2xx_set_dmac_info(int nb_channels, int nb_requestors)
+void __init pxa2xx_set_dmac_info(struct mmp_dma_platdata *dma_pdata)
 {
-	pxa_dma_pdata.dma_channels = nb_channels;
-	pxa_dma_pdata.nb_requestors = nb_requestors;
-	pxa_register_device(&pxa2xx_pxa_dma, &pxa_dma_pdata);
+	pxa_register_device(&pxa2xx_pxa_dma, dma_pdata);
 }

arch/arm/mach-pxa/devices.h

@@ -1,4 +1,8 @@
 /* SPDX-License-Identifier: GPL-2.0 */
+#define PDMA_FILTER_PARAM(_prio, _requestor) (&(struct pxad_param) { \
+	.prio = PXAD_PRIO_##_prio, .drcmr = _requestor })
+struct mmp_dma_platdata;
+
 extern struct platform_device pxa_device_pmu;
 extern struct platform_device pxa_device_mci;
 extern struct platform_device pxa3xx_device_mci2;
@@ -55,7 +59,7 @@ extern struct platform_device pxa3xx_device_gpio;
 extern struct platform_device pxa93x_device_gpio;
 
 void __init pxa_register_device(struct platform_device *dev, void *data);
-void __init pxa2xx_set_dmac_info(int nb_channels, int nb_requestors);
+void __init pxa2xx_set_dmac_info(struct mmp_dma_platdata *dma_pdata);
 
 struct i2c_pxa_platform_data;
 extern void pxa_set_i2c_info(struct i2c_pxa_platform_data *info);

arch/arm/mach-pxa/pxa25x.c

@@ -16,6 +16,8 @@
  * initialization stuff for PXA machines which can be overridden later if
  * need be.
  */
+#include <linux/dmaengine.h>
+#include <linux/dma/pxa-dma.h>
 #include <linux/gpio.h>
 #include <linux/gpio-pxa.h>
 #include <linux/module.h>
@@ -26,6 +28,7 @@
 #include <linux/syscore_ops.h>
 #include <linux/irq.h>
 #include <linux/irqchip.h>
+#include <linux/platform_data/mmp_dma.h>
 
 #include <asm/mach/map.h>
 #include <asm/suspend.h>
@@ -201,6 +204,39 @@ static struct platform_device *pxa25x_devices[] __initdata = {
 	&pxa_device_asoc_platform,
 };
 
+static const struct dma_slave_map pxa25x_slave_map[] = {
+	/* PXA25x, PXA27x and PXA3xx common entries */
+	{ "pxa2xx-ac97", "pcm_pcm_mic_mono", PDMA_FILTER_PARAM(LOWEST, 8) },
+	{ "pxa2xx-ac97", "pcm_pcm_aux_mono_in", PDMA_FILTER_PARAM(LOWEST, 9) },
+	{ "pxa2xx-ac97", "pcm_pcm_aux_mono_out",
+	  PDMA_FILTER_PARAM(LOWEST, 10) },
+	{ "pxa2xx-ac97", "pcm_pcm_stereo_in", PDMA_FILTER_PARAM(LOWEST, 11) },
+	{ "pxa2xx-ac97", "pcm_pcm_stereo_out", PDMA_FILTER_PARAM(LOWEST, 12) },
+	{ "pxa-ssp-dai.1", "rx", PDMA_FILTER_PARAM(LOWEST, 13) },
+	{ "pxa-ssp-dai.1", "tx", PDMA_FILTER_PARAM(LOWEST, 14) },
+	{ "pxa-ssp-dai.2", "rx", PDMA_FILTER_PARAM(LOWEST, 15) },
+	{ "pxa-ssp-dai.2", "tx", PDMA_FILTER_PARAM(LOWEST, 16) },
+	{ "pxa2xx-ir", "rx", PDMA_FILTER_PARAM(LOWEST, 17) },
+	{ "pxa2xx-ir", "tx", PDMA_FILTER_PARAM(LOWEST, 18) },
+	{ "pxa2xx-mci.0", "rx", PDMA_FILTER_PARAM(LOWEST, 21) },
+	{ "pxa2xx-mci.0", "tx", PDMA_FILTER_PARAM(LOWEST, 22) },
+
+	/* PXA25x specific map */
+	{ "pxa25x-ssp.0", "rx", PDMA_FILTER_PARAM(LOWEST, 13) },
+	{ "pxa25x-ssp.0", "tx", PDMA_FILTER_PARAM(LOWEST, 14) },
+	{ "pxa25x-nssp.1", "rx", PDMA_FILTER_PARAM(LOWEST, 15) },
+	{ "pxa25x-nssp.1", "tx", PDMA_FILTER_PARAM(LOWEST, 16) },
+	{ "pxa25x-nssp.2", "rx", PDMA_FILTER_PARAM(LOWEST, 23) },
+	{ "pxa25x-nssp.2", "tx", PDMA_FILTER_PARAM(LOWEST, 24) },
+};
+
+static struct mmp_dma_platdata pxa25x_dma_pdata = {
+	.dma_channels	= 16,
+	.nb_requestors	= 40,
+	.slave_map	= pxa25x_slave_map,
+	.slave_map_cnt	= ARRAY_SIZE(pxa25x_slave_map),
+};
+
 static int __init pxa25x_init(void)
 {
 	int ret = 0;
@@ -215,7 +251,7 @@ static int __init pxa25x_init(void)
 		register_syscore_ops(&pxa2xx_mfp_syscore_ops);
 
 		if (!of_have_populated_dt()) {
-			pxa2xx_set_dmac_info(16, 40);
+			pxa2xx_set_dmac_info(&pxa25x_dma_pdata);
 			pxa_register_device(&pxa25x_device_gpio, &pxa25x_gpio_info);
 			ret = platform_add_devices(pxa25x_devices,
 						   ARRAY_SIZE(pxa25x_devices));

arch/arm/mach-pxa/pxa27x.c

@@ -11,6 +11,8 @@
  * it under the terms of the GNU General Public License version 2 as
  * published by the Free Software Foundation.
  */
+#include <linux/dmaengine.h>
+#include <linux/dma/pxa-dma.h>
 #include <linux/gpio.h>
 #include <linux/gpio-pxa.h>
 #include <linux/module.h>
@@ -23,6 +25,7 @@
 #include <linux/io.h>
 #include <linux/irq.h>
 #include <linux/platform_data/i2c-pxa.h>
+#include <linux/platform_data/mmp_dma.h>
 
 #include <asm/mach/map.h>
 #include <mach/hardware.h>
@@ -297,6 +300,40 @@ static struct platform_device *devices[] __initdata = {
 	&pxa27x_device_pwm1,
 };
 
+static const struct dma_slave_map pxa27x_slave_map[] = {
+	/* PXA25x, PXA27x and PXA3xx common entries */
+	{ "pxa2xx-ac97", "pcm_pcm_mic_mono", PDMA_FILTER_PARAM(LOWEST, 8) },
+	{ "pxa2xx-ac97", "pcm_pcm_aux_mono_in", PDMA_FILTER_PARAM(LOWEST, 9) },
+	{ "pxa2xx-ac97", "pcm_pcm_aux_mono_out",
+	  PDMA_FILTER_PARAM(LOWEST, 10) },
+	{ "pxa2xx-ac97", "pcm_pcm_stereo_in", PDMA_FILTER_PARAM(LOWEST, 11) },
+	{ "pxa2xx-ac97", "pcm_pcm_stereo_out", PDMA_FILTER_PARAM(LOWEST, 12) },
+	{ "pxa-ssp-dai.0", "rx", PDMA_FILTER_PARAM(LOWEST, 13) },
+	{ "pxa-ssp-dai.0", "tx", PDMA_FILTER_PARAM(LOWEST, 14) },
+	{ "pxa-ssp-dai.1", "rx", PDMA_FILTER_PARAM(LOWEST, 15) },
+	{ "pxa-ssp-dai.1", "tx", PDMA_FILTER_PARAM(LOWEST, 16) },
+	{ "pxa2xx-ir", "rx", PDMA_FILTER_PARAM(LOWEST, 17) },
+	{ "pxa2xx-ir", "tx", PDMA_FILTER_PARAM(LOWEST, 18) },
+	{ "pxa2xx-mci.0", "rx", PDMA_FILTER_PARAM(LOWEST, 21) },
+	{ "pxa2xx-mci.0", "tx", PDMA_FILTER_PARAM(LOWEST, 22) },
+	{ "pxa-ssp-dai.2", "rx", PDMA_FILTER_PARAM(LOWEST, 66) },
+	{ "pxa-ssp-dai.2", "tx", PDMA_FILTER_PARAM(LOWEST, 67) },
+
+	/* PXA27x specific map */
+	{ "pxa2xx-i2s", "rx", PDMA_FILTER_PARAM(LOWEST, 2) },
+	{ "pxa2xx-i2s", "tx", PDMA_FILTER_PARAM(LOWEST, 3) },
+	{ "pxa27x-camera.0", "CI_Y", PDMA_FILTER_PARAM(HIGHEST, 68) },
+	{ "pxa27x-camera.0", "CI_U", PDMA_FILTER_PARAM(HIGHEST, 69) },
+	{ "pxa27x-camera.0", "CI_V", PDMA_FILTER_PARAM(HIGHEST, 70) },
+};
+
+static struct mmp_dma_platdata pxa27x_dma_pdata = {
+	.dma_channels	= 32,
+	.nb_requestors	= 75,
+	.slave_map	= pxa27x_slave_map,
+	.slave_map_cnt	= ARRAY_SIZE(pxa27x_slave_map),
+};
+
 static int __init pxa27x_init(void)
 {
 	int ret = 0;
@@ -313,7 +350,7 @@ static int __init pxa27x_init(void)
 		if (!of_have_populated_dt()) {
 			pxa_register_device(&pxa27x_device_gpio,
 					    &pxa27x_gpio_info);
-			pxa2xx_set_dmac_info(32, 75);
+			pxa2xx_set_dmac_info(&pxa27x_dma_pdata);
 			ret = platform_add_devices(devices,
 						   ARRAY_SIZE(devices));
 		}

arch/arm/mach-pxa/pxa3xx.c

@@ -12,6 +12,8 @@
  * it under the terms of the GNU General Public License version 2 as
  * published by the Free Software Foundation.
  */
+#include <linux/dmaengine.h>
+#include <linux/dma/pxa-dma.h>
 #include <linux/module.h>
 #include <linux/kernel.h>
 #include <linux/init.h>
@@ -24,6 +26,7 @@
 #include <linux/of.h>
 #include <linux/syscore_ops.h>
 #include <linux/platform_data/i2c-pxa.h>
+#include <linux/platform_data/mmp_dma.h>
 
 #include <asm/mach/map.h>
 #include <asm/suspend.h>
@@ -421,6 +424,42 @@ static struct platform_device *devices[] __initdata = {
 	&pxa27x_device_pwm1,
 };
 
+static const struct dma_slave_map pxa3xx_slave_map[] = {
+	/* PXA25x, PXA27x and PXA3xx common entries */
+	{ "pxa2xx-ac97", "pcm_pcm_mic_mono", PDMA_FILTER_PARAM(LOWEST, 8) },
+	{ "pxa2xx-ac97", "pcm_pcm_aux_mono_in", PDMA_FILTER_PARAM(LOWEST, 9) },
+	{ "pxa2xx-ac97", "pcm_pcm_aux_mono_out",
+	  PDMA_FILTER_PARAM(LOWEST, 10) },
+	{ "pxa2xx-ac97", "pcm_pcm_stereo_in", PDMA_FILTER_PARAM(LOWEST, 11) },
+	{ "pxa2xx-ac97", "pcm_pcm_stereo_out", PDMA_FILTER_PARAM(LOWEST, 12) },
+	{ "pxa-ssp-dai.0", "rx", PDMA_FILTER_PARAM(LOWEST, 13) },
+	{ "pxa-ssp-dai.0", "tx", PDMA_FILTER_PARAM(LOWEST, 14) },
+	{ "pxa-ssp-dai.1", "rx", PDMA_FILTER_PARAM(LOWEST, 15) },
+	{ "pxa-ssp-dai.1", "tx", PDMA_FILTER_PARAM(LOWEST, 16) },
+	{ "pxa2xx-ir", "rx", PDMA_FILTER_PARAM(LOWEST, 17) },
+	{ "pxa2xx-ir", "tx", PDMA_FILTER_PARAM(LOWEST, 18) },
+	{ "pxa2xx-mci.0", "rx", PDMA_FILTER_PARAM(LOWEST, 21) },
+	{ "pxa2xx-mci.0", "tx", PDMA_FILTER_PARAM(LOWEST, 22) },
+	{ "pxa-ssp-dai.2", "rx", PDMA_FILTER_PARAM(LOWEST, 66) },
+	{ "pxa-ssp-dai.2", "tx", PDMA_FILTER_PARAM(LOWEST, 67) },
+
+	/* PXA3xx specific map */
+	{ "pxa-ssp-dai.3", "rx", PDMA_FILTER_PARAM(LOWEST, 2) },
+	{ "pxa-ssp-dai.3", "tx", PDMA_FILTER_PARAM(LOWEST, 3) },
+	{ "pxa2xx-mci.1", "rx", PDMA_FILTER_PARAM(LOWEST, 93) },
+	{ "pxa2xx-mci.1", "tx", PDMA_FILTER_PARAM(LOWEST, 94) },
+	{ "pxa3xx-nand", "data", PDMA_FILTER_PARAM(LOWEST, 97) },
+	{ "pxa2xx-mci.2", "rx", PDMA_FILTER_PARAM(LOWEST, 100) },
+	{ "pxa2xx-mci.2", "tx", PDMA_FILTER_PARAM(LOWEST, 101) },
+};
+
+static struct mmp_dma_platdata pxa3xx_dma_pdata = {
+	.dma_channels	= 32,
+	.nb_requestors	= 100,
+	.slave_map	= pxa3xx_slave_map,
+	.slave_map_cnt	= ARRAY_SIZE(pxa3xx_slave_map),
+};
+
 static int __init pxa3xx_init(void)
 {
 	int ret = 0;
@@ -456,7 +495,7 @@ static int __init pxa3xx_init(void)
 		if (of_have_populated_dt())
 			return 0;
 
-		pxa2xx_set_dmac_info(32, 100);
+		pxa2xx_set_dmac_info(&pxa3xx_dma_pdata);
 		ret = platform_add_devices(devices, ARRAY_SIZE(devices));
 		if (ret)
 			return ret;

arch/arm/plat-pxa/ssp.c

@@ -127,53 +127,6 @@ static int pxa_ssp_probe(struct platform_device *pdev)
 	if (IS_ERR(ssp->clk))
 		return PTR_ERR(ssp->clk);
 
-	if (dev->of_node) {
-		struct of_phandle_args dma_spec;
-		struct device_node *np = dev->of_node;
-		int ret;
-
-		/*
-		 * FIXME: we should allocate the DMA channel from this
-		 * context and pass the channel down to the ssp users.
-		 * For now, we lookup the rx and tx indices manually
-		 */
-
-		/* rx */
-		ret = of_parse_phandle_with_args(np, "dmas", "#dma-cells",
-						 0, &dma_spec);
-
-		if (ret) {
-			dev_err(dev, "Can't parse dmas property\n");
-			return -ENODEV;
-		}
-		ssp->drcmr_rx = dma_spec.args[0];
-		of_node_put(dma_spec.np);
-
-		/* tx */
-		ret = of_parse_phandle_with_args(np, "dmas", "#dma-cells",
-						 1, &dma_spec);
-		if (ret) {
-			dev_err(dev, "Can't parse dmas property\n");
-			return -ENODEV;
-		}
-		ssp->drcmr_tx = dma_spec.args[0];
-		of_node_put(dma_spec.np);
-	} else {
-		res = platform_get_resource(pdev, IORESOURCE_DMA, 0);
-		if (res == NULL) {
-			dev_err(dev, "no SSP RX DRCMR defined\n");
-			return -ENODEV;
-		}
-		ssp->drcmr_rx = res->start;
-
-		res = platform_get_resource(pdev, IORESOURCE_DMA, 1);
-		if (res == NULL) {
-			dev_err(dev, "no SSP TX DRCMR defined\n");
-			return -ENODEV;
-		}
-		ssp->drcmr_tx = res->start;
-	}
-
 	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
 	if (res == NULL) {
 		dev_err(dev, "no memory resource defined\n");

drivers/ata/pata_pxa.c

@@ -25,7 +25,6 @@
 #include <linux/libata.h>
 #include <linux/platform_device.h>
 #include <linux/dmaengine.h>
-#include <linux/dma/pxa-dma.h>
 #include <linux/gpio.h>
 #include <linux/slab.h>
 #include <linux/completion.h>
@@ -180,8 +179,6 @@ static int pxa_ata_probe(struct platform_device *pdev)
 	struct resource *irq_res;
 	struct pata_pxa_pdata *pdata = dev_get_platdata(&pdev->dev);
 	struct dma_slave_config	config;
-	dma_cap_mask_t mask;
-	struct pxad_param param;
 	int ret = 0;
 
 	/*
@@ -278,10 +275,6 @@ static int pxa_ata_probe(struct platform_device *pdev)
 
 	ap->private_data = data;
 
-	dma_cap_zero(mask);
-	dma_cap_set(DMA_SLAVE, mask);
-	param.prio = PXAD_PRIO_LOWEST;
-	param.drcmr = pdata->dma_dreq;
 	memset(&config, 0, sizeof(config));
 	config.src_addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES;
 	config.dst_addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES;
@@ -294,8 +287,7 @@ static int pxa_ata_probe(struct platform_device *pdev)
 	 * Request the DMA channel
 	 */
 	data->dma_chan =
-		dma_request_slave_channel_compat(mask, pxad_filter_fn,
-						 &param, &pdev->dev, "data");
+		dma_request_slave_channel(&pdev->dev, "data");
 	if (!data->dma_chan)
 		return -EBUSY;
 	ret = dmaengine_slave_config(data->dma_chan, &config);

drivers/clk/clk.c

@@ -67,6 +67,7 @@ struct clk_core {
 	unsigned long		max_rate;
 	unsigned long		accuracy;
 	int			phase;
+	struct clk_duty		duty;
 	struct hlist_head	children;
 	struct hlist_node	child_node;
 	struct hlist_head	clks;
@@ -2401,6 +2402,172 @@ int clk_get_phase(struct clk *clk)
 }
 EXPORT_SYMBOL_GPL(clk_get_phase);
 
+static void clk_core_reset_duty_cycle_nolock(struct clk_core *core)
+{
+	/* Assume a default value of 50% */
+	core->duty.num = 1;
+	core->duty.den = 2;
+}
+
+static int clk_core_update_duty_cycle_parent_nolock(struct clk_core *core);
+
+static int clk_core_update_duty_cycle_nolock(struct clk_core *core)
+{
+	struct clk_duty *duty = &core->duty;
+	int ret = 0;
+
+	if (!core->ops->get_duty_cycle)
+		return clk_core_update_duty_cycle_parent_nolock(core);
+
+	ret = core->ops->get_duty_cycle(core->hw, duty);
+	if (ret)
+		goto reset;
+
+	/* Don't trust the clock provider too much */
+	if (duty->den == 0 || duty->num > duty->den) {
+		ret = -EINVAL;
+		goto reset;
+	}
+
+	return 0;
+
+reset:
+	clk_core_reset_duty_cycle_nolock(core);
+	return ret;
+}
+
+static int clk_core_update_duty_cycle_parent_nolock(struct clk_core *core)
+{
+	int ret = 0;
+
+	if (core->parent &&
+	    core->flags & CLK_DUTY_CYCLE_PARENT) {
+		ret = clk_core_update_duty_cycle_nolock(core->parent);
+		memcpy(&core->duty, &core->parent->duty, sizeof(core->duty));
+	} else {
+		clk_core_reset_duty_cycle_nolock(core);
+	}
+
+	return ret;
+}
+
+static int clk_core_set_duty_cycle_parent_nolock(struct clk_core *core,
+						 struct clk_duty *duty);
+
+static int clk_core_set_duty_cycle_nolock(struct clk_core *core,
+					  struct clk_duty *duty)
+{
+	int ret;
+
+	lockdep_assert_held(&prepare_lock);
+
+	if (clk_core_rate_is_protected(core))
+		return -EBUSY;
+
+	trace_clk_set_duty_cycle(core, duty);
+
+	if (!core->ops->set_duty_cycle)
+		return clk_core_set_duty_cycle_parent_nolock(core, duty);
+
+	ret = core->ops->set_duty_cycle(core->hw, duty);
+	if (!ret)
+		memcpy(&core->duty, duty, sizeof(*duty));
+
+	trace_clk_set_duty_cycle_complete(core, duty);
+
+	return ret;
+}
+
+static int clk_core_set_duty_cycle_parent_nolock(struct clk_core *core,
+						 struct clk_duty *duty)
+{
+	int ret = 0;
+
+	if (core->parent &&
+	    core->flags & (CLK_DUTY_CYCLE_PARENT | CLK_SET_RATE_PARENT)) {
+		ret = clk_core_set_duty_cycle_nolock(core->parent, duty);
+		memcpy(&core->duty, &core->parent->duty, sizeof(core->duty));
+	}
+
+	return ret;
+}
+
+/**
+ * clk_set_duty_cycle - adjust the duty cycle ratio of a clock signal
+ * @clk: clock signal source
+ * @num: numerator of the duty cycle ratio to be applied
+ * @den: denominator of the duty cycle ratio to be applied
+ *
+ * Apply the duty cycle ratio if the ratio is valid and the clock can
+ * perform this operation
+ *
+ * Returns (0) on success, a negative errno otherwise.
+ */
+int clk_set_duty_cycle(struct clk *clk, unsigned int num, unsigned int den)
+{
+	int ret;
+	struct clk_duty duty;
+
+	if (!clk)
+		return 0;
+
+	/* sanity check the ratio */
+	if (den == 0 || num > den)
+		return -EINVAL;
+
+	duty.num = num;
+	duty.den = den;
+
+	clk_prepare_lock();
+
+	if (clk->exclusive_count)
+		clk_core_rate_unprotect(clk->core);
+
+	ret = clk_core_set_duty_cycle_nolock(clk->core, &duty);
+
+	if (clk->exclusive_count)
+		clk_core_rate_protect(clk->core);
+
+	clk_prepare_unlock();
+
+	return ret;
+}
+EXPORT_SYMBOL_GPL(clk_set_duty_cycle);
+
+static int clk_core_get_scaled_duty_cycle(struct clk_core *core,
+					  unsigned int scale)
+{
+	struct clk_duty *duty = &core->duty;
+	int ret;
+
+	clk_prepare_lock();
+
+	ret = clk_core_update_duty_cycle_nolock(core);
+	if (!ret)
+		ret = mult_frac(scale, duty->num, duty->den);
+
+	clk_prepare_unlock();
+
+	return ret;
+}
+
+/**
+ * clk_get_scaled_duty_cycle - return the duty cycle ratio of a clock signal
+ * @clk: clock signal source
+ * @scale: scaling factor to be applied to represent the ratio as an integer
+ *
+ * Returns the duty cycle ratio of a clock node multiplied by the provided
+ * scaling factor, or negative errno on error.
+ */
+int clk_get_scaled_duty_cycle(struct clk *clk, unsigned int scale)
+{
+	if (!clk)
+		return 0;
+
+	return clk_core_get_scaled_duty_cycle(clk->core, scale);
+}
+EXPORT_SYMBOL_GPL(clk_get_scaled_duty_cycle);
+
 /**
  * clk_is_match - check if two clk's point to the same hardware clock
  * @p: clk compared against q
@@ -2454,12 +2621,13 @@ static void clk_summary_show_one(struct seq_file *s, struct clk_core *c,
 	if (!c)
 		return;
 
-	seq_printf(s, "%*s%-*s %7d %8d %8d %11lu %10lu %-3d\n",
+	seq_printf(s, "%*s%-*s %7d %8d %8d %11lu %10lu %5d %6d\n",
 		   level * 3 + 1, "",
 		   30 - level * 3, c->name,
 		   c->enable_count, c->prepare_count, c->protect_count,
 		   clk_core_get_rate(c), clk_core_get_accuracy(c),
-		   clk_core_get_phase(c));
+		   clk_core_get_phase(c),
+		   clk_core_get_scaled_duty_cycle(c, 100000));
 }
 
 static void clk_summary_show_subtree(struct seq_file *s, struct clk_core *c,
@@ -2481,9 +2649,9 @@ static int clk_summary_show(struct seq_file *s, void *data)
 	struct clk_core *c;
 	struct hlist_head **lists = (struct hlist_head **)s->private;
 
-	seq_puts(s, "                                 enable  prepare  protect                               \n");
-	seq_puts(s, "   clock                          count    count    count        rate   accuracy   phase\n");
-	seq_puts(s, "----------------------------------------------------------------------------------------\n");
+	seq_puts(s, "                                 enable  prepare  protect                                duty\n");
+	seq_puts(s, "   clock                          count    count    count        rate   accuracy phase  cycle\n");
+	seq_puts(s, "---------------------------------------------------------------------------------------------\n");
 
 	clk_prepare_lock();
 
@@ -2510,6 +2678,8 @@ static void clk_dump_one(struct seq_file *s, struct clk_core *c, int level)
 	seq_printf(s, "\"rate\": %lu,", clk_core_get_rate(c));
 	seq_printf(s, "\"accuracy\": %lu,", clk_core_get_accuracy(c));
 	seq_printf(s, "\"phase\": %d", clk_core_get_phase(c));
+	seq_printf(s, "\"duty_cycle\": %u",
+		   clk_core_get_scaled_duty_cycle(c, 100000));
 }
 
 static void clk_dump_subtree(struct seq_file *s, struct clk_core *c, int level)
@@ -2571,6 +2741,7 @@ static const struct {
 	ENTRY(CLK_SET_RATE_UNGATE),
 	ENTRY(CLK_IS_CRITICAL),
 	ENTRY(CLK_OPS_PARENT_ENABLE),
+	ENTRY(CLK_DUTY_CYCLE_PARENT),
 #undef ENTRY
 };
 
@@ -2609,6 +2780,17 @@ static int possible_parents_show(struct seq_file *s, void *data)
 }
 DEFINE_SHOW_ATTRIBUTE(possible_parents);
 
+static int clk_duty_cycle_show(struct seq_file *s, void *data)
+{
+	struct clk_core *core = s->private;
+	struct clk_duty *duty = &core->duty;
+
+	seq_printf(s, "%u/%u\n", duty->num, duty->den);
+
+	return 0;
+}
+DEFINE_SHOW_ATTRIBUTE(clk_duty_cycle);
+
 static void clk_debug_create_one(struct clk_core *core, struct dentry *pdentry)
 {
 	struct dentry *root;
@@ -2627,6 +2809,8 @@ static void clk_debug_create_one(struct clk_core *core, struct dentry *pdentry)
 	debugfs_create_u32("clk_enable_count", 0444, root, &core->enable_count);
 	debugfs_create_u32("clk_protect_count", 0444, root, &core->protect_count);
 	debugfs_create_u32("clk_notifier_count", 0444, root, &core->notifier_count);
+	debugfs_create_file("clk_duty_cycle", 0444, root, core,
+			    &clk_duty_cycle_fops);
 
 	if (core->num_parents > 1)
 		debugfs_create_file("clk_possible_parents", 0444, root, core,
@@ -2844,6 +3028,11 @@ static int __clk_core_init(struct clk_core *core)
 	else
 		core->phase = 0;
 
+	/*
+	 * Set clk's duty cycle.
+	 */
+	clk_core_update_duty_cycle_nolock(core);
+
 	/*
 	 * Set clk's rate.  The preferred method is to use .recalc_rate.  For
 	 * simple clocks and lazy developers the default fallback is to use the

drivers/dma/pxa_dma.c

@@ -179,6 +179,8 @@ static unsigned int pxad_drcmr(unsigned int line)
 	return 0x1000 + line * 4;
 }
 
+bool pxad_filter_fn(struct dma_chan *chan, void *param);
+
 /*
  * Debug fs
  */
@@ -760,6 +762,8 @@ static void pxad_free_chan_resources(struct dma_chan *dchan)
 	dma_pool_destroy(chan->desc_pool);
 	chan->desc_pool = NULL;
 
+	chan->drcmr = U32_MAX;
+	chan->prio = PXAD_PRIO_LOWEST;
 }
 
 static void pxad_free_desc(struct virt_dma_desc *vd)
@@ -1384,6 +1388,9 @@ static int pxad_init_dmadev(struct platform_device *op,
 		c = devm_kzalloc(&op->dev, sizeof(*c), GFP_KERNEL);
 		if (!c)
 			return -ENOMEM;
+
+		c->drcmr = U32_MAX;
+		c->prio = PXAD_PRIO_LOWEST;
 		c->vc.desc_free = pxad_free_desc;
 		vchan_init(&c->vc, &pdev->slave);
 		init_waitqueue_head(&c->wq_state);
@@ -1396,9 +1403,10 @@ static int pxad_probe(struct platform_device *op)
 {
 	struct pxad_device *pdev;
 	const struct of_device_id *of_id;
+	const struct dma_slave_map *slave_map = NULL;
 	struct mmp_dma_platdata *pdata = dev_get_platdata(&op->dev);
 	struct resource *iores;
-	int ret, dma_channels = 0, nb_requestors = 0;
+	int ret, dma_channels = 0, nb_requestors = 0, slave_map_cnt = 0;
 	const enum dma_slave_buswidth widths =
 		DMA_SLAVE_BUSWIDTH_1_BYTE   | DMA_SLAVE_BUSWIDTH_2_BYTES |
 		DMA_SLAVE_BUSWIDTH_4_BYTES;
@@ -1429,6 +1437,8 @@ static int pxad_probe(struct platform_device *op)
 	} else if (pdata && pdata->dma_channels) {
 		dma_channels = pdata->dma_channels;
 		nb_requestors = pdata->nb_requestors;
+		slave_map = pdata->slave_map;
+		slave_map_cnt = pdata->slave_map_cnt;
 	} else {
 		dma_channels = 32;	/* default 32 channel */
 	}
@@ -1440,6 +1450,9 @@ static int pxad_probe(struct platform_device *op)
 	pdev->slave.device_prep_dma_memcpy = pxad_prep_memcpy;
 	pdev->slave.device_prep_slave_sg = pxad_prep_slave_sg;
 	pdev->slave.device_prep_dma_cyclic = pxad_prep_dma_cyclic;
+	pdev->slave.filter.map = slave_map;
+	pdev->slave.filter.mapcnt = slave_map_cnt;
+	pdev->slave.filter.fn = pxad_filter_fn;
 
 	pdev->slave.copy_align = PDMA_ALIGNMENT;
 	pdev->slave.src_addr_widths = widths;

drivers/gpu/drm/i915/Kconfig

@@ -23,6 +23,7 @@ config DRM_I915
 	select SYNC_FILE
 	select IOSF_MBI
 	select CRC32
+	select SND_HDA_I915 if SND_HDA_CORE
 	help
 	  Choose this option if you have a system that has "Intel Graphics
 	  Media Accelerator" or "HD Graphics" integrated graphics,

drivers/gpu/drm/i915/intel_audio.c

@@ -639,11 +639,12 @@ void intel_audio_codec_enable(struct intel_encoder *encoder,
 	dev_priv->av_enc_map[pipe] = encoder;
 	mutex_unlock(&dev_priv->av_mutex);
 
-	if (acomp && acomp->audio_ops && acomp->audio_ops->pin_eld_notify) {
+	if (acomp && acomp->base.audio_ops &&
+	    acomp->base.audio_ops->pin_eld_notify) {
 		/* audio drivers expect pipe = -1 to indicate Non-MST cases */
 		if (!intel_crtc_has_type(crtc_state, INTEL_OUTPUT_DP_MST))
 			pipe = -1;
-		acomp->audio_ops->pin_eld_notify(acomp->audio_ops->audio_ptr,
+		acomp->base.audio_ops->pin_eld_notify(acomp->base.audio_ops->audio_ptr,
 						 (int) port, (int) pipe);
 	}
 
@@ -681,11 +682,12 @@ void intel_audio_codec_disable(struct intel_encoder *encoder,
 	dev_priv->av_enc_map[pipe] = NULL;
 	mutex_unlock(&dev_priv->av_mutex);
 
-	if (acomp && acomp->audio_ops && acomp->audio_ops->pin_eld_notify) {
+	if (acomp && acomp->base.audio_ops &&
+	    acomp->base.audio_ops->pin_eld_notify) {
 		/* audio drivers expect pipe = -1 to indicate Non-MST cases */
 		if (!intel_crtc_has_type(old_crtc_state, INTEL_OUTPUT_DP_MST))
 			pipe = -1;
-		acomp->audio_ops->pin_eld_notify(acomp->audio_ops->audio_ptr,
+		acomp->base.audio_ops->pin_eld_notify(acomp->base.audio_ops->audio_ptr,
 						 (int) port, (int) pipe);
 	}
 
@@ -880,7 +882,7 @@ static int i915_audio_component_get_eld(struct device *kdev, int port,
 	return ret;
 }
 
-static const struct i915_audio_component_ops i915_audio_component_ops = {
+static const struct drm_audio_component_ops i915_audio_component_ops = {
 	.owner		= THIS_MODULE,
 	.get_power	= i915_audio_component_get_power,
 	.put_power	= i915_audio_component_put_power,
@@ -897,12 +899,12 @@ static int i915_audio_component_bind(struct device *i915_kdev,
 	struct drm_i915_private *dev_priv = kdev_to_i915(i915_kdev);
 	int i;
 
-	if (WARN_ON(acomp->ops || acomp->dev))
+	if (WARN_ON(acomp->base.ops || acomp->base.dev))
 		return -EEXIST;
 
 	drm_modeset_lock_all(&dev_priv->drm);
-	acomp->ops = &i915_audio_component_ops;
-	acomp->dev = i915_kdev;
+	acomp->base.ops = &i915_audio_component_ops;
+	acomp->base.dev = i915_kdev;
 	BUILD_BUG_ON(MAX_PORTS != I915_MAX_PORTS);
 	for (i = 0; i < ARRAY_SIZE(acomp->aud_sample_rate); i++)
 		acomp->aud_sample_rate[i] = 0;
@@ -919,8 +921,8 @@ static void i915_audio_component_unbind(struct device *i915_kdev,
 	struct drm_i915_private *dev_priv = kdev_to_i915(i915_kdev);
 
 	drm_modeset_lock_all(&dev_priv->drm);
-	acomp->ops = NULL;
-	acomp->dev = NULL;
+	acomp->base.ops = NULL;
+	acomp->base.dev = NULL;
 	dev_priv->audio_component = NULL;
 	drm_modeset_unlock_all(&dev_priv->drm);
 }

drivers/gpu/vga/vga_switcheroo.c

@@ -103,9 +103,11 @@
  *	runtime pm. If true, writing ON and OFF to the vga_switcheroo debugfs
  *	interface is a no-op so as not to interfere with runtime pm
  * @list: client list
+ * @vga_dev: pci device, indicate which GPU is bound to current audio client
  *
  * Registered client. A client can be either a GPU or an audio device on a GPU.
- * For audio clients, the @fb_info and @active members are bogus.
+ * For audio clients, the @fb_info and @active members are bogus. For GPU
+ * clients, the @vga_dev is bogus.
  */
 struct vga_switcheroo_client {
 	struct pci_dev *pdev;
@@ -116,6 +118,7 @@ struct vga_switcheroo_client {
 	bool active;
 	bool driver_power_control;
 	struct list_head list;
+	struct pci_dev *vga_dev;
 };
 
 /*
@@ -161,9 +164,8 @@ struct vgasr_priv {
 };
 
 #define ID_BIT_AUDIO		0x100
-#define client_is_audio(c)	((c)->id & ID_BIT_AUDIO)
-#define client_is_vga(c)	((c)->id == VGA_SWITCHEROO_UNKNOWN_ID || \
-				 !client_is_audio(c))
+#define client_is_audio(c)		((c)->id & ID_BIT_AUDIO)
+#define client_is_vga(c)		(!client_is_audio(c))
 #define client_id(c)		((c)->id & ~ID_BIT_AUDIO)
 
 static int vga_switcheroo_debugfs_init(struct vgasr_priv *priv);
@@ -192,14 +194,29 @@ static void vga_switcheroo_enable(void)
 		vgasr_priv.handler->init();
 
 	list_for_each_entry(client, &vgasr_priv.clients, list) {
-		if (client->id != VGA_SWITCHEROO_UNKNOWN_ID)
+		if (!client_is_vga(client) ||
+		     client_id(client) != VGA_SWITCHEROO_UNKNOWN_ID)
 			continue;
+
 		ret = vgasr_priv.handler->get_client_id(client->pdev);
 		if (ret < 0)
 			return;
 
 		client->id = ret;
 	}
+
+	list_for_each_entry(client, &vgasr_priv.clients, list) {
+		if (!client_is_audio(client) ||
+		     client_id(client) != VGA_SWITCHEROO_UNKNOWN_ID)
+			continue;
+
+		ret = vgasr_priv.handler->get_client_id(client->vga_dev);
+		if (ret < 0)
+			return;
+
+		client->id = ret | ID_BIT_AUDIO;
+	}
+
 	vga_switcheroo_debugfs_init(&vgasr_priv);
 	vgasr_priv.active = true;
 }
@@ -272,7 +289,9 @@ EXPORT_SYMBOL(vga_switcheroo_handler_flags);
 
 static int register_client(struct pci_dev *pdev,
 			   const struct vga_switcheroo_client_ops *ops,
-			   enum vga_switcheroo_client_id id, bool active,
+			   enum vga_switcheroo_client_id id,
+			   struct pci_dev *vga_dev,
+			   bool active,
 			   bool driver_power_control)
 {
 	struct vga_switcheroo_client *client;
@@ -287,6 +306,7 @@ static int register_client(struct pci_dev *pdev,
 	client->id = id;
 	client->active = active;
 	client->driver_power_control = driver_power_control;
+	client->vga_dev = vga_dev;
 
 	mutex_lock(&vgasr_mutex);
 	list_add_tail(&client->list, &vgasr_priv.clients);
@@ -319,7 +339,7 @@ int vga_switcheroo_register_client(struct pci_dev *pdev,
 				   const struct vga_switcheroo_client_ops *ops,
 				   bool driver_power_control)
 {
-	return register_client(pdev, ops, VGA_SWITCHEROO_UNKNOWN_ID,
+	return register_client(pdev, ops, VGA_SWITCHEROO_UNKNOWN_ID, NULL,
 			       pdev == vga_default_device(),
 			       driver_power_control);
 }
@@ -329,19 +349,40 @@ EXPORT_SYMBOL(vga_switcheroo_register_client);
  * vga_switcheroo_register_audio_client - register audio client
  * @pdev: client pci device
  * @ops: client callbacks
- * @id: client identifier
+ * @vga_dev:  pci device which is bound to current audio client
  *
  * Register audio client (audio device on a GPU). The client is assumed
  * to use runtime PM. Beforehand, vga_switcheroo_client_probe_defer()
  * shall be called to ensure that all prerequisites are met.
  *
- * Return: 0 on success, -ENOMEM on memory allocation error.
+ * Return: 0 on success, -ENOMEM on memory allocation error, -EINVAL on getting
+ * client id error.
  */
 int vga_switcheroo_register_audio_client(struct pci_dev *pdev,
 			const struct vga_switcheroo_client_ops *ops,
-			enum vga_switcheroo_client_id id)
+			struct pci_dev *vga_dev)
 {
-	return register_client(pdev, ops, id | ID_BIT_AUDIO, false, true);
+	enum vga_switcheroo_client_id id = VGA_SWITCHEROO_UNKNOWN_ID;
+
+	/*
+	 * if vga_switcheroo has enabled, that mean two GPU clients and also
+	 * handler are registered. Get audio client id from bound GPU client
+	 * id directly, otherwise, set it as VGA_SWITCHEROO_UNKNOWN_ID,
+	 * it will set to correct id in later when vga_switcheroo_enable()
+	 * is called.
+	 */
+	mutex_lock(&vgasr_mutex);
+	if (vgasr_priv.active) {
+		id = vgasr_priv.handler->get_client_id(vga_dev);
+		if (id < 0) {
+			mutex_unlock(&vgasr_mutex);
+			return -EINVAL;
+		}
+	}
+	mutex_unlock(&vgasr_mutex);
+
+	return register_client(pdev, ops, id | ID_BIT_AUDIO, vga_dev,
+			       false, true);
 }
 EXPORT_SYMBOL(vga_switcheroo_register_audio_client);
 

drivers/media/platform/pxa_camera.c

@@ -2375,8 +2375,6 @@ static int pxa_camera_probe(struct platform_device *pdev)
 		.src_maxburst = 8,
 		.direction = DMA_DEV_TO_MEM,
 	};
-	dma_cap_mask_t mask;
-	struct pxad_param params;
 	char clk_name[V4L2_CLK_NAME_SIZE];
 	int irq;
 	int err = 0, i;
@@ -2450,34 +2448,20 @@ static int pxa_camera_probe(struct platform_device *pdev)
 	pcdev->base = base;
 
 	/* request dma */
-	dma_cap_zero(mask);
-	dma_cap_set(DMA_SLAVE, mask);
-	dma_cap_set(DMA_PRIVATE, mask);
-
-	params.prio = 0;
-	params.drcmr = 68;
-	pcdev->dma_chans[0] =
-		dma_request_slave_channel_compat(mask, pxad_filter_fn,
-						 &params, &pdev->dev, "CI_Y");
+	pcdev->dma_chans[0] = dma_request_slave_channel(&pdev->dev, "CI_Y");
 	if (!pcdev->dma_chans[0]) {
 		dev_err(&pdev->dev, "Can't request DMA for Y\n");
 		return -ENODEV;
 	}
 
-	params.drcmr = 69;
-	pcdev->dma_chans[1] =
-		dma_request_slave_channel_compat(mask, pxad_filter_fn,
-						 &params, &pdev->dev, "CI_U");
+	pcdev->dma_chans[1] = dma_request_slave_channel(&pdev->dev, "CI_U");
 	if (!pcdev->dma_chans[1]) {
 		dev_err(&pdev->dev, "Can't request DMA for Y\n");
 		err = -ENODEV;
 		goto exit_free_dma_y;
 	}
 
-	params.drcmr = 70;
-	pcdev->dma_chans[2] =
-		dma_request_slave_channel_compat(mask, pxad_filter_fn,
-						 &params, &pdev->dev, "CI_V");
+	pcdev->dma_chans[2] = dma_request_slave_channel(&pdev->dev, "CI_V");
 	if (!pcdev->dma_chans[2]) {
 		dev_err(&pdev->dev, "Can't request DMA for V\n");
 		err = -ENODEV;

drivers/mmc/host/pxamci.c

@@ -24,7 +24,6 @@
 #include <linux/interrupt.h>
 #include <linux/dmaengine.h>
 #include <linux/dma-mapping.h>
-#include <linux/dma/pxa-dma.h>
 #include <linux/clk.h>
 #include <linux/err.h>
 #include <linux/mmc/host.h>
@@ -637,10 +636,8 @@ static int pxamci_probe(struct platform_device *pdev)
 {
 	struct mmc_host *mmc;
 	struct pxamci_host *host = NULL;
-	struct resource *r, *dmarx, *dmatx;
-	struct pxad_param param_rx, param_tx;
+	struct resource *r;
 	int ret, irq, gpio_cd = -1, gpio_ro = -1, gpio_power = -1;
-	dma_cap_mask_t mask;
 
 	ret = pxamci_of_init(pdev);
 	if (ret)
@@ -739,34 +736,14 @@ static int pxamci_probe(struct platform_device *pdev)
 
 	platform_set_drvdata(pdev, mmc);
 
-	if (!pdev->dev.of_node) {
-		dmarx = platform_get_resource(pdev, IORESOURCE_DMA, 0);
-		dmatx = platform_get_resource(pdev, IORESOURCE_DMA, 1);
-		if (!dmarx || !dmatx) {
-			ret = -ENXIO;
-			goto out;
-		}
-		param_rx.prio = PXAD_PRIO_LOWEST;
-		param_rx.drcmr = dmarx->start;
-		param_tx.prio = PXAD_PRIO_LOWEST;
-		param_tx.drcmr = dmatx->start;
-	}
-
-	dma_cap_zero(mask);
-	dma_cap_set(DMA_SLAVE, mask);
-
-	host->dma_chan_rx =
-		dma_request_slave_channel_compat(mask, pxad_filter_fn,
-						 &param_rx, &pdev->dev, "rx");
+	host->dma_chan_rx = dma_request_slave_channel(&pdev->dev, "rx");
 	if (host->dma_chan_rx == NULL) {
 		dev_err(&pdev->dev, "unable to request rx dma channel\n");
 		ret = -ENODEV;
 		goto out;
 	}
 
-	host->dma_chan_tx =
-		dma_request_slave_channel_compat(mask, pxad_filter_fn,
-						 &param_tx,  &pdev->dev, "tx");
+	host->dma_chan_tx = dma_request_slave_channel(&pdev->dev, "tx");
 	if (host->dma_chan_tx == NULL) {
 		dev_err(&pdev->dev, "unable to request tx dma channel\n");
 		ret = -ENODEV;

drivers/mtd/nand/raw/marvell_nand.c

@@ -2618,8 +2618,6 @@ static int marvell_nfc_init_dma(struct marvell_nfc *nfc)
 						    dev);
 	struct dma_slave_config config = {};
 	struct resource *r;
-	dma_cap_mask_t mask;
-	struct pxad_param param;
 	int ret;
 
 	if (!IS_ENABLED(CONFIG_PXA_DMA)) {
@@ -2632,20 +2630,7 @@ static int marvell_nfc_init_dma(struct marvell_nfc *nfc)
 	if (ret)
 		return ret;
 
-	r = platform_get_resource(pdev, IORESOURCE_DMA, 0);
-	if (!r) {
-		dev_err(nfc->dev, "No resource defined for data DMA\n");
-		return -ENXIO;
-	}
-
-	param.drcmr = r->start;
-	param.prio = PXAD_PRIO_LOWEST;
-	dma_cap_zero(mask);
-	dma_cap_set(DMA_SLAVE, mask);
-	nfc->dma_chan =
-		dma_request_slave_channel_compat(mask, pxad_filter_fn,
-						 &param, nfc->dev,
-						 "data");
+	nfc->dma_chan =	dma_request_slave_channel(nfc->dev, "data");
 	if (!nfc->dma_chan) {
 		dev_err(nfc->dev,
 			"Unable to request data DMA channel\n");

drivers/staging/most/sound/sound.c

@@ -457,7 +457,6 @@ static const struct snd_pcm_ops pcm_ops = {
 	.trigger    = pcm_trigger,
 	.pointer    = pcm_pointer,
 	.page       = snd_pcm_lib_get_vmalloc_page,
-	.mmap       = snd_pcm_lib_mmap_vmalloc,
 };
 
 static int split_arg_list(char *buf, char **card_name, u16 *ch_num,

include/drm/drm_audio_component.h

@@ -0,0 +1,118 @@
+// SPDX-License-Identifier: MIT
+// Copyright © 2014 Intel Corporation
+
+#ifndef _DRM_AUDIO_COMPONENT_H_
+#define _DRM_AUDIO_COMPONENT_H_
+
+struct drm_audio_component;
+
+/**
+ * struct drm_audio_component_ops - Ops implemented by DRM driver, called by hda driver
+ */
+struct drm_audio_component_ops {
+	/**
+	 * @owner: drm module to pin down
+	 */
+	struct module *owner;
+	/**
+	 * @get_power: get the POWER_DOMAIN_AUDIO power well
+	 *
+	 * Request the power well to be turned on.
+	 */
+	void (*get_power)(struct device *);
+	/**
+	 * @put_power: put the POWER_DOMAIN_AUDIO power well
+	 *
+	 * Allow the power well to be turned off.
+	 */
+	void (*put_power)(struct device *);
+	/**
+	 * @codec_wake_override: Enable/disable codec wake signal
+	 */
+	void (*codec_wake_override)(struct device *, bool enable);
+	/**
+	 * @get_cdclk_freq: Get the Core Display Clock in kHz
+	 */
+	int (*get_cdclk_freq)(struct device *);
+	/**
+	 * @sync_audio_rate: set n/cts based on the sample rate
+	 *
+	 * Called from audio driver. After audio driver sets the
+	 * sample rate, it will call this function to set n/cts
+	 */
+	int (*sync_audio_rate)(struct device *, int port, int pipe, int rate);
+	/**
+	 * @get_eld: fill the audio state and ELD bytes for the given port
+	 *
+	 * Called from audio driver to get the HDMI/DP audio state of the given
+	 * digital port, and also fetch ELD bytes to the given pointer.
+	 *
+	 * It returns the byte size of the original ELD (not the actually
+	 * copied size), zero for an invalid ELD, or a negative error code.
+	 *
+	 * Note that the returned size may be over @max_bytes.  Then it
+	 * implies that only a part of ELD has been copied to the buffer.
+	 */
+	int (*get_eld)(struct device *, int port, int pipe, bool *enabled,
+		       unsigned char *buf, int max_bytes);
+};
+
+/**
+ * struct drm_audio_component_audio_ops - Ops implemented by hda driver, called by DRM driver
+ */
+struct drm_audio_component_audio_ops {
+	/**
+	 * @audio_ptr: Pointer to be used in call to pin_eld_notify
+	 */
+	void *audio_ptr;
+	/**
+	 * @pin_eld_notify: Notify the HDA driver that pin sense and/or ELD information has changed
+	 *
+	 * Called when the DRM driver has set up audio pipeline or has just
+	 * begun to tear it down. This allows the HDA driver to update its
+	 * status accordingly (even when the HDA controller is in power save
+	 * mode).
+	 */
+	void (*pin_eld_notify)(void *audio_ptr, int port, int pipe);
+	/**
+	 * @pin2port: Check and convert from pin node to port number
+	 *
+	 * Called by HDA driver to check and convert from the pin widget node
+	 * number to a port number in the graphics side.
+	 */
+	int (*pin2port)(void *audio_ptr, int pin);
+	/**
+	 * @master_bind: (Optional) component master bind callback
+	 *
+	 * Called at binding master component, for HDA codec-specific
+	 * handling of dynamic binding.
+	 */
+	int (*master_bind)(struct device *dev, struct drm_audio_component *);
+	/**
+	 * @master_unbind: (Optional) component master unbind callback
+	 *
+	 * Called at unbinding master component, for HDA codec-specific
+	 * handling of dynamic unbinding.
+	 */
+	void (*master_unbind)(struct device *dev, struct drm_audio_component *);
+};
+
+/**
+ * struct drm_audio_component - Used for direct communication between DRM and hda drivers
+ */
+struct drm_audio_component {
+	/**
+	 * @dev: DRM device, used as parameter for ops
+	 */
+	struct device *dev;
+	/**
+	 * @ops: Ops implemented by DRM driver, called by hda driver
+	 */
+	const struct drm_audio_component_ops *ops;
+	/**
+	 * @audio_ops: Ops implemented by hda driver, called by DRM driver
+	 */
+	const struct drm_audio_component_audio_ops *audio_ops;
+};
+
+#endif /* _DRM_AUDIO_COMPONENT_H_ */

include/drm/i915_component.h

@@ -24,101 +24,26 @@
 #ifndef _I915_COMPONENT_H_
 #define _I915_COMPONENT_H_
 
+#include "drm_audio_component.h"
+
 /* MAX_PORT is the number of port
  * It must be sync with I915_MAX_PORTS defined i915_drv.h
  */
 #define MAX_PORTS 6
 
-/**
- * struct i915_audio_component_ops - Ops implemented by i915 driver, called by hda driver
- */
-struct i915_audio_component_ops {
-	/**
-	 * @owner: i915 module
-	 */
-	struct module *owner;
-	/**
-	 * @get_power: get the POWER_DOMAIN_AUDIO power well
-	 *
-	 * Request the power well to be turned on.
-	 */
-	void (*get_power)(struct device *);
-	/**
-	 * @put_power: put the POWER_DOMAIN_AUDIO power well
-	 *
-	 * Allow the power well to be turned off.
-	 */
-	void (*put_power)(struct device *);
-	/**
-	 * @codec_wake_override: Enable/disable codec wake signal
-	 */
-	void (*codec_wake_override)(struct device *, bool enable);
-	/**
-	 * @get_cdclk_freq: Get the Core Display Clock in kHz
-	 */
-	int (*get_cdclk_freq)(struct device *);
-	/**
-	 * @sync_audio_rate: set n/cts based on the sample rate
-	 *
-	 * Called from audio driver. After audio driver sets the
-	 * sample rate, it will call this function to set n/cts
-	 */
-	int (*sync_audio_rate)(struct device *, int port, int pipe, int rate);
-	/**
-	 * @get_eld: fill the audio state and ELD bytes for the given port
-	 *
-	 * Called from audio driver to get the HDMI/DP audio state of the given
-	 * digital port, and also fetch ELD bytes to the given pointer.
-	 *
-	 * It returns the byte size of the original ELD (not the actually
-	 * copied size), zero for an invalid ELD, or a negative error code.
-	 *
-	 * Note that the returned size may be over @max_bytes.  Then it
-	 * implies that only a part of ELD has been copied to the buffer.
-	 */
-	int (*get_eld)(struct device *, int port, int pipe, bool *enabled,
-		       unsigned char *buf, int max_bytes);
-};
-
-/**
- * struct i915_audio_component_audio_ops - Ops implemented by hda driver, called by i915 driver
- */
-struct i915_audio_component_audio_ops {
-	/**
-	 * @audio_ptr: Pointer to be used in call to pin_eld_notify
-	 */
-	void *audio_ptr;
-	/**
-	 * @pin_eld_notify: Notify the HDA driver that pin sense and/or ELD information has changed
-	 *
-	 * Called when the i915 driver has set up audio pipeline or has just
-	 * begun to tear it down. This allows the HDA driver to update its
-	 * status accordingly (even when the HDA controller is in power save
-	 * mode).
-	 */
-	void (*pin_eld_notify)(void *audio_ptr, int port, int pipe);
-};
-
 /**
  * struct i915_audio_component - Used for direct communication between i915 and hda drivers
  */
 struct i915_audio_component {
 	/**
-	 * @dev: i915 device, used as parameter for ops
+	 * @base: the drm_audio_component base class
 	 */
-	struct device *dev;
+	struct drm_audio_component	base;
+
 	/**
 	 * @aud_sample_rate: the array of audio sample rate per port
 	 */
 	int aud_sample_rate[MAX_PORTS];
-	/**
-	 * @ops: Ops implemented by i915 driver, called by hda driver
-	 */
-	const struct i915_audio_component_ops *ops;
-	/**
-	 * @audio_ops: Ops implemented by hda driver, called by i915 driver
-	 */
-	const struct i915_audio_component_audio_ops *audio_ops;
 };
 
 #endif /* _I915_COMPONENT_H_ */

include/linux/clk-provider.h

@@ -38,6 +38,8 @@
 #define CLK_IS_CRITICAL		BIT(11) /* do not gate, ever */
 /* parents need enable during gate/ungate, set rate and re-parent */
 #define CLK_OPS_PARENT_ENABLE	BIT(12)
+/* duty cycle call may be forwarded to the parent clock */
+#define CLK_DUTY_CYCLE_PARENT	BIT(13)
 
 struct clk;
 struct clk_hw;
@@ -66,6 +68,17 @@ struct clk_rate_request {
 	struct clk_hw *best_parent_hw;
 };
 
+/**
+ * struct clk_duty - Struture encoding the duty cycle ratio of a clock
+ *
+ * @num:	Numerator of the duty cycle ratio
+ * @den:	Denominator of the duty cycle ratio
+ */
+struct clk_duty {
+	unsigned int num;
+	unsigned int den;
+};
+
 /**
  * struct clk_ops -  Callback operations for hardware clocks; these are to
  * be provided by the clock implementation, and will be called by drivers
@@ -169,6 +182,15 @@ struct clk_rate_request {
  *		by the second argument. Valid values for degrees are
  *		0-359. Return 0 on success, otherwise -EERROR.
  *
+ * @get_duty_cycle: Queries the hardware to get the current duty cycle ratio
+ *              of a clock. Returned values denominator cannot be 0 and must be
+ *              superior or equal to the numerator.
+ *
+ * @set_duty_cycle: Apply the duty cycle ratio to this clock signal specified by
+ *              the numerator (2nd argurment) and denominator (3rd  argument).
+ *              Argument must be a valid ratio (denominator > 0
+ *              and >= numerator) Return 0 on success, otherwise -EERROR.
+ *
  * @init:	Perform platform-specific initialization magic.
  *		This is not not used by any of the basic clock types.
  *		Please consider other ways of solving initialization problems
@@ -218,6 +240,10 @@ struct clk_ops {
 					   unsigned long parent_accuracy);
 	int		(*get_phase)(struct clk_hw *hw);
 	int		(*set_phase)(struct clk_hw *hw, int degrees);
+	int		(*get_duty_cycle)(struct clk_hw *hw,
+					  struct clk_duty *duty);
+	int		(*set_duty_cycle)(struct clk_hw *hw,
+					  struct clk_duty *duty);
 	void		(*init)(struct clk_hw *hw);
 	void		(*debug_init)(struct clk_hw *hw, struct dentry *dentry);
 };

include/linux/clk.h

@@ -141,6 +141,27 @@ int clk_set_phase(struct clk *clk, int degrees);
  */
 int clk_get_phase(struct clk *clk);
 
+/**
+ * clk_set_duty_cycle - adjust the duty cycle ratio of a clock signal
+ * @clk: clock signal source
+ * @num: numerator of the duty cycle ratio to be applied
+ * @den: denominator of the duty cycle ratio to be applied
+ *
+ * Adjust the duty cycle of a clock signal by the specified ratio. Returns 0 on
+ * success, -EERROR otherwise.
+ */
+int clk_set_duty_cycle(struct clk *clk, unsigned int num, unsigned int den);
+
+/**
+ * clk_get_duty_cycle - return the duty cycle ratio of a clock signal
+ * @clk: clock signal source
+ * @scale: scaling factor to be applied to represent the ratio as an integer
+ *
+ * Returns the duty cycle ratio multiplied by the scale provided, otherwise
+ * returns -EERROR.
+ */
+int clk_get_scaled_duty_cycle(struct clk *clk, unsigned int scale);
+
 /**
  * clk_is_match - check if two clk's point to the same hardware clock
  * @p: clk compared against q
@@ -183,6 +204,18 @@ static inline long clk_get_phase(struct clk *clk)
 	return -ENOTSUPP;
 }
 
+static inline int clk_set_duty_cycle(struct clk *clk, unsigned int num,
+				     unsigned int den)
+{
+	return -ENOTSUPP;
+}
+
+static inline unsigned int clk_get_scaled_duty_cycle(struct clk *clk,
+						     unsigned int scale)
+{
+	return 0;
+}
+
 static inline bool clk_is_match(const struct clk *p, const struct clk *q)
 {
 	return p == q;

include/linux/dma/pxa-dma.h

@@ -9,6 +9,15 @@ enum pxad_chan_prio {
 	PXAD_PRIO_LOWEST,
 };
 
+/**
+ * struct pxad_param - dma channel request parameters
+ * @drcmr: requestor line number
+ * @prio: minimal mandatory priority of the channel
+ *
+ * If a requested channel is granted, its priority will be at least @prio,
+ * ie. if PXAD_PRIO_LOW is required, the requested channel will be either
+ * PXAD_PRIO_LOW, PXAD_PRIO_NORMAL or PXAD_PRIO_HIGHEST.
+ */
 struct pxad_param {
 	unsigned int drcmr;
 	enum pxad_chan_prio prio;

include/linux/platform_data/mmp_dma.h

@@ -12,9 +12,13 @@
 #ifndef MMP_DMA_H
 #define MMP_DMA_H
 
+struct dma_slave_map;
+
 struct mmp_dma_platdata {
 	int dma_channels;
 	int nb_requestors;
+	int slave_map_cnt;
+	const struct dma_slave_map *slave_map;
 };
 
 #endif /* MMP_DMA_H */

include/linux/pxa2xx_ssp.h

@@ -171,6 +171,14 @@
 #define SSACD_SCDB		(1 << 3)	/* SSPSYSCLK Divider Bypass */
 #define SSACD_ACPS(x)		((x) << 4)	/* Audio clock PLL select */
 #define SSACD_ACDS(x)		((x) << 0)	/* Audio clock divider select */
+#define SSACD_ACDS_1		(0)
+#define SSACD_ACDS_2		(1)
+#define SSACD_ACDS_4		(2)
+#define SSACD_ACDS_8		(3)
+#define SSACD_ACDS_16		(4)
+#define SSACD_ACDS_32		(5)
+#define SSACD_SCDB_4X		(0)
+#define SSACD_SCDB_1X		(1)
 #define SSACD_SCDX8		(1 << 7)	/* SYSCLK division ratio select */
 
 /* LPSS SSP */
@@ -212,8 +220,6 @@ struct ssp_device {
 	int		type;
 	int		use_count;
 	int		irq;
-	int		drcmr_rx;
-	int		drcmr_tx;
 
 	struct device_node	*of_node;
 };

include/linux/usb/audio-v3.h

@@ -387,6 +387,12 @@ struct uac3_interrupt_data_msg {
 #define UAC3_CONNECTORS			0x0f
 #define UAC3_POWER_DOMAIN		0x10
 
+/* A.20 PROCESSING UNIT PROCESS TYPES */
+#define UAC3_PROCESS_UNDEFINED		0x00
+#define UAC3_PROCESS_UP_DOWNMIX		0x01
+#define UAC3_PROCESS_STEREO_EXTENDER	0x02
+#define UAC3_PROCESS_MULTI_FUNCTION	0x03
+
 /* A.22 AUDIO CLASS-SPECIFIC REQUEST CODES */
 /* see audio-v2.h for the rest, which is identical to v2 */
 #define UAC3_CS_REQ_INTEN			0x04
@@ -406,6 +412,15 @@ struct uac3_interrupt_data_msg {
 #define UAC3_TE_OVERFLOW			0x04
 #define UAC3_TE_LATENCY 			0x05
 
+/* A.23.10 PROCESSING UNITS CONTROL SELECTROS */
+
+/* Up/Down Mixer */
+#define UAC3_UD_MODE_SELECT			0x01
+
+/* Stereo Extender */
+#define UAC3_EXT_WIDTH_CONTROL			0x01
+
+
 /* BADD predefined Unit/Terminal values */
 #define UAC3_BADD_IT_ID1	1  /* Input Terminal ID1: bTerminalID = 1 */
 #define UAC3_BADD_FU_ID2	2  /* Feature Unit ID2: bUnitID = 2 */
@@ -432,4 +447,8 @@ struct uac3_interrupt_data_msg {
 /* BADD sample rate is always fixed to 48kHz */
 #define UAC3_BADD_SAMPLING_RATE				48000
 
+/* BADD power domains recovery times in 50us increments */
+#define UAC3_BADD_PD_RECOVER_D1D0			0x0258	/* 30ms */
+#define UAC3_BADD_PD_RECOVER_D2D0			0x1770	/* 300ms */
+
 #endif /* __LINUX_USB_AUDIO_V3_H */

include/linux/vga_switcheroo.h

@@ -84,8 +84,8 @@ enum vga_switcheroo_state {
  * Client identifier. Audio clients use the same identifier & 0x100.
  */
 enum vga_switcheroo_client_id {
-	VGA_SWITCHEROO_UNKNOWN_ID = -1,
-	VGA_SWITCHEROO_IGD,
+	VGA_SWITCHEROO_UNKNOWN_ID = 0x1000,
+	VGA_SWITCHEROO_IGD = 0,
 	VGA_SWITCHEROO_DIS,
 	VGA_SWITCHEROO_MAX_CLIENTS,
 };
@@ -151,7 +151,7 @@ int vga_switcheroo_register_client(struct pci_dev *dev,
 				   bool driver_power_control);
 int vga_switcheroo_register_audio_client(struct pci_dev *pdev,
 					 const struct vga_switcheroo_client_ops *ops,
-					 enum vga_switcheroo_client_id id);
+					 struct pci_dev *vga_dev);
 
 void vga_switcheroo_client_fb_set(struct pci_dev *dev,
 				  struct fb_info *info);
@@ -180,7 +180,7 @@ static inline int vga_switcheroo_register_handler(const struct vga_switcheroo_ha
 		enum vga_switcheroo_handler_flags_t handler_flags) { return 0; }
 static inline int vga_switcheroo_register_audio_client(struct pci_dev *pdev,
 	const struct vga_switcheroo_client_ops *ops,
-	enum vga_switcheroo_client_id id) { return 0; }
+	struct pci_dev *vga_dev) { return 0; }
 static inline void vga_switcheroo_unregister_handler(void) {}
 static inline enum vga_switcheroo_handler_flags_t vga_switcheroo_handler_flags(void) { return 0; }
 static inline int vga_switcheroo_lock_ddc(struct pci_dev *pdev) { return -ENODEV; }

include/sound/ac97/codec.h

@@ -1,10 +1,8 @@
-/*
- *  Copyright (C) 2016 Robert Jarzmik <robert.jarzmik@free.fr>
+/* SPDX-License-Identifier: GPL-2.0
  *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
+ *  Copyright (C) 2016 Robert Jarzmik <robert.jarzmik@free.fr>
  */
+
 #ifndef __SOUND_AC97_CODEC2_H
 #define __SOUND_AC97_CODEC2_H
 

include/sound/ac97/compat.h

@@ -1,14 +1,11 @@
-/*
- *  Copyright (C) 2016 Robert Jarzmik <robert.jarzmik@free.fr>
+/* SPDX-License-Identifier: GPL-2.0
  *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
+ *  Copyright (C) 2016 Robert Jarzmik <robert.jarzmik@free.fr>
  *
  * This file is for backward compatibility with snd_ac97 structure and its
  * multiple usages, such as the snd_ac97_bus and snd_ac97_build_ops.
- *
  */
+
 #ifndef AC97_COMPAT_H
 #define AC97_COMPAT_H
 

include/sound/ac97/controller.h

@@ -1,10 +1,8 @@
-/*
- *  Copyright (C) 2016 Robert Jarzmik <robert.jarzmik@free.fr>
+/* SPDX-License-Identifier: GPL-2.0
  *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
+ * Copyright (C) 2016 Robert Jarzmik <robert.jarzmik@free.fr>
  */
+
 #ifndef AC97_CONTROLLER_H
 #define AC97_CONTROLLER_H
 

include/sound/ac97/regs.h

@@ -1,27 +1,11 @@
-/*
+/* SPDX-License-Identifier: GPL-2.0+
+ *
  *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>
  *  Universal interface for Audio Codec '97
  *
  *  For more details look to AC '97 component specification revision 2.1
  *  by Intel Corporation (http://developer.intel.com).
- *
- *
- *   This program is free software; you can redistribute it and/or modify
- *   it under the terms of the GNU General Public License as published by
- *   the Free Software Foundation; either version 2 of the License, or
- *   (at your option) any later version.
- *
- *   This program is distributed in the hope that it will be useful,
- *   but WITHOUT ANY WARRANTY; without even the implied warranty of
- *   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- *   GNU General Public License for more details.
- *
- *   You should have received a copy of the GNU General Public License
- *   along with this program; if not, write to the Free Software
- *   Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
- *
  */
-
 /*
  *  AC'97 codec registers
  */

include/sound/ac97_codec.h

@@ -1,30 +1,15 @@
-#ifndef __SOUND_AC97_CODEC_H
-#define __SOUND_AC97_CODEC_H
-
-/*
+/* SPDX-License-Identifier: GPL-2.0+
+ *
  *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>
  *  Universal interface for Audio Codec '97
  *
  *  For more details look to AC '97 component specification revision 2.1
  *  by Intel Corporation (http://developer.intel.com).
- *
- *
- *   This program is free software; you can redistribute it and/or modify
- *   it under the terms of the GNU General Public License as published by
- *   the Free Software Foundation; either version 2 of the License, or
- *   (at your option) any later version.
- *
- *   This program is distributed in the hope that it will be useful,
- *   but WITHOUT ANY WARRANTY; without even the implied warranty of
- *   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- *   GNU General Public License for more details.
- *
- *   You should have received a copy of the GNU General Public License
- *   along with this program; if not, write to the Free Software
- *   Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
- *
  */
 
+#ifndef __SOUND_AC97_CODEC_H
+#define __SOUND_AC97_CODEC_H
+
 #include <linux/bitops.h>
 #include <linux/device.h>
 #include <linux/workqueue.h>

include/sound/compress_driver.h

@@ -1,27 +1,12 @@
-/*
+/* SPDX-License-Identifier: GPL-2.0
+ *
  *  compress_driver.h - compress offload driver definations
  *
  *  Copyright (C) 2011 Intel Corporation
  *  Authors:	Vinod Koul <vinod.koul@linux.intel.com>
  *		Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
- *  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
- *
- *  This program is free software; you can redistribute it and/or modify
- *  it under the terms of the GNU General Public License as published by
- *  the Free Software Foundation; version 2 of the License.
- *
- *  This program is distributed in the hope that it will be useful, but
- *  WITHOUT ANY WARRANTY; without even the implied warranty of
- *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- *  General Public License for more details.
- *
- *  You should have received a copy of the GNU General Public License along
- *  with this program; if not, write to the Free Software Foundation, Inc.,
- *  59 Temple Place, Suite 330, Boston, MA 02111-1307 USA.
- *
- * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
- *
  */
+
 #ifndef __COMPRESS_DRIVER_H
 #define __COMPRESS_DRIVER_H
 

include/sound/dmaengine_pcm.h

@@ -1,17 +1,9 @@
-/*
+/* SPDX-License-Identifier: GPL-2.0+
+ *
  *  Copyright (C) 2012, Analog Devices Inc.
  *	Author: Lars-Peter Clausen <lars@metafoo.de>
- *
- *  This program is free software; you can redistribute it and/or modify it
- *  under  the terms of the GNU General  Public License as published by the
- *  Free Software Foundation;  either version 2 of the License, or (at your
- *  option) any later version.
- *
- *  You should have received a copy of the GNU General Public License along
- *  with this program; if not, write to the Free Software Foundation, Inc.,
- *  675 Mass Ave, Cambridge, MA 02139, USA.
- *
  */
+
 #ifndef __SOUND_DMAENGINE_PCM_H__
 #define __SOUND_DMAENGINE_PCM_H__
 

include/sound/hda_component.h

@@ -0,0 +1,61 @@
+// SPDX-License-Identifier: GPL-2.0
+// HD-Audio helpers to sync with DRM driver
+
+#ifndef __SOUND_HDA_COMPONENT_H
+#define __SOUND_HDA_COMPONENT_H
+
+#include <drm/drm_audio_component.h>
+
+#ifdef CONFIG_SND_HDA_COMPONENT
+int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable);
+int snd_hdac_display_power(struct hdac_bus *bus, bool enable);
+int snd_hdac_sync_audio_rate(struct hdac_device *codec, hda_nid_t nid,
+			     int dev_id, int rate);
+int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, int dev_id,
+			   bool *audio_enabled, char *buffer, int max_bytes);
+int snd_hdac_acomp_init(struct hdac_bus *bus,
+			const struct drm_audio_component_audio_ops *aops,
+			int (*match_master)(struct device *, void *),
+			size_t extra_size);
+int snd_hdac_acomp_exit(struct hdac_bus *bus);
+int snd_hdac_acomp_register_notifier(struct hdac_bus *bus,
+				    const struct drm_audio_component_audio_ops *ops);
+#else
+static inline int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable)
+{
+	return 0;
+}
+static inline int snd_hdac_display_power(struct hdac_bus *bus, bool enable)
+{
+	return 0;
+}
+static inline int snd_hdac_sync_audio_rate(struct hdac_device *codec,
+					   hda_nid_t nid, int dev_id, int rate)
+{
+	return 0;
+}
+static inline int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid,
+					 int dev_id, bool *audio_enabled,
+					 char *buffer, int max_bytes)
+{
+	return -ENODEV;
+}
+static inline int snd_hdac_acomp_init(struct hdac_bus *bus,
+				      const struct drm_audio_component_audio_ops *aops,
+				      int (*match_master)(struct device *, void *),
+				      size_t extra_size)
+{
+	return -ENODEV;
+}
+static inline int snd_hdac_acomp_exit(struct hdac_bus *bus)
+{
+	return 0;
+}
+static inline int snd_hdac_acomp_register_notifier(struct hdac_bus *bus,
+						  const struct drm_audio_component_audio_ops *ops)
+{
+	return -ENODEV;
+}
+#endif
+
+#endif /* __SOUND_HDA_COMPONENT_H */

include/sound/hda_i915.h

@@ -5,54 +5,23 @@
 #ifndef __SOUND_HDA_I915_H
 #define __SOUND_HDA_I915_H
 
-#include <drm/i915_component.h>
+#include "hda_component.h"
 
 #ifdef CONFIG_SND_HDA_I915
-int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable);
-int snd_hdac_display_power(struct hdac_bus *bus, bool enable);
 void snd_hdac_i915_set_bclk(struct hdac_bus *bus);
-int snd_hdac_sync_audio_rate(struct hdac_device *codec, hda_nid_t nid,
-			     int dev_id, int rate);
-int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, int dev_id,
-			   bool *audio_enabled, char *buffer, int max_bytes);
 int snd_hdac_i915_init(struct hdac_bus *bus);
-int snd_hdac_i915_exit(struct hdac_bus *bus);
-int snd_hdac_i915_register_notifier(const struct i915_audio_component_audio_ops *);
 #else
-static inline int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable)
-{
-	return 0;
-}
-static inline int snd_hdac_display_power(struct hdac_bus *bus, bool enable)
-{
-	return 0;
-}
 static inline void snd_hdac_i915_set_bclk(struct hdac_bus *bus)
 {
 }
-static inline int snd_hdac_sync_audio_rate(struct hdac_device *codec,
-					   hda_nid_t nid, int dev_id, int rate)
-{
-	return 0;
-}
-static inline int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid,
-					 int dev_id, bool *audio_enabled,
-					 char *buffer, int max_bytes)
-{
-	return -ENODEV;
-}
 static inline int snd_hdac_i915_init(struct hdac_bus *bus)
 {
 	return -ENODEV;
 }
+#endif
 static inline int snd_hdac_i915_exit(struct hdac_bus *bus)
 {
-	return 0;
+	return snd_hdac_acomp_exit(bus);
 }
-static inline int snd_hdac_i915_register_notifier(const struct i915_audio_component_audio_ops *ops)
-{
-	return -ENODEV;
-}
-#endif
 
 #endif /* __SOUND_HDA_I915_H */

include/sound/hdaudio.h

@@ -8,8 +8,10 @@
 
 #include <linux/device.h>
 #include <linux/interrupt.h>
+#include <linux/pm_runtime.h>
 #include <linux/timecounter.h>
 #include <sound/core.h>
+#include <sound/pcm.h>
 #include <sound/memalloc.h>
 #include <sound/hda_verbs.h>
 #include <drm/i915_component.h>
@@ -132,7 +134,7 @@ int snd_hdac_get_sub_nodes(struct hdac_device *codec, hda_nid_t nid,
 			   hda_nid_t *start_id);
 unsigned int snd_hdac_calc_stream_format(unsigned int rate,
 					 unsigned int channels,
-					 unsigned int format,
+					 snd_pcm_format_t format,
 					 unsigned int maxbps,
 					 unsigned short spdif_ctls);
 int snd_hdac_query_supported_pcm(struct hdac_device *codec, hda_nid_t nid,
@@ -171,12 +173,38 @@ int snd_hdac_power_down(struct hdac_device *codec);
 int snd_hdac_power_up_pm(struct hdac_device *codec);
 int snd_hdac_power_down_pm(struct hdac_device *codec);
 int snd_hdac_keep_power_up(struct hdac_device *codec);
+
+/* call this at entering into suspend/resume callbacks in codec driver */
+static inline void snd_hdac_enter_pm(struct hdac_device *codec)
+{
+	atomic_inc(&codec->in_pm);
+}
+
+/* call this at leaving from suspend/resume callbacks in codec driver */
+static inline void snd_hdac_leave_pm(struct hdac_device *codec)
+{
+	atomic_dec(&codec->in_pm);
+}
+
+static inline bool snd_hdac_is_in_pm(struct hdac_device *codec)
+{
+	return atomic_read(&codec->in_pm);
+}
+
+static inline bool snd_hdac_is_power_on(struct hdac_device *codec)
+{
+	return !pm_runtime_suspended(&codec->dev);
+}
 #else
 static inline int snd_hdac_power_up(struct hdac_device *codec) { return 0; }
 static inline int snd_hdac_power_down(struct hdac_device *codec) { return 0; }
 static inline int snd_hdac_power_up_pm(struct hdac_device *codec) { return 0; }
 static inline int snd_hdac_power_down_pm(struct hdac_device *codec) { return 0; }
 static inline int snd_hdac_keep_power_up(struct hdac_device *codec) { return 0; }
+static inline void snd_hdac_enter_pm(struct hdac_device *codec) {}
+static inline void snd_hdac_leave_pm(struct hdac_device *codec) {}
+static inline bool snd_hdac_is_in_pm(struct hdac_device *codec) { return 0; }
+static inline bool snd_hdac_is_power_on(struct hdac_device *codec) { return 1; }
 #endif
 
 /*
@@ -188,6 +216,11 @@ struct hdac_driver {
 	const struct hda_device_id *id_table;
 	int (*match)(struct hdac_device *dev, struct hdac_driver *drv);
 	void (*unsol_event)(struct hdac_device *dev, unsigned int event);
+
+	/* fields used by ext bus APIs */
+	int (*probe)(struct hdac_device *dev);
+	int (*remove)(struct hdac_device *dev);
+	void (*shutdown)(struct hdac_device *dev);
 };
 
 #define drv_to_hdac_driver(_drv) container_of(_drv, struct hdac_driver, driver)
@@ -208,6 +241,14 @@ struct hdac_bus_ops {
 	int (*link_power)(struct hdac_bus *bus, bool enable);
 };
 
+/*
+ * ops used for ASoC HDA codec drivers
+ */
+struct hdac_ext_bus_ops {
+	int (*hdev_attach)(struct hdac_device *hdev);
+	int (*hdev_detach)(struct hdac_device *hdev);
+};
+
 /*
  * Lowlevel I/O operators
  */
@@ -250,11 +291,17 @@ struct hdac_rb {
  * @mlcap: MultiLink capabilities pointer
  * @gtscap: gts capabilities pointer
  * @drsmcap: dma resume capabilities pointer
+ * @num_streams: streams supported
+ * @idx: HDA link index
+ * @hlink_list: link list of HDA links
+ * @lock: lock for link mgmt
+ * @cmd_dma_state: state of cmd DMAs: CORB and RIRB
  */
 struct hdac_bus {
 	struct device *dev;
 	const struct hdac_bus_ops *ops;
 	const struct hdac_io_ops *io_ops;
+	const struct hdac_ext_bus_ops *ext_ops;
 
 	/* h/w resources */
 	unsigned long addr;
@@ -314,9 +361,19 @@ struct hdac_bus {
 	spinlock_t reg_lock;
 	struct mutex cmd_mutex;
 
-	/* i915 component interface */
-	struct i915_audio_component *audio_component;
-	int i915_power_refcount;
+	/* DRM component interface */
+	struct drm_audio_component *audio_component;
+	int drm_power_refcount;
+
+	/* parameters required for enhanced capabilities */
+	int num_streams;
+	int idx;
+
+	struct list_head hlink_list;
+
+	struct mutex lock;
+	bool cmd_dma_state;
+
 };
 
 int snd_hdac_bus_init(struct hdac_bus *bus, struct device *dev,

include/sound/hdaudio_ext.h

@@ -4,38 +4,16 @@
 
 #include <sound/hdaudio.h>
 
-/**
- * hdac_ext_bus: HDAC extended bus for extended HDA caps
- *
- * @bus: hdac bus
- * @num_streams: streams supported
- * @hlink_list: link list of HDA links
- * @lock: lock for link mgmt
- * @cmd_dma_state: state of cmd DMAs: CORB and RIRB
- */
-struct hdac_ext_bus {
-	struct hdac_bus bus;
-	int num_streams;
-	int idx;
-
-	struct list_head hlink_list;
-
-	struct mutex lock;
-	bool cmd_dma_state;
-};
-
-int snd_hdac_ext_bus_init(struct hdac_ext_bus *sbus, struct device *dev,
+int snd_hdac_ext_bus_init(struct hdac_bus *bus, struct device *dev,
 		      const struct hdac_bus_ops *ops,
-		      const struct hdac_io_ops *io_ops);
+		      const struct hdac_io_ops *io_ops,
+		      const struct hdac_ext_bus_ops *ext_ops);
 
-void snd_hdac_ext_bus_exit(struct hdac_ext_bus *sbus);
-int snd_hdac_ext_bus_device_init(struct hdac_ext_bus *sbus, int addr);
+void snd_hdac_ext_bus_exit(struct hdac_bus *bus);
+int snd_hdac_ext_bus_device_init(struct hdac_bus *bus, int addr,
+						struct hdac_device *hdev);
 void snd_hdac_ext_bus_device_exit(struct hdac_device *hdev);
-void snd_hdac_ext_bus_device_remove(struct hdac_ext_bus *ebus);
-
-#define ebus_to_hbus(ebus)	(&(ebus)->bus)
-#define hbus_to_ebus(_bus) \
-	container_of(_bus, struct hdac_ext_bus, bus)
+void snd_hdac_ext_bus_device_remove(struct hdac_bus *bus);
 
 #define HDA_CODEC_REV_EXT_ENTRY(_vid, _rev, _name, drv_data) \
 	{ .vendor_id = (_vid), .rev_id = (_rev), .name = (_name), \
@@ -44,14 +22,14 @@ void snd_hdac_ext_bus_device_remove(struct hdac_ext_bus *ebus);
 #define HDA_CODEC_EXT_ENTRY(_vid, _revid, _name, _drv_data) \
 	HDA_CODEC_REV_EXT_ENTRY(_vid, _revid, _name, _drv_data)
 
-void snd_hdac_ext_bus_ppcap_enable(struct hdac_ext_bus *chip, bool enable);
-void snd_hdac_ext_bus_ppcap_int_enable(struct hdac_ext_bus *chip, bool enable);
+void snd_hdac_ext_bus_ppcap_enable(struct hdac_bus *chip, bool enable);
+void snd_hdac_ext_bus_ppcap_int_enable(struct hdac_bus *chip, bool enable);
 
-void snd_hdac_ext_stream_spbcap_enable(struct hdac_ext_bus *chip,
+void snd_hdac_ext_stream_spbcap_enable(struct hdac_bus *chip,
 				 bool enable, int index);
 
-int snd_hdac_ext_bus_get_ml_capabilities(struct hdac_ext_bus *bus);
-struct hdac_ext_link *snd_hdac_ext_bus_get_link(struct hdac_ext_bus *bus,
+int snd_hdac_ext_bus_get_ml_capabilities(struct hdac_bus *bus);
+struct hdac_ext_link *snd_hdac_ext_bus_get_link(struct hdac_bus *bus,
 						const char *codec_name);
 
 enum hdac_ext_stream_type {
@@ -100,28 +78,28 @@ struct hdac_ext_stream {
 #define stream_to_hdac_ext_stream(s) \
 	container_of(s, struct hdac_ext_stream, hstream)
 
-void snd_hdac_ext_stream_init(struct hdac_ext_bus *bus,
+void snd_hdac_ext_stream_init(struct hdac_bus *bus,
 				struct hdac_ext_stream *stream, int idx,
 				int direction, int tag);
-int snd_hdac_ext_stream_init_all(struct hdac_ext_bus *ebus, int start_idx,
+int snd_hdac_ext_stream_init_all(struct hdac_bus *bus, int start_idx,
 		int num_stream, int dir);
-void snd_hdac_stream_free_all(struct hdac_ext_bus *ebus);
-void snd_hdac_link_free_all(struct hdac_ext_bus *ebus);
-struct hdac_ext_stream *snd_hdac_ext_stream_assign(struct hdac_ext_bus *bus,
+void snd_hdac_stream_free_all(struct hdac_bus *bus);
+void snd_hdac_link_free_all(struct hdac_bus *bus);
+struct hdac_ext_stream *snd_hdac_ext_stream_assign(struct hdac_bus *bus,
 					   struct snd_pcm_substream *substream,
 					   int type);
 void snd_hdac_ext_stream_release(struct hdac_ext_stream *azx_dev, int type);
-void snd_hdac_ext_stream_decouple(struct hdac_ext_bus *bus,
+void snd_hdac_ext_stream_decouple(struct hdac_bus *bus,
 				struct hdac_ext_stream *azx_dev, bool decouple);
-void snd_hdac_ext_stop_streams(struct hdac_ext_bus *sbus);
+void snd_hdac_ext_stop_streams(struct hdac_bus *bus);
 
-int snd_hdac_ext_stream_set_spib(struct hdac_ext_bus *ebus,
+int snd_hdac_ext_stream_set_spib(struct hdac_bus *bus,
 				 struct hdac_ext_stream *stream, u32 value);
-int snd_hdac_ext_stream_get_spbmaxfifo(struct hdac_ext_bus *ebus,
+int snd_hdac_ext_stream_get_spbmaxfifo(struct hdac_bus *bus,
 				 struct hdac_ext_stream *stream);
-void snd_hdac_ext_stream_drsm_enable(struct hdac_ext_bus *ebus,
+void snd_hdac_ext_stream_drsm_enable(struct hdac_bus *bus,
 				bool enable, int index);
-int snd_hdac_ext_stream_set_dpibr(struct hdac_ext_bus *ebus,
+int snd_hdac_ext_stream_set_dpibr(struct hdac_bus *bus,
 				struct hdac_ext_stream *stream, u32 value);
 int snd_hdac_ext_stream_set_lpib(struct hdac_ext_stream *stream, u32 value);
 
@@ -144,17 +122,15 @@ struct hdac_ext_link {
 
 int snd_hdac_ext_bus_link_power_up(struct hdac_ext_link *link);
 int snd_hdac_ext_bus_link_power_down(struct hdac_ext_link *link);
-int snd_hdac_ext_bus_link_power_up_all(struct hdac_ext_bus *ebus);
-int snd_hdac_ext_bus_link_power_down_all(struct hdac_ext_bus *ebus);
+int snd_hdac_ext_bus_link_power_up_all(struct hdac_bus *bus);
+int snd_hdac_ext_bus_link_power_down_all(struct hdac_bus *bus);
 void snd_hdac_ext_link_set_stream_id(struct hdac_ext_link *link,
 				 int stream);
 void snd_hdac_ext_link_clear_stream_id(struct hdac_ext_link *link,
 				 int stream);
 
-int snd_hdac_ext_bus_link_get(struct hdac_ext_bus *ebus,
-				struct hdac_ext_link *link);
-int snd_hdac_ext_bus_link_put(struct hdac_ext_bus *ebus,
-				struct hdac_ext_link *link);
+int snd_hdac_ext_bus_link_get(struct hdac_bus *bus, struct hdac_ext_link *link);
+int snd_hdac_ext_bus_link_put(struct hdac_bus *bus, struct hdac_ext_link *link);
 
 /* update register macro */
 #define snd_hdac_updatel(addr, reg, mask, val)		\
@@ -181,53 +157,12 @@ struct hda_dai_map {
 	u32	maxbps;
 };
 
-#define HDA_MAX_NIDS 16
-
-/**
- * struct hdac_ext_device - HDAC Ext device
- *
- * @hdac: hdac core device
- * @nid_list - the dai map which matches the dai-name with the nid
- * @map_cur_idx - the idx in use in dai_map
- * @ops - the hda codec ops common to all codec drivers
- * @pvt_data - private data, for asoc contains asoc codec object
- */
-struct hdac_ext_device {
-	struct hdac_device hdev;
-	struct hdac_ext_bus *ebus;
-
-	/* soc-dai to nid map */
-	struct hda_dai_map nid_list[HDA_MAX_NIDS];
-	unsigned int map_cur_idx;
-
-	/* codec ops */
-	struct hdac_ext_codec_ops ops;
-
-	struct snd_card *card;
-	void *scodec;
-	void *private_data;
-};
-
 struct hdac_ext_dma_params {
 	u32 format;
 	u8 stream_tag;
 };
-#define to_ehdac_device(dev) (container_of((dev), \
-				 struct hdac_ext_device, hdev))
-/*
- * HD-audio codec base driver
- */
-struct hdac_ext_driver {
-	struct hdac_driver hdac;
-
-	int	(*probe)(struct hdac_ext_device *dev);
-	int	(*remove)(struct hdac_ext_device *dev);
-	void	(*shutdown)(struct hdac_ext_device *dev);
-};
-
-int snd_hda_ext_driver_register(struct hdac_ext_driver *drv);
-void snd_hda_ext_driver_unregister(struct hdac_ext_driver *drv);
 
-#define to_ehdac_driver(_drv) container_of(_drv, struct hdac_ext_driver, hdac)
+int snd_hda_ext_driver_register(struct hdac_driver *drv);
+void snd_hda_ext_driver_unregister(struct hdac_driver *drv);
 
 #endif /* __SOUND_HDAUDIO_EXT_H */

include/sound/memalloc.h

@@ -24,6 +24,8 @@
 #ifndef __SOUND_MEMALLOC_H
 #define __SOUND_MEMALLOC_H
 
+#include <asm/page.h>
+
 struct device;
 
 /*
@@ -67,6 +69,14 @@ struct snd_dma_buffer {
 	void *private_data;	/* private for allocator; don't touch */
 };
 
+/*
+ * return the pages matching with the given byte size
+ */
+static inline unsigned int snd_sgbuf_aligned_pages(size_t size)
+{
+	return (size + PAGE_SIZE - 1) >> PAGE_SHIFT;
+}
+
 #ifdef CONFIG_SND_DMA_SGBUF
 /*
  * Scatter-Gather generic device pages
@@ -90,14 +100,6 @@ struct snd_sg_buf {
 	struct device *dev;
 };
 
-/*
- * return the pages matching with the given byte size
- */
-static inline unsigned int snd_sgbuf_aligned_pages(size_t size)
-{
-	return (size + PAGE_SIZE - 1) >> PAGE_SHIFT;
-}
-
 /*
  * return the physical address at the corresponding offset
  */

include/sound/pcm.h

@@ -462,6 +462,7 @@ struct snd_pcm_substream {
         /* -- timer section -- */
 	struct snd_timer *timer;		/* timer */
 	unsigned timer_running: 1;	/* time is running */
+	long wait_time;	/* time in ms for R/W to wait for avail */
 	/* -- next substream -- */
 	struct snd_pcm_substream *next;
 	/* -- linked substreams -- */
@@ -1089,14 +1090,14 @@ static inline snd_pcm_sframes_t
 snd_pcm_lib_write(struct snd_pcm_substream *substream,
 		  const void __user *buf, snd_pcm_uframes_t frames)
 {
-	return __snd_pcm_lib_xfer(substream, (void *)buf, true, frames, false);
+	return __snd_pcm_lib_xfer(substream, (void __force *)buf, true, frames, false);
 }
 
 static inline snd_pcm_sframes_t
 snd_pcm_lib_read(struct snd_pcm_substream *substream,
 		 void __user *buf, snd_pcm_uframes_t frames)
 {
-	return __snd_pcm_lib_xfer(substream, (void *)buf, true, frames, false);
+	return __snd_pcm_lib_xfer(substream, (void __force *)buf, true, frames, false);
 }
 
 static inline snd_pcm_sframes_t
@@ -1341,8 +1342,6 @@ int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream, struct vm_area_s
 #define snd_pcm_lib_mmap_iomem	NULL
 #endif
 
-#define snd_pcm_lib_mmap_vmalloc NULL
-
 /**
  * snd_pcm_limit_isa_dma_size - Get the max size fitting with ISA DMA transfer
  * @dma: DMA number

include/sound/pcm_params.h

@@ -87,6 +87,13 @@ static inline void snd_mask_set(struct snd_mask *mask, unsigned int val)
 	mask->bits[MASK_OFS(val)] |= MASK_BIT(val);
 }
 
+/* Most of drivers need only this one */
+static inline void snd_mask_set_format(struct snd_mask *mask,
+				       snd_pcm_format_t format)
+{
+	snd_mask_set(mask, (__force unsigned int)format);
+}
+
 static inline void snd_mask_reset(struct snd_mask *mask, unsigned int val)
 {
 	mask->bits[MASK_OFS(val)] &= ~MASK_BIT(val);
@@ -369,8 +376,7 @@ static inline int params_physical_width(const struct snd_pcm_hw_params *p)
 static inline void
 params_set_format(struct snd_pcm_hw_params *p, snd_pcm_format_t fmt)
 {
-	snd_mask_set(hw_param_mask(p, SNDRV_PCM_HW_PARAM_FORMAT),
-		(__force int)fmt);
+	snd_mask_set_format(hw_param_mask(p, SNDRV_PCM_HW_PARAM_FORMAT), fmt);
 }
 
 #endif /* __SOUND_PCM_PARAMS_H */

include/sound/pxa2xx-lib.h

@@ -8,20 +8,23 @@
 /* PCM */
 struct snd_pcm_substream;
 struct snd_pcm_hw_params;
+struct snd_soc_pcm_runtime;
 struct snd_pcm;
 
-extern int __pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream,
+extern int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream,
 				struct snd_pcm_hw_params *params);
-extern int __pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream);
+extern int pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream);
 extern int pxa2xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd);
 extern snd_pcm_uframes_t pxa2xx_pcm_pointer(struct snd_pcm_substream *substream);
-extern int __pxa2xx_pcm_prepare(struct snd_pcm_substream *substream);
-extern int __pxa2xx_pcm_open(struct snd_pcm_substream *substream);
-extern int __pxa2xx_pcm_close(struct snd_pcm_substream *substream);
+extern int pxa2xx_pcm_prepare(struct snd_pcm_substream *substream);
+extern int pxa2xx_pcm_open(struct snd_pcm_substream *substream);
+extern int pxa2xx_pcm_close(struct snd_pcm_substream *substream);
 extern int pxa2xx_pcm_mmap(struct snd_pcm_substream *substream,
 	struct vm_area_struct *vma);
 extern int pxa2xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream);
 extern void pxa2xx_pcm_free_dma_buffers(struct snd_pcm *pcm);
+extern int pxa2xx_soc_pcm_new(struct snd_soc_pcm_runtime *rtd);
+extern const struct snd_pcm_ops pxa2xx_pcm_ops;
 
 /* AC97 */
 

include/sound/rt5682.h

@@ -0,0 +1,40 @@
+/*
+ * linux/sound/rt5682.h -- Platform data for RT5682
+ *
+ * Copyright 2018 Realtek Microelectronics
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __LINUX_SND_RT5682_H
+#define __LINUX_SND_RT5682_H
+
+enum rt5682_dmic1_data_pin {
+	RT5682_DMIC1_NULL,
+	RT5682_DMIC1_DATA_GPIO2,
+	RT5682_DMIC1_DATA_GPIO5,
+};
+
+enum rt5682_dmic1_clk_pin {
+	RT5682_DMIC1_CLK_GPIO1,
+	RT5682_DMIC1_CLK_GPIO3,
+};
+
+enum rt5682_jd_src {
+	RT5682_JD_NULL,
+	RT5682_JD1,
+};
+
+struct rt5682_platform_data {
+
+	int ldo1_en; /* GPIO for LDO1_EN */
+
+	enum rt5682_dmic1_data_pin dmic1_data_pin;
+	enum rt5682_dmic1_clk_pin dmic1_clk_pin;
+	enum rt5682_jd_src jd_src;
+};
+
+#endif
+

include/sound/sb16_csp.h

@@ -46,7 +46,7 @@ enum {
 struct snd_sb_csp_ops {
 	int (*csp_use) (struct snd_sb_csp * p);
 	int (*csp_unuse) (struct snd_sb_csp * p);
-	int (*csp_autoload) (struct snd_sb_csp * p, int pcm_sfmt, int play_rec_mode);
+	int (*csp_autoload) (struct snd_sb_csp * p, snd_pcm_format_t pcm_sfmt, int play_rec_mode);
 	int (*csp_start) (struct snd_sb_csp * p, int sample_width, int channels);
 	int (*csp_stop) (struct snd_sb_csp * p);
 	int (*csp_qsound_transfer) (struct snd_sb_csp * p);

include/sound/seq_midi_event.h

@@ -43,10 +43,8 @@ void snd_midi_event_free(struct snd_midi_event *dev);
 void snd_midi_event_reset_encode(struct snd_midi_event *dev);
 void snd_midi_event_reset_decode(struct snd_midi_event *dev);
 void snd_midi_event_no_status(struct snd_midi_event *dev, int on);
-/* encode from byte stream - return number of written bytes if success */
-long snd_midi_event_encode(struct snd_midi_event *dev, unsigned char *buf, long count,
-			   struct snd_seq_event *ev);
-int snd_midi_event_encode_byte(struct snd_midi_event *dev, int c, struct snd_seq_event *ev);
+bool snd_midi_event_encode_byte(struct snd_midi_event *dev, unsigned char c,
+				struct snd_seq_event *ev);
 /* decode from event to bytes - return number of written bytes if success */
 long snd_midi_event_decode(struct snd_midi_event *dev, unsigned char *buf, long count,
 			   struct snd_seq_event *ev);

include/sound/seq_virmidi.h

@@ -36,11 +36,12 @@ struct snd_virmidi {
 	int seq_mode;
 	int client;
 	int port;
-	unsigned int trigger: 1;
+	bool trigger;
 	struct snd_midi_event *parser;
 	struct snd_seq_event event;
 	struct snd_virmidi_dev *rdev;
 	struct snd_rawmidi_substream *substream;
+	struct work_struct output_work;
 };
 
 #define SNDRV_VIRMIDI_SUBSCRIBE		(1<<0)

include/sound/sh_fsi.h

@@ -1,16 +1,13 @@
-#ifndef __SOUND_FSI_H
-#define __SOUND_FSI_H
-
-/*
+/* SPDX-License-Identifier: GPL-2.0
+ *
  * Fifo-attached Serial Interface (FSI) support for SH7724
  *
  * Copyright (C) 2009 Renesas Solutions Corp.
  * Kuninori Morimoto <morimoto.kuninori@renesas.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
  */
+#ifndef __SOUND_FSI_H
+#define __SOUND_FSI_H
+
 #include <linux/clk.h>
 #include <sound/soc.h>
 

include/sound/simple_card.h

@@ -1,12 +1,9 @@
-/*
+/* SPDX-License-Identifier: GPL-2.0
+ *
  * ASoC simple sound card support
  *
  * Copyright (C) 2012 Renesas Solutions Corp.
  * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
  */
 
 #ifndef __SIMPLE_CARD_H

include/sound/simple_card_utils.h

@@ -1,17 +1,20 @@
-/*
+/* SPDX-License-Identifier: GPL-2.0
+ *
  * simple_card_utils.h
  *
  * Copyright (c) 2016 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
  */
+
 #ifndef __SIMPLE_CARD_UTILS_H
 #define __SIMPLE_CARD_UTILS_H
 
 #include <sound/soc.h>
 
+#define asoc_simple_card_init_hp(card, sjack, prefix) \
+	asoc_simple_card_init_jack(card, sjack, 1, prefix)
+#define asoc_simple_card_init_mic(card, sjack, prefix) \
+	asoc_simple_card_init_jack(card, sjack, 0, prefix)
+
 struct asoc_simple_dai {
 	const char *name;
 	unsigned int sysclk;
@@ -28,6 +31,12 @@ struct asoc_simple_card_data {
 	u32 convert_channels;
 };
 
+struct asoc_simple_jack {
+	struct snd_soc_jack jack;
+	struct snd_soc_jack_pin pin;
+	struct snd_soc_jack_gpio gpio;
+};
+
 int asoc_simple_card_parse_daifmt(struct device *dev,
 				  struct device_node *node,
 				  struct device_node *codec,
@@ -107,4 +116,8 @@ int asoc_simple_card_of_parse_routing(struct snd_soc_card *card,
 int asoc_simple_card_of_parse_widgets(struct snd_soc_card *card,
 				      char *prefix);
 
+int asoc_simple_card_init_jack(struct snd_soc_card *card,
+			       struct asoc_simple_jack *sjack,
+			       int is_hp, char *prefix);
+
 #endif /* __SIMPLE_CARD_UTILS_H */

include/sound/soc-acpi-intel-match.h

@@ -1,16 +1,6 @@
-
-/*
- * Copyright (C) 2017, Intel Corporation. All rights reserved.
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License version
- * 2 as published by the Free Software Foundation.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
+/* SPDX-License-Identifier: GPL-2.0
  *
+ * Copyright (C) 2017, Intel Corporation. All rights reserved.
  */
 
 #ifndef __LINUX_SND_SOC_ACPI_INTEL_MATCH_H
@@ -29,5 +19,10 @@ extern struct snd_soc_acpi_mach snd_soc_acpi_intel_broadwell_machines[];
 extern struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_legacy_machines[];
 extern struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_machines[];
 extern struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[];
+extern struct snd_soc_acpi_mach snd_soc_acpi_intel_skl_machines[];
+extern struct snd_soc_acpi_mach snd_soc_acpi_intel_kbl_machines[];
+extern struct snd_soc_acpi_mach snd_soc_acpi_intel_bxt_machines[];
+extern struct snd_soc_acpi_mach snd_soc_acpi_intel_glk_machines[];
+extern struct snd_soc_acpi_mach snd_soc_acpi_intel_cnl_machines[];
 
 #endif

include/sound/soc-acpi.h

@@ -1,15 +1,6 @@
-/*
- * Copyright (C) 2013-15, Intel Corporation. All rights reserved.
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License version
- * 2 as published by the Free Software Foundation.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
+/* SPDX-License-Identifier: GPL-2.0
  *
+ * Copyright (C) 2013-15, Intel Corporation. All rights reserved.
  */
 
 #ifndef __LINUX_SND_SOC_ACPI_H

include/sound/soc-dai.h

@@ -1,12 +1,9 @@
-/*
+/* SPDX-License-Identifier: GPL-2.0
+ *
  * linux/sound/soc-dai.h -- ALSA SoC Layer
  *
  * Copyright:	2005-2008 Wolfson Microelectronics. PLC.
  *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- *
  * Digital Audio Interface (DAI) API.
  */
 
@@ -141,6 +138,11 @@ int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate);
 int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute,
 			     int direction);
 
+
+int snd_soc_dai_get_channel_map(struct snd_soc_dai *dai,
+		unsigned int *tx_num, unsigned int *tx_slot,
+		unsigned int *rx_num, unsigned int *rx_slot);
+
 int snd_soc_dai_is_dummy(struct snd_soc_dai *dai);
 
 struct snd_soc_dai_ops {
@@ -168,6 +170,9 @@ struct snd_soc_dai_ops {
 	int (*set_channel_map)(struct snd_soc_dai *dai,
 		unsigned int tx_num, unsigned int *tx_slot,
 		unsigned int rx_num, unsigned int *rx_slot);
+	int (*get_channel_map)(struct snd_soc_dai *dai,
+			unsigned int *tx_num, unsigned int *tx_slot,
+			unsigned int *rx_num, unsigned int *rx_slot);
 	int (*set_tristate)(struct snd_soc_dai *dai, int tristate);
 
 	int (*set_sdw_stream)(struct snd_soc_dai *dai,

include/sound/soc-dapm.h

@@ -1,13 +1,10 @@
-/*
+/* SPDX-License-Identifier: GPL-2.0
+ *
  * linux/sound/soc-dapm.h -- ALSA SoC Dynamic Audio Power Management
  *
- * Author:		Liam Girdwood
- * Created:		Aug 11th 2005
+ * Author:	Liam Girdwood
+ * Created:	Aug 11th 2005
  * Copyright:	Wolfson Microelectronics. PLC.
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
  */
 
 #ifndef __LINUX_SND_SOC_DAPM_H

include/sound/soc-dpcm.h

@@ -1,11 +1,8 @@
-/*
+/* SPDX-License-Identifier: GPL-2.0
+ *
  * linux/sound/soc-dpcm.h -- ALSA SoC Dynamic PCM Support
  *
  * Author:		Liam Girdwood <lrg@ti.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
  */
 
 #ifndef __LINUX_SND_SOC_DPCM_H

include/sound/soc-topology.h

@@ -1,13 +1,10 @@
-/*
+/* SPDX-License-Identifier: GPL-2.0
+ *
  * linux/sound/soc-topology.h -- ALSA SoC Firmware Controls and DAPM
  *
  * Copyright (C) 2012 Texas Instruments Inc.
  * Copyright (C) 2015 Intel Corporation.
  *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- *
  * Simple file API to load FW that includes mixers, coefficients, DAPM graphs,
  * algorithms, equalisers, DAIs, widgets, FE caps, BE caps, codec link caps etc.
  */
@@ -30,6 +27,9 @@ struct snd_soc_dapm_context;
 struct snd_soc_card;
 struct snd_kcontrol_new;
 struct snd_soc_dai_link;
+struct snd_soc_dai_driver;
+struct snd_soc_dai;
+struct snd_soc_dapm_route;
 
 /* object scan be loaded and unloaded in groups with identfying indexes */
 #define SND_SOC_TPLG_INDEX_ALL	0	/* ID that matches all FW objects */
@@ -109,35 +109,44 @@ struct snd_soc_tplg_widget_events {
 struct snd_soc_tplg_ops {
 
 	/* external kcontrol init - used for any driver specific init */
-	int (*control_load)(struct snd_soc_component *,
+	int (*control_load)(struct snd_soc_component *, int index,
 		struct snd_kcontrol_new *, struct snd_soc_tplg_ctl_hdr *);
 	int (*control_unload)(struct snd_soc_component *,
 		struct snd_soc_dobj *);
 
+	/* DAPM graph route element loading and unloading */
+	int (*dapm_route_load)(struct snd_soc_component *, int index,
+		struct snd_soc_dapm_route *route);
+	int (*dapm_route_unload)(struct snd_soc_component *,
+		struct snd_soc_dobj *);
+
 	/* external widget init - used for any driver specific init */
-	int (*widget_load)(struct snd_soc_component *,
+	int (*widget_load)(struct snd_soc_component *, int index,
 		struct snd_soc_dapm_widget *,
 		struct snd_soc_tplg_dapm_widget *);
-	int (*widget_ready)(struct snd_soc_component *,
+	int (*widget_ready)(struct snd_soc_component *, int index,
 		struct snd_soc_dapm_widget *,
 		struct snd_soc_tplg_dapm_widget *);
 	int (*widget_unload)(struct snd_soc_component *,
 		struct snd_soc_dobj *);
 
 	/* FE DAI - used for any driver specific init */
-	int (*dai_load)(struct snd_soc_component *,
-		struct snd_soc_dai_driver *dai_drv);
+	int (*dai_load)(struct snd_soc_component *, int index,
+		struct snd_soc_dai_driver *dai_drv,
+		struct snd_soc_tplg_pcm *pcm, struct snd_soc_dai *dai);
+
 	int (*dai_unload)(struct snd_soc_component *,
 		struct snd_soc_dobj *);
 
 	/* DAI link - used for any driver specific init */
-	int (*link_load)(struct snd_soc_component *,
-		struct snd_soc_dai_link *link);
+	int (*link_load)(struct snd_soc_component *, int index,
+		struct snd_soc_dai_link *link,
+		struct snd_soc_tplg_link_config *cfg);
 	int (*link_unload)(struct snd_soc_component *,
 		struct snd_soc_dobj *);
 
 	/* callback to handle vendor bespoke data */
-	int (*vendor_load)(struct snd_soc_component *,
+	int (*vendor_load)(struct snd_soc_component *, int index,
 		struct snd_soc_tplg_hdr *);
 	int (*vendor_unload)(struct snd_soc_component *,
 		struct snd_soc_tplg_hdr *);
@@ -146,7 +155,7 @@ struct snd_soc_tplg_ops {
 	void (*complete)(struct snd_soc_component *);
 
 	/* manifest - optional to inform component of manifest */
-	int (*manifest)(struct snd_soc_component *,
+	int (*manifest)(struct snd_soc_component *, int index,
 		struct snd_soc_tplg_manifest *);
 
 	/* vendor specific kcontrol handlers available for binding */

include/sound/soc.h

@@ -1,13 +1,10 @@
-/*
+/* SPDX-License-Identifier: GPL-2.0
+ *
  * linux/sound/soc.h -- ALSA SoC Layer
  *
- * Author:		Liam Girdwood
- * Created:		Aug 11th 2005
+ * Author:	Liam Girdwood
+ * Created:	Aug 11th 2005
  * Copyright:	Wolfson Microelectronics. PLC.
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
  */
 
 #ifndef __LINUX_SND_SOC_H
@@ -806,6 +803,14 @@ struct snd_soc_component_driver {
 	unsigned int use_pmdown_time:1; /* care pmdown_time at stop */
 	unsigned int endianness:1;
 	unsigned int non_legacy_dai_naming:1;
+
+	/* this component uses topology and ignore machine driver FEs */
+	const char *ignore_machine;
+	const char *topology_name_prefix;
+	int (*be_hw_params_fixup)(struct snd_soc_pcm_runtime *rtd,
+				  struct snd_pcm_hw_params *params);
+	bool use_dai_pcm_id;	/* use the DAI link PCM ID as PCM device number */
+	int be_pcm_base;	/* base device ID for all BE PCMs */
 };
 
 struct snd_soc_component {
@@ -957,10 +962,17 @@ struct snd_soc_dai_link {
 
 	/* DPCM used FE & BE merged format */
 	unsigned int dpcm_merged_format:1;
+	/* DPCM used FE & BE merged channel */
+	unsigned int dpcm_merged_chan:1;
+	/* DPCM used FE & BE merged rate */
+	unsigned int dpcm_merged_rate:1;
 
 	/* pmdown_time is ignored at stop */
 	unsigned int ignore_pmdown_time:1;
 
+	/* Do not create a PCM for this DAI link (Backend link) */
+	unsigned int ignore:1;
+
 	struct list_head list; /* DAI link list of the soc card */
 	struct snd_soc_dobj dobj; /* For topology */
 };
@@ -1000,6 +1012,7 @@ struct snd_soc_card {
 	const char *long_name;
 	const char *driver_name;
 	char dmi_longname[80];
+	char topology_shortname[32];
 
 	struct device *dev;
 	struct snd_card *snd_card;
@@ -1009,6 +1022,7 @@ struct snd_soc_card {
 	struct mutex dapm_mutex;
 
 	bool instantiated;
+	bool topology_shortname_created;
 
 	int (*probe)(struct snd_soc_card *card);
 	int (*late_probe)(struct snd_soc_card *card);
@@ -1412,6 +1426,9 @@ int snd_soc_of_parse_card_name(struct snd_soc_card *card,
 			       const char *propname);
 int snd_soc_of_parse_audio_simple_widgets(struct snd_soc_card *card,
 					  const char *propname);
+int snd_soc_of_get_slot_mask(struct device_node *np,
+			     const char *prop_name,
+			     unsigned int *mask);
 int snd_soc_of_parse_tdm_slot(struct device_node *np,
 			      unsigned int *tx_mask,
 			      unsigned int *rx_mask,

include/trace/events/clk.h

@@ -192,6 +192,42 @@ DEFINE_EVENT(clk_phase, clk_set_phase_complete,
 	TP_ARGS(core, phase)
 );
 
+DECLARE_EVENT_CLASS(clk_duty_cycle,
+
+	TP_PROTO(struct clk_core *core, struct clk_duty *duty),
+
+	TP_ARGS(core, duty),
+
+	TP_STRUCT__entry(
+		__string(        name,           core->name              )
+		__field( unsigned int,           num                     )
+		__field( unsigned int,           den                     )
+	),
+
+	TP_fast_assign(
+		__assign_str(name, core->name);
+		__entry->num = duty->num;
+		__entry->den = duty->den;
+	),
+
+	TP_printk("%s %u/%u", __get_str(name), (unsigned int)__entry->num,
+		  (unsigned int)__entry->den)
+);
+
+DEFINE_EVENT(clk_duty_cycle, clk_set_duty_cycle,
+
+	TP_PROTO(struct clk_core *core, struct clk_duty *duty),
+
+	TP_ARGS(core, duty)
+);
+
+DEFINE_EVENT(clk_duty_cycle, clk_set_duty_cycle_complete,
+
+	TP_PROTO(struct clk_core *core, struct clk_duty *duty),
+
+	TP_ARGS(core, duty)
+);
+
 #endif /* _TRACE_CLK_H */
 
 /* This part must be outside protection */

include/uapi/linux/usb/audio.h

@@ -390,33 +390,64 @@ static inline __u8 uac_processing_unit_iChannelNames(struct uac_processing_unit_
 static inline __u8 uac_processing_unit_bControlSize(struct uac_processing_unit_descriptor *desc,
 						    int protocol)
 {
-	return (protocol == UAC_VERSION_1) ?
-		desc->baSourceID[desc->bNrInPins + 4] :
-		2; /* in UAC2, this value is constant */
+	switch (protocol) {
+	case UAC_VERSION_1:
+		return desc->baSourceID[desc->bNrInPins + 4];
+	case UAC_VERSION_2:
+		return 2; /* 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_processing_unit_bmControls(struct uac_processing_unit_descriptor *desc,
 						   int protocol)
 {
-	return (protocol == UAC_VERSION_1) ?
-		&desc->baSourceID[desc->bNrInPins + 5] :
-		&desc->baSourceID[desc->bNrInPins + 6];
+	switch (protocol) {
+	case UAC_VERSION_1:
+		return &desc->baSourceID[desc->bNrInPins + 5];
+	case UAC_VERSION_2:
+		return &desc->baSourceID[desc->bNrInPins + 6];
+	case UAC_VERSION_3:
+		return &desc->baSourceID[desc->bNrInPins + 2];
+	default:
+		return NULL;
+	}
 }
 
 static inline __u8 uac_processing_unit_iProcessing(struct uac_processing_unit_descriptor *desc,
 						   int protocol)
 {
 	__u8 control_size = uac_processing_unit_bControlSize(desc, protocol);
-	return *(uac_processing_unit_bmControls(desc, protocol)
-			+ control_size);
+
+	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 */
+	}
 }
 
 static inline __u8 *uac_processing_unit_specific(struct uac_processing_unit_descriptor *desc,
 						 int protocol)
 {
 	__u8 control_size = uac_processing_unit_bControlSize(desc, protocol);
-	return uac_processing_unit_bmControls(desc, protocol)
+
+	switch (protocol) {
+	case UAC_VERSION_1:
+	case UAC_VERSION_2:
+	default:
+		return uac_processing_unit_bmControls(desc, protocol)
 			+ control_size + 1;
+	case UAC_VERSION_3:
+		return uac_processing_unit_bmControls(desc, protocol)
+			+ control_size;
+	}
 }
 
 /* 4.5.2 Class-Specific AS Interface Descriptor */

sound/ac97/bus.c

@@ -13,6 +13,7 @@
 #include <linux/idr.h>
 #include <linux/list.h>
 #include <linux/mutex.h>
+#include <linux/of.h>
 #include <linux/pm.h>
 #include <linux/pm_runtime.h>
 #include <linux/slab.h>
@@ -68,6 +69,27 @@ ac97_codec_find(struct ac97_controller *ac97_ctrl, unsigned int codec_num)
 	return ac97_ctrl->codecs[codec_num];
 }
 
+static struct device_node *
+ac97_of_get_child_device(struct ac97_controller *ac97_ctrl, int idx,
+			 unsigned int vendor_id)
+{
+	struct device_node *node;
+	u32 reg;
+	char compat[] = "ac97,0000,0000";
+
+	snprintf(compat, sizeof(compat), "ac97,%04x,%04x",
+		 vendor_id >> 16, vendor_id & 0xffff);
+
+	for_each_child_of_node(ac97_ctrl->parent->of_node, node) {
+		if ((idx != of_property_read_u32(node, "reg", &reg)) ||
+		    !of_device_is_compatible(node, compat))
+			continue;
+		return of_node_get(node);
+	}
+
+	return NULL;
+}
+
 static void ac97_codec_release(struct device *dev)
 {
 	struct ac97_codec_device *adev;
@@ -76,6 +98,7 @@ static void ac97_codec_release(struct device *dev)
 	adev = to_ac97_device(dev);
 	ac97_ctrl = adev->ac97_ctrl;
 	ac97_ctrl->codecs[adev->num] = NULL;
+	of_node_put(dev->of_node);
 	kfree(adev);
 }
 
@@ -98,6 +121,8 @@ static int ac97_codec_add(struct ac97_controller *ac97_ctrl, int idx,
 
 	device_initialize(&codec->dev);
 	dev_set_name(&codec->dev, "%s:%u", dev_name(ac97_ctrl->parent), idx);
+	codec->dev.of_node = ac97_of_get_child_device(ac97_ctrl, idx,
+						      vendor_id);
 
 	ret = device_add(&codec->dev);
 	if (ret)
@@ -105,6 +130,7 @@ static int ac97_codec_add(struct ac97_controller *ac97_ctrl, int idx,
 
 	return 0;
 err_free_codec:
+	of_node_put(codec->dev.of_node);
 	put_device(&codec->dev);
 	kfree(codec);
 	ac97_ctrl->codecs[idx] = NULL;

sound/aoa/core/gpio-feature.c

@@ -88,8 +88,10 @@ static struct device_node *get_gpio(char *name,
 	}
 
 	reg = of_get_property(np, "reg", NULL);
-	if (!reg)
+	if (!reg) {
+		of_node_put(np);
 		return NULL;
+	}
 
 	*gpioptr = *reg;
 

sound/arm/Kconfig

@@ -17,14 +17,9 @@ config SND_ARMAACI
 	select SND_PCM
 	select SND_AC97_CODEC
 
-config SND_PXA2XX_PCM
-	tristate
-	select SND_PCM
-
 config SND_PXA2XX_AC97
 	tristate "AC97 driver for the Intel PXA2xx chip"
 	depends on ARCH_PXA
-	select SND_PXA2XX_PCM
 	select SND_AC97_CODEC
 	select SND_PXA2XX_LIB
 	select SND_PXA2XX_LIB_AC97

sound/arm/Makefile

@@ -6,9 +6,6 @@
 obj-$(CONFIG_SND_ARMAACI)	+= snd-aaci.o
 snd-aaci-objs			:= aaci.o
 
-obj-$(CONFIG_SND_PXA2XX_PCM)	+= snd-pxa2xx-pcm.o
-snd-pxa2xx-pcm-objs		:= pxa2xx-pcm.o
-
 obj-$(CONFIG_SND_PXA2XX_LIB)	+= snd-pxa2xx-lib.o
 snd-pxa2xx-lib-y		:= pxa2xx-pcm-lib.o
 snd-pxa2xx-lib-$(CONFIG_SND_PXA2XX_LIB_AC97)	+= pxa2xx-ac97-lib.o

sound/arm/pxa2xx-ac97-lib.c

@@ -19,6 +19,7 @@
 #include <linux/module.h>
 #include <linux/io.h>
 #include <linux/gpio.h>
+#include <linux/of_gpio.h>
 
 #include <sound/pxa2xx-lib.h>
 
@@ -337,6 +338,17 @@ int pxa2xx_ac97_hw_probe(struct platform_device *dev)
 			dev_err(&dev->dev, "Invalid reset GPIO %d\n",
 				pdata->reset_gpio);
 		}
+	} else if (!pdata && dev->dev.of_node) {
+		pdata = devm_kzalloc(&dev->dev, sizeof(*pdata), GFP_KERNEL);
+		if (!pdata)
+			return -ENOMEM;
+		pdata->reset_gpio = of_get_named_gpio(dev->dev.of_node,
+						      "reset-gpios", 0);
+		if (pdata->reset_gpio == -ENOENT)
+			pdata->reset_gpio = -1;
+		else if (pdata->reset_gpio < 0)
+			return pdata->reset_gpio;
+		reset_gpio = pdata->reset_gpio;
 	} else {
 		if (cpu_is_pxa27x())
 			reset_gpio = 113;

sound/arm/pxa2xx-ac97.c

@@ -15,7 +15,7 @@
 #include <linux/module.h>
 #include <linux/platform_device.h>
 #include <linux/dmaengine.h>
-#include <linux/dma/pxa-dma.h>
+#include <linux/dma-mapping.h>
 
 #include <sound/core.h>
 #include <sound/pcm.h>
@@ -27,8 +27,6 @@
 #include <mach/regs-ac97.h>
 #include <mach/audio.h>
 
-#include "pxa2xx-pcm.h"
-
 static void pxa2xx_ac97_legacy_reset(struct snd_ac97 *ac97)
 {
 	if (!pxa2xx_ac97_try_cold_reset())
@@ -63,61 +61,46 @@ static struct snd_ac97_bus_ops pxa2xx_ac97_ops = {
 	.reset	= pxa2xx_ac97_legacy_reset,
 };
 
-static struct pxad_param pxa2xx_ac97_pcm_out_req = {
-	.prio = PXAD_PRIO_LOWEST,
-	.drcmr = 12,
-};
-
-static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_out = {
-	.addr		= __PREG(PCDR),
-	.addr_width	= DMA_SLAVE_BUSWIDTH_4_BYTES,
-	.maxburst	= 32,
-	.filter_data	= &pxa2xx_ac97_pcm_out_req,
-};
-
-static struct pxad_param pxa2xx_ac97_pcm_in_req = {
-	.prio = PXAD_PRIO_LOWEST,
-	.drcmr = 11,
-};
-
-static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_in = {
-	.addr		= __PREG(PCDR),
-	.addr_width	= DMA_SLAVE_BUSWIDTH_4_BYTES,
-	.maxburst	= 32,
-	.filter_data	= &pxa2xx_ac97_pcm_in_req,
-};
-
 static struct snd_pcm *pxa2xx_ac97_pcm;
 static struct snd_ac97 *pxa2xx_ac97_ac97;
 
-static int pxa2xx_ac97_pcm_startup(struct snd_pcm_substream *substream)
+static int pxa2xx_ac97_pcm_open(struct snd_pcm_substream *substream)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	pxa2xx_audio_ops_t *platform_ops;
-	int r;
+	int ret, i;
+
+	ret = pxa2xx_pcm_open(substream);
+	if (ret)
+		return ret;
 
 	runtime->hw.channels_min = 2;
 	runtime->hw.channels_max = 2;
 
-	r = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
-	    AC97_RATES_FRONT_DAC : AC97_RATES_ADC;
-	runtime->hw.rates = pxa2xx_ac97_ac97->rates[r];
+	i = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
+		AC97_RATES_FRONT_DAC : AC97_RATES_ADC;
+	runtime->hw.rates = pxa2xx_ac97_ac97->rates[i];
 	snd_pcm_limit_hw_rates(runtime);
 
-       	platform_ops = substream->pcm->card->dev->platform_data;
-	if (platform_ops && platform_ops->startup)
-		return platform_ops->startup(substream, platform_ops->priv);
-	else
-		return 0;
+	platform_ops = substream->pcm->card->dev->platform_data;
+	if (platform_ops && platform_ops->startup) {
+		ret = platform_ops->startup(substream, platform_ops->priv);
+		if (ret < 0)
+			pxa2xx_pcm_close(substream);
+	}
+
+	return ret;
 }
 
-static void pxa2xx_ac97_pcm_shutdown(struct snd_pcm_substream *substream)
+static int pxa2xx_ac97_pcm_close(struct snd_pcm_substream *substream)
 {
 	pxa2xx_audio_ops_t *platform_ops;
 
-       	platform_ops = substream->pcm->card->dev->platform_data;
+	platform_ops = substream->pcm->card->dev->platform_data;
 	if (platform_ops && platform_ops->shutdown)
 		platform_ops->shutdown(substream, platform_ops->priv);
+
+	return 0;
 }
 
 static int pxa2xx_ac97_pcm_prepare(struct snd_pcm_substream *substream)
@@ -125,17 +108,15 @@ static int pxa2xx_ac97_pcm_prepare(struct snd_pcm_substream *substream)
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	int reg = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
 		  AC97_PCM_FRONT_DAC_RATE : AC97_PCM_LR_ADC_RATE;
+	int ret;
+
+	ret = pxa2xx_pcm_prepare(substream);
+	if (ret < 0)
+		return ret;
+
 	return snd_ac97_set_rate(pxa2xx_ac97_ac97, reg, runtime->rate);
 }
 
-static struct pxa2xx_pcm_client pxa2xx_ac97_pcm_client = {
-	.playback_params	= &pxa2xx_ac97_pcm_out,
-	.capture_params		= &pxa2xx_ac97_pcm_in,
-	.startup		= pxa2xx_ac97_pcm_startup,
-	.shutdown		= pxa2xx_ac97_pcm_shutdown,
-	.prepare		= pxa2xx_ac97_pcm_prepare,
-};
-
 #ifdef CONFIG_PM_SLEEP
 
 static int pxa2xx_ac97_do_suspend(struct snd_card *card)
@@ -193,6 +174,53 @@ static int pxa2xx_ac97_resume(struct device *dev)
 static SIMPLE_DEV_PM_OPS(pxa2xx_ac97_pm_ops, pxa2xx_ac97_suspend, pxa2xx_ac97_resume);
 #endif
 
+static const struct snd_pcm_ops pxa2xx_ac97_pcm_ops = {
+	.open		= pxa2xx_ac97_pcm_open,
+	.close		= pxa2xx_ac97_pcm_close,
+	.ioctl		= snd_pcm_lib_ioctl,
+	.hw_params	= pxa2xx_pcm_hw_params,
+	.hw_free	= pxa2xx_pcm_hw_free,
+	.prepare	= pxa2xx_ac97_pcm_prepare,
+	.trigger	= pxa2xx_pcm_trigger,
+	.pointer	= pxa2xx_pcm_pointer,
+	.mmap		= pxa2xx_pcm_mmap,
+};
+
+
+static int pxa2xx_ac97_pcm_new(struct snd_card *card)
+{
+	struct snd_pcm *pcm;
+	int stream, ret;
+
+	ret = snd_pcm_new(card, "PXA2xx-PCM", 0, 1, 1, &pcm);
+	if (ret)
+		goto out;
+
+	pcm->private_free = pxa2xx_pcm_free_dma_buffers;
+
+	ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32));
+	if (ret)
+		goto out;
+
+	stream = SNDRV_PCM_STREAM_PLAYBACK;
+	snd_pcm_set_ops(pcm, stream, &pxa2xx_ac97_pcm_ops);
+	ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, stream);
+	if (ret)
+		goto out;
+
+	stream = SNDRV_PCM_STREAM_CAPTURE;
+	snd_pcm_set_ops(pcm, stream, &pxa2xx_ac97_pcm_ops);
+	ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, stream);
+	if (ret)
+		goto out;
+
+	pxa2xx_ac97_pcm = pcm;
+	ret = 0;
+
+ out:
+	return ret;
+}
+
 static int pxa2xx_ac97_probe(struct platform_device *dev)
 {
 	struct snd_card *card;
@@ -214,7 +242,7 @@ static int pxa2xx_ac97_probe(struct platform_device *dev)
 
 	strlcpy(card->driver, dev->dev.driver->name, sizeof(card->driver));
 
-	ret = pxa2xx_pcm_new(card, &pxa2xx_ac97_pcm_client, &pxa2xx_ac97_pcm);
+	ret = pxa2xx_ac97_pcm_new(card);
 	if (ret)
 		goto err;
 

sound/arm/pxa2xx-pcm-lib.c

@@ -16,8 +16,6 @@
 #include <sound/pxa2xx-lib.h>
 #include <sound/dmaengine_pcm.h>
 
-#include "pxa2xx-pcm.h"
-
 static const struct snd_pcm_hardware pxa2xx_pcm_hardware = {
 	.info			= SNDRV_PCM_INFO_MMAP |
 				  SNDRV_PCM_INFO_MMAP_VALID |
@@ -25,8 +23,8 @@ static const struct snd_pcm_hardware pxa2xx_pcm_hardware = {
 				  SNDRV_PCM_INFO_PAUSE |
 				  SNDRV_PCM_INFO_RESUME,
 	.formats		= SNDRV_PCM_FMTBIT_S16_LE |
-					SNDRV_PCM_FMTBIT_S24_LE |
-					SNDRV_PCM_FMTBIT_S32_LE,
+				  SNDRV_PCM_FMTBIT_S24_LE |
+				  SNDRV_PCM_FMTBIT_S32_LE,
 	.period_bytes_min	= 32,
 	.period_bytes_max	= 8192 - 32,
 	.periods_min		= 1,
@@ -35,8 +33,8 @@ static const struct snd_pcm_hardware pxa2xx_pcm_hardware = {
 	.fifo_size		= 32,
 };
 
-int __pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream,
-				struct snd_pcm_hw_params *params)
+int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream,
+			 struct snd_pcm_hw_params *params)
 {
 	struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream);
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
@@ -64,14 +62,14 @@ int __pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream,
 
 	return 0;
 }
-EXPORT_SYMBOL(__pxa2xx_pcm_hw_params);
+EXPORT_SYMBOL(pxa2xx_pcm_hw_params);
 
-int __pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream)
+int pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream)
 {
 	snd_pcm_set_runtime_buffer(substream, NULL);
 	return 0;
 }
-EXPORT_SYMBOL(__pxa2xx_pcm_hw_free);
+EXPORT_SYMBOL(pxa2xx_pcm_hw_free);
 
 int pxa2xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 {
@@ -86,13 +84,13 @@ pxa2xx_pcm_pointer(struct snd_pcm_substream *substream)
 }
 EXPORT_SYMBOL(pxa2xx_pcm_pointer);
 
-int __pxa2xx_pcm_prepare(struct snd_pcm_substream *substream)
+int pxa2xx_pcm_prepare(struct snd_pcm_substream *substream)
 {
 	return 0;
 }
-EXPORT_SYMBOL(__pxa2xx_pcm_prepare);
+EXPORT_SYMBOL(pxa2xx_pcm_prepare);
 
-int __pxa2xx_pcm_open(struct snd_pcm_substream *substream)
+int pxa2xx_pcm_open(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_pcm_runtime *runtime = substream->runtime;
@@ -125,17 +123,17 @@ int __pxa2xx_pcm_open(struct snd_pcm_substream *substream)
 	if (ret < 0)
 		return ret;
 
-	return snd_dmaengine_pcm_open_request_chan(substream,
-					pxad_filter_fn,
-					dma_params->filter_data);
+	return snd_dmaengine_pcm_open(
+		substream, dma_request_slave_channel(rtd->cpu_dai->dev,
+						     dma_params->chan_name));
 }
-EXPORT_SYMBOL(__pxa2xx_pcm_open);
+EXPORT_SYMBOL(pxa2xx_pcm_open);
 
-int __pxa2xx_pcm_close(struct snd_pcm_substream *substream)
+int pxa2xx_pcm_close(struct snd_pcm_substream *substream)
 {
 	return snd_dmaengine_pcm_close_release_chan(substream);
 }
-EXPORT_SYMBOL(__pxa2xx_pcm_close);
+EXPORT_SYMBOL(pxa2xx_pcm_close);
 
 int pxa2xx_pcm_mmap(struct snd_pcm_substream *substream,
 	struct vm_area_struct *vma)
@@ -181,6 +179,47 @@ void pxa2xx_pcm_free_dma_buffers(struct snd_pcm *pcm)
 }
 EXPORT_SYMBOL(pxa2xx_pcm_free_dma_buffers);
 
+int pxa2xx_soc_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_card *card = rtd->card->snd_card;
+	struct snd_pcm *pcm = rtd->pcm;
+	int ret;
+
+	ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32));
+	if (ret)
+		return ret;
+
+	if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
+		ret = pxa2xx_pcm_preallocate_dma_buffer(pcm,
+			SNDRV_PCM_STREAM_PLAYBACK);
+		if (ret)
+			goto out;
+	}
+
+	if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
+		ret = pxa2xx_pcm_preallocate_dma_buffer(pcm,
+			SNDRV_PCM_STREAM_CAPTURE);
+		if (ret)
+			goto out;
+	}
+ out:
+	return ret;
+}
+EXPORT_SYMBOL(pxa2xx_soc_pcm_new);
+
+const struct snd_pcm_ops pxa2xx_pcm_ops = {
+	.open		= pxa2xx_pcm_open,
+	.close		= pxa2xx_pcm_close,
+	.ioctl		= snd_pcm_lib_ioctl,
+	.hw_params	= pxa2xx_pcm_hw_params,
+	.hw_free	= pxa2xx_pcm_hw_free,
+	.prepare	= pxa2xx_pcm_prepare,
+	.trigger	= pxa2xx_pcm_trigger,
+	.pointer	= pxa2xx_pcm_pointer,
+	.mmap		= pxa2xx_pcm_mmap,
+};
+EXPORT_SYMBOL(pxa2xx_pcm_ops);
+
 MODULE_AUTHOR("Nicolas Pitre");
 MODULE_DESCRIPTION("Intel PXA2xx sound library");
 MODULE_LICENSE("GPL");

sound/arm/pxa2xx-pcm.c

@@ -1,129 +0,0 @@
-/*
- * linux/sound/arm/pxa2xx-pcm.c -- ALSA PCM interface for the Intel PXA2xx chip
- *
- * Author:	Nicolas Pitre
- * Created:	Nov 30, 2004
- * Copyright:	(C) 2004 MontaVista Software, Inc.
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- */
-
-#include <linux/module.h>
-#include <linux/dma-mapping.h>
-#include <linux/dmaengine.h>
-
-#include <mach/dma.h>
-
-#include <sound/core.h>
-#include <sound/pxa2xx-lib.h>
-#include <sound/dmaengine_pcm.h>
-
-#include "pxa2xx-pcm.h"
-
-static int pxa2xx_pcm_prepare(struct snd_pcm_substream *substream)
-{
-	struct pxa2xx_pcm_client *client = substream->private_data;
-
-	__pxa2xx_pcm_prepare(substream);
-
-	return client->prepare(substream);
-}
-
-static int pxa2xx_pcm_open(struct snd_pcm_substream *substream)
-{
-	struct pxa2xx_pcm_client *client = substream->private_data;
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct pxa2xx_runtime_data *rtd;
-	int ret;
-
-	ret = __pxa2xx_pcm_open(substream);
-	if (ret)
-		goto out;
-
-	rtd = runtime->private_data;
-
-	rtd->params = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
-		      client->playback_params : client->capture_params;
-
-	ret = client->startup(substream);
-	if (!ret)
-		goto err2;
-
-	return 0;
-
- err2:
-	__pxa2xx_pcm_close(substream);
- out:
-	return ret;
-}
-
-static int pxa2xx_pcm_close(struct snd_pcm_substream *substream)
-{
-	struct pxa2xx_pcm_client *client = substream->private_data;
-
-	client->shutdown(substream);
-
-	return __pxa2xx_pcm_close(substream);
-}
-
-static const struct snd_pcm_ops pxa2xx_pcm_ops = {
-	.open		= pxa2xx_pcm_open,
-	.close		= pxa2xx_pcm_close,
-	.ioctl		= snd_pcm_lib_ioctl,
-	.hw_params	= __pxa2xx_pcm_hw_params,
-	.hw_free	= __pxa2xx_pcm_hw_free,
-	.prepare	= pxa2xx_pcm_prepare,
-	.trigger	= pxa2xx_pcm_trigger,
-	.pointer	= pxa2xx_pcm_pointer,
-	.mmap		= pxa2xx_pcm_mmap,
-};
-
-int pxa2xx_pcm_new(struct snd_card *card, struct pxa2xx_pcm_client *client,
-		   struct snd_pcm **rpcm)
-{
-	struct snd_pcm *pcm;
-	int play = client->playback_params ? 1 : 0;
-	int capt = client->capture_params ? 1 : 0;
-	int ret;
-
-	ret = snd_pcm_new(card, "PXA2xx-PCM", 0, play, capt, &pcm);
-	if (ret)
-		goto out;
-
-	pcm->private_data = client;
-	pcm->private_free = pxa2xx_pcm_free_dma_buffers;
-
-	ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32));
-	if (ret)
-		goto out;
-
-	if (play) {
-		int stream = SNDRV_PCM_STREAM_PLAYBACK;
-		snd_pcm_set_ops(pcm, stream, &pxa2xx_pcm_ops);
-		ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, stream);
-		if (ret)
-			goto out;
-	}
-	if (capt) {
-		int stream = SNDRV_PCM_STREAM_CAPTURE;
-		snd_pcm_set_ops(pcm, stream, &pxa2xx_pcm_ops);
-		ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, stream);
-		if (ret)
-			goto out;
-	}
-
-	if (rpcm)
-		*rpcm = pcm;
-	ret = 0;
-
- out:
-	return ret;
-}
-
-EXPORT_SYMBOL(pxa2xx_pcm_new);
-
-MODULE_AUTHOR("Nicolas Pitre");
-MODULE_DESCRIPTION("Intel PXA2xx PCM DMA module");
-MODULE_LICENSE("GPL");

sound/arm/pxa2xx-pcm.h

@@ -1,27 +0,0 @@
-/*
- * linux/sound/arm/pxa2xx-pcm.h -- ALSA PCM interface for the Intel PXA2xx chip
- *
- * Author:	Nicolas Pitre
- * Created:	Nov 30, 2004
- * Copyright:	MontaVista Software, Inc.
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- */
-
-struct pxa2xx_runtime_data {
-	int dma_ch;
-	struct snd_dmaengine_dai_dma_data *params;
-};
-
-struct pxa2xx_pcm_client {
-	struct snd_dmaengine_dai_dma_data *playback_params;
-	struct snd_dmaengine_dai_dma_data *capture_params;
-	int (*startup)(struct snd_pcm_substream *);
-	void (*shutdown)(struct snd_pcm_substream *);
-	int (*prepare)(struct snd_pcm_substream *);
-};
-
-extern int pxa2xx_pcm_new(struct snd_card *, struct pxa2xx_pcm_client *, struct snd_pcm **);
-

sound/core/compress_offload.c

@@ -1160,18 +1160,6 @@ int snd_compress_deregister(struct snd_compr *device)
 }
 EXPORT_SYMBOL_GPL(snd_compress_deregister);
 
-static int __init snd_compress_init(void)
-{
-	return 0;
-}
-
-static void __exit snd_compress_exit(void)
-{
-}
-
-module_init(snd_compress_init);
-module_exit(snd_compress_exit);
-
 MODULE_DESCRIPTION("ALSA Compressed offload framework");
 MODULE_AUTHOR("Vinod Koul <vinod.koul@linux.intel.com>");
 MODULE_LICENSE("GPL v2");

sound/core/memalloc.c

@@ -242,16 +242,12 @@ int snd_dma_alloc_pages_fallback(int type, struct device *device, size_t size,
 	int err;
 
 	while ((err = snd_dma_alloc_pages(type, device, size, dmab)) < 0) {
-		size_t aligned_size;
 		if (err != -ENOMEM)
 			return err;
 		if (size <= PAGE_SIZE)
 			return -ENOMEM;
-		aligned_size = PAGE_SIZE << get_order(size);
-		if (size != aligned_size)
-			size = aligned_size;
-		else
-			size >>= 1;
+		size >>= 1;
+		size = PAGE_SIZE << get_order(size);
 	}
 	if (! dmab->area)
 		return -ENOMEM;

sound/core/oss/pcm_oss.c

@@ -1851,7 +1851,7 @@ static int snd_pcm_oss_get_formats(struct snd_pcm_oss_file *pcm_oss_file)
 	format_mask = hw_param_mask_c(params, SNDRV_PCM_HW_PARAM_FORMAT);
 	for (fmt = 0; fmt < 32; ++fmt) {
 		if (snd_mask_test(format_mask, fmt)) {
-			int f = snd_pcm_oss_format_to(fmt);
+			int f = snd_pcm_oss_format_to((__force snd_pcm_format_t)fmt);
 			if (f >= 0)
 				formats |= f;
 		}