Merge tag 'sound-4.18-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "We've got many code additions at this cycle as a result of quite a few
  new drivers. Below are highlights:

  Core stuff:
   - Fix the long-standing issue with the device registration order; the
     control device is now registered at last
   - PCM locking code cleanups for RT kernels
   - Fixes for possible races in ALSA timer resolution accesses
   - TLV offset definitions in uapi

  ASoC:
   - Many fixes for the topology stuff, including fixes for v4 ABI
     compatibility
   - Lots of cleanups / quirks for Intel platforms based on Realtek
     CODECs
   - Continued componentization works, removing legacy CODEC stuff
   - Conversion of OMAP DMA to the new, more standard SDMA-PCM driver
   - Fixes and updates to Cirrus Logic SoC drivers
   - New Qualcomm DSP support
   - New drivers for Analog SSM2305, Atmel I2S controllers, Mediatek
     MT6351, MT6797 and MT7622, Qualcomm DSPs, Realtek RT1305, RT1306
     and RT5668 and TI TSCS454

  HD-audio:
   - Finally better support for some CA0132 boards, allowing Windows
     firmware
   - HP Spectre x360 support along with a bulk of COEF stuff
   - Blacklisting power save default some known boards reported on
     Fedora

  USB-audio:
   - Continued improvements on UAC3 support; now BADD is supported
   - Fixes / improvements for Dell WD15 dock
   - Allow DMA coherent pages for PCM buffers for ARCH, MIPS & co

  Others:
   - New Xen sound frontend driver support
   - Cache implementation and other improvements for FireWire DICE
   - Conversions to octal permissions in allover places"

* tag 'sound-4.18-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (386 commits)
  ASoC: dapm: delete dapm_kcontrol_data paths list before freeing it
  ALSA: usb-audio: remove redundant check on err
  ASoC: topology: Move skl-tplg-interface.h to uapi
  ASoC: topology: Move v4 manifest header data structures to uapi
  ASoC: topology: Improve backwards compatibility with v4 topology files
  ALSA: pci/hda: Remove unused, broken, header file
  ASoC: TSCS454: Add Support
  ASoC: Intel: kbl: Move codec sysclk config to codec_init function
  ASoC: simple-card: set cpu dai clk in hw_params
  ALSA: hda - Handle kzalloc() failure in snd_hda_attach_pcm_stream()
  ALSA: oxygen: use match_string() helper
  ASoC: dapm: use match_string() helper
  ASoC: max98095: use match_string() helper
  ASoC: max98088: use match_string() helper
  ASoC: Intel: bytcr_rt5651: Set card long_name based on quirks
  ASoC: mt6797-mt6351: add hostless phone call path
  ASoC: mt6797: add Hostless DAI
  ASoC: mt6797: add PCM interface
  ASoC: mediatek: export mtk-afe symbols as needed
  ASoC: codecs: PCM1789: include gpio/consumer.h
  ...

.mailmap

@@ -186,6 +186,9 @@ Uwe Kleine-König <ukleinek@informatik.uni-freiburg.de>
 Uwe Kleine-König <ukl@pengutronix.de>
 Uwe Kleine-König <Uwe.Kleine-Koenig@digi.com>
 Valdis Kletnieks <Valdis.Kletnieks@vt.edu>
+Vinod Koul <vkoul@kernel.org> <vinod.koul@intel.com>
+Vinod Koul <vkoul@kernel.org> <vinod.koul@linux.intel.com>
+Vinod Koul <vkoul@kernel.org> <vkoul@infradead.org>
 Viresh Kumar <vireshk@kernel.org> <viresh.kumar@st.com>
 Viresh Kumar <vireshk@kernel.org> <viresh.linux@gmail.com>
 Viresh Kumar <vireshk@kernel.org> <viresh.kumar2@arm.com>

Documentation/devicetree/bindings/soc/qcom/qcom,apr.txt

+Qualcomm APR (Asynchronous Packet Router) binding
+
+This binding describes the Qualcomm APR. APR is a IPC protocol for
+communication between Application processor and QDSP. APR is mainly
+used for audio/voice services on the QDSP.
+
+- compatible:
+	Usage: required
+	Value type: <stringlist>
+	Definition: must be "qcom,apr-v<VERSION-NUMBER>", example "qcom,apr-v2"
+
+- reg
+	Usage: required
+	Value type: <u32>
+	Definition: Destination processor ID.
+	Possible values are :
+			1 - APR simulator
+			2 - PC
+			3 - MODEM
+			4 - ADSP
+			5 - APPS
+			6 - MODEM2
+			7 - APPS2
+
+= APR SERVICES
+Each subnode of the APR node represents service tied to this apr. The name
+of the nodes are not important. The properties of these nodes are defined
+by the individual bindings for the specific service
+- All APR services MUST contain the following property:
+
+- reg
+	Usage: required
+	Value type: <u32>
+	Definition: APR Service ID
+	Possible values are :
+			3 - DSP Core Service
+			4 - Audio Front End Service.
+			5 - Voice Stream Manager Service.
+			6 - Voice processing manager.
+			7 - Audio Stream Manager Service.
+			8 - Audio Device Manager Service.
+			9 - Multimode voice manager.
+			10 - Core voice stream.
+			11 - Core voice processor.
+			12 - Ultrasound stream manager.
+			13 - Listen stream manager.
+
+= EXAMPLE
+The following example represents a QDSP based sound card on a MSM8996 device
+which uses apr as communication between Apps and QDSP.
+
+	apr@4 {
+		compatible = "qcom,apr-v2";
+		reg = <APR_DOMAIN_ADSP>;
+
+		q6core@3 {
+			compatible = "qcom,q6core";
+			reg = <APR_SVC_ADSP_CORE>;
+		};
+
+		q6afe@4 {
+			compatible = "qcom,q6afe";
+			reg = <APR_SVC_AFE>;
+
+			dais {
+				#sound-dai-cells = <1>;
+				hdmi@1 {
+					reg = <1>;
+				};
+			};
+		};
+
+		q6asm@7 {
+			compatible = "qcom,q6asm";
+			reg = <APR_SVC_ASM>;
+			...
+		};
+
+		q6adm@8 {
+			compatible = "qcom,q6adm";
+			reg = <APR_SVC_ADM>;
+			...
+		};
+	};

Documentation/devicetree/bindings/sound/adi,ssm2305.txt

+Analog Devices SSM2305 Speaker Amplifier
+========================================
+
+Required properties:
+  - compatible : "adi,ssm2305"
+  - shutdown-gpios : The gpio connected to the shutdown pin.
+                     The gpio signal is ACTIVE_LOW.
+
+Example:
+
+ssm2305: analog-amplifier {
+	compatible = "adi,ssm2305";
+	shutdown-gpios = <&gpio3 20 GPIO_ACTIVE_LOW>;
+};

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

+* Atmel I2S controller
+
+Required properties:
+- compatible:     Should be "atmel,sama5d2-i2s".
+- reg:            Should be the physical base address of the controller and the
+                  length of memory mapped region.
+- interrupts:     Should contain the interrupt for the controller.
+- dmas:           Should be one per channel name listed in the dma-names property,
+                  as described in atmel-dma.txt and dma.txt files.
+- dma-names:      Two dmas have to be defined, "tx" and "rx".
+                  This IP also supports one shared channel for both rx and tx;
+                  if this mode is used, one "rx-tx" name must be used.
+- clocks:         Must contain an entry for each entry in clock-names.
+                  Please refer to clock-bindings.txt.
+- 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:
+- pinctrl-0:      Should specify pin control groups used for this controller.
+- 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.
+
+Example:
+
+	i2s@f8050000 {
+		compatible = "atmel,sama5d2-i2s";
+		reg = <0xf8050000 0x300>;
+		interrupts = <54 IRQ_TYPE_LEVEL_HIGH 7>;
+		dmas = <&dma0
+			(AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1) |
+			 AT91_XDMAC_DT_PERID(31))>,
+		       <&dma0
+			(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";
+		pinctrl-names = "default";
+		pinctrl-0 = <&pinctrl_i2s0_default>;
+	};

Documentation/devicetree/bindings/sound/cs42xx8.txt

@@ -16,7 +16,7 @@ Required properties:
 
 Example:
 
-codec: cs42888@48 {
+cs42888: codec@48 {
 	compatible = "cirrus,cs42888";
 	reg = <0x48>;
 	clocks = <&codec_mclk 0>;

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

@@ -31,14 +31,16 @@ Required properties:
 			  it. This property is optional depending on the SoC
 			  design.
 
-   - big-endian		: If this property is absent, the little endian mode
-			  will be in use as default. Otherwise, the big endian
-			  mode will be in use for all the device registers.
-
    - fsl,asrc-rate	: Defines a mutual sample rate used by DPCM Back Ends.
 
    - fsl,asrc-width	: Defines a mutual sample width used by DPCM Back Ends.
 
+Optional properties:
+
+   - big-endian		: If this property is absent, the little endian mode
+			  will be in use as default. Otherwise, the big endian
+			  mode will be in use for all the device registers.
+
 Example:
 
 asrc: asrc@2034000 {

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

@@ -42,6 +42,8 @@ Required properties:
 			  means all the settings for Receiving would be
 			  duplicated from Transmition related registers.
 
+Optional properties:
+
   - big-endian		: If this property is absent, the native endian mode
 			  will be in use as default, or the big endian mode
 			  will be in use for all the device registers.

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

@@ -33,6 +33,8 @@ Required properties:
 			  it. This property is optional depending on the SoC
 			  design.
 
+Optional properties:
+
    - big-endian		: If this property is absent, the native endian mode
 			  will be in use as default, or the big endian mode
 			  will be in use for all the device registers.

Documentation/devicetree/bindings/sound/fsl-sai.txt

@@ -28,9 +28,6 @@ Required properties:
 			  pinctrl-names. See ../pinctrl/pinctrl-bindings.txt
 			  for details of the property values.
 
-  - big-endian		: Boolean property, required if all the FTM_PWM
-			  registers are big-endian rather than little-endian.
-
   - lsb-first		: Configures whether the LSB or the MSB is transmitted
 			  first for the fifo data. If this property is absent,
 			  the MSB is transmitted first as default, or the LSB
@@ -48,6 +45,11 @@ Required properties:
 			  receive data by following their own bit clocks and
 			  frame sync clocks separately.
 
+Optional properties:
+
+  - big-endian		: Boolean property, required if all the SAI
+			  registers are big-endian rather than little-endian.
+
 Optional properties (for mx6ul):
 
   - fsl,sai-mclk-direction-output: This is a boolean property. If present,

Documentation/devicetree/bindings/sound/mt2701-afe-pcm.txt

 Mediatek AFE PCM controller for mt2701
 
 Required properties:
-- compatible = "mediatek,mt2701-audio";
+- compatible: should be one of the followings.
+	      - "mediatek,mt2701-audio"
+	      - "mediatek,mt7622-audio"
 - interrupts: should contain AFE and ASYS interrupts
 - interrupt-names: should be "afe" and "asys"
 - power-domains: should define the power domain

Documentation/devicetree/bindings/sound/mt6351.txt

+Mediatek MT6351 Audio Codec
+
+The communication between MT6351 and SoC is through Mediatek PMIC wrapper.
+For more detail, please visit Mediatek PMIC wrapper documentation.
+
+Must be a child node of PMIC wrapper.
+
+Required properties:
+
+- compatible : "mediatek,mt6351-sound".
+
+Example:
+
+mt6351_snd {
+	compatible = "mediatek,mt6351-sound";
+};

Documentation/devicetree/bindings/sound/mt6797-afe-pcm.txt

+Mediatek AFE PCM controller for mt6797
+
+Required properties:
+- compatible = "mediatek,mt6797-audio";
+- reg: register location and size
+- interrupts: should contain AFE interrupt
+- power-domains: should define the power domain
+- clocks: Must contain an entry for each entry in clock-names
+- clock-names: should have these clock names:
+		"infra_sys_audio_clk",
+		"infra_sys_audio_26m",
+		"mtkaif_26m_clk",
+		"top_mux_audio",
+		"top_mux_aud_intbus",
+		"top_sys_pll3_d4",
+		"top_sys_pll1_d4",
+		"top_clk26m_clk";
+
+Example:
+
+	afe: mt6797-afe-pcm@11220000  {
+		compatible = "mediatek,mt6797-audio";
+		reg = <0 0x11220000 0 0x1000>;
+		interrupts = <GIC_SPI 151 IRQ_TYPE_LEVEL_LOW>;
+		power-domains = <&scpsys MT6797_POWER_DOMAIN_AUDIO>;
+		clocks = <&infrasys CLK_INFRA_AUDIO>,
+			 <&infrasys CLK_INFRA_AUDIO_26M>,
+			 <&infrasys CLK_INFRA_AUDIO_26M_PAD_TOP>,
+			 <&topckgen CLK_TOP_MUX_AUDIO>,
+			 <&topckgen CLK_TOP_MUX_AUD_INTBUS>,
+			 <&topckgen CLK_TOP_SYSPLL3_D4>,
+			 <&topckgen CLK_TOP_SYSPLL1_D4>,
+			 <&clk26m>;
+		clock-names = "infra_sys_audio_clk",
+			      "infra_sys_audio_26m",
+			      "mtkaif_26m_clk",
+			      "top_mux_audio",
+			      "top_mux_aud_intbus",
+			      "top_sys_pll3_d4",
+			      "top_sys_pll1_d4",
+			      "top_clk26m_clk";
+	};

Documentation/devicetree/bindings/sound/mt6797-mt6351.txt

+MT6797 with MT6351 CODEC
+
+Required properties:
+- compatible: "mediatek,mt6797-mt6351-sound"
+- mediatek,platform: the phandle of MT6797 ASoC platform
+- mediatek,audio-codec: the phandles of MT6351 codec
+
+Example:
+
+	sound {
+		compatible = "mediatek,mt6797-mt6351-sound";
+		mediatek,audio-codec = <&mt6351_snd>;
+		mediatek,platform = <&afe>;
+	};

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

+* Qualcomm Technologies APQ8096 ASoC sound card driver
+
+This binding describes the APQ8096 sound card, which uses qdsp for audio.
+
+- compatible:
+	Usage: required
+	Value type: <stringlist>
+	Definition: must be "qcom,apq8096-sndcard"
+
+- qcom,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:
+		  Valid names include:
+
+		Board Connectors:
+			"Headphone Left"
+			"Headphone Right"
+			"Earphone"
+			"Line Out1"
+			"Line Out2"
+			"Line Out3"
+			"Line Out4"
+			"Analog Mic1"
+			"Analog Mic2"
+			"Analog Mic3"
+			"Analog Mic4"
+			"Analog Mic5"
+			"Analog Mic6"
+			"Digital Mic2"
+			"Digital Mic3"
+
+		Audio pins and MicBias on WCD9335 Codec:
+			"MIC_BIAS1
+			"MIC_BIAS2"
+			"MIC_BIAS3"
+			"MIC_BIAS4"
+			"AMIC1"
+			"AMIC2"
+			"AMIC3"
+			"AMIC4"
+			"AMIC5"
+			"AMIC6"
+			"AMIC6"
+			"DMIC1"
+			"DMIC2"
+			"DMIC3"
+= 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: Optional
+	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 with arguments>
+	Definition: dai phandle/s and port of CPU/CODEC/PLATFORM node.
+
+Example:
+
+audio {
+	compatible = "qcom,apq8096-sndcard";
+	qcom,model = "DB820c";
+
+	mm1-dai-link {
+		link-name = "MultiMedia1";
+		cpu {
+			sound-dai = <&q6asmdai MSM_FRONTEND_DAI_MULTIMEDIA1>;
+		};
+	};
+
+	hdmi-dai-link {
+		link-name = "HDMI Playback";
+		cpu {
+			sound-dai = <&q6afe HDMI_RX>;
+		};
+
+		platform {
+			sound-dai = <&q6adm>;
+		};
+
+		codec {
+			sound-dai = <&hdmi 0>;
+		};
+	};
+};

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

+Qualcomm Audio Device Manager (Q6ADM) binding
+
+Q6ADM is one of the APR audio service on Q6DSP.
+Please refer to qcom,apr.txt for details of the coommon apr service bindings
+used by the apr service device.
+
+- but must contain the following property:
+
+- compatible:
+	Usage: required
+	Value type: <stringlist>
+	Definition: must be "qcom,q6adm-v<MAJOR-NUMBER>.<MINOR-NUMBER>".
+		   Or "qcom,q6adm" where the version number can be queried
+		   from DSP.
+		   example "qcom,q6adm-v2.0"
+
+
+= ADM routing
+"routing" subnode of the ADM node represents adm routing specific configuration
+
+- #sound-dai-cells
+	Usage: required
+	Value type: <u32>
+	Definition: Must be 0
+
+= EXAMPLE
+q6adm@8 {
+	compatible = "qcom,q6adm";
+	reg = <APR_SVC_ADM>;
+	q6routing: routing {
+		#sound-dai-cells = <0>;
+	};
+};

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

+Qualcomm Audio Front End (Q6AFE) binding
+
+AFE is one of the APR audio service on Q6DSP
+Please refer to qcom,apr.txt for details of the common apr service bindings
+used by all apr services. Must contain the following properties.
+
+- compatible:
+	Usage: required
+	Value type: <stringlist>
+	Definition: must be "qcom,q6afe-v<MAJOR-NUMBER>.<MINOR-NUMBER>"
+		  Or "qcom,q6afe" where the version number can be queried
+		  from DSP.
+		  example "qcom,q6afe"
+
+= AFE DAIs (Digial Audio Interface)
+"dais" subnode of the AFE node. It represents afe dais, each afe dai is a
+subnode of "dais" representing board specific dai setup.
+"dais" node should have following properties followed by dai children.
+
+- #sound-dai-cells
+	Usage: required
+	Value type: <u32>
+	Definition: Must be 1
+
+- #address-cells
+	Usage: required
+	Value type: <u32>
+	Definition: Must be 1
+
+- #size-cells
+	Usage: required
+	Value type: <u32>
+	Definition: Must be 0
+
+== AFE DAI is subnode of "dais" and represent a dai, it includes board specific
+configuration of each dai. Must contain the following properties.
+
+- reg
+	Usage: required
+	Value type: <u32>
+	Definition: Must be dai id
+
+- qcom,sd-lines
+	Usage: required for mi2s interface
+	Value type: <prop-encoded-array>
+	Definition: Must be list of serial data lines used by this dai.
+	should be one or more of the 1-4 sd lines.
+
+ - qcom,tdm-sync-mode:
+	Usage: required for tdm interface
+	Value type: <prop-encoded-array>
+	Definition: Synchronization mode.
+		0 - Short sync bit mode
+		1 - Long sync mode
+		2 - Short sync slot mode
+
+ - qcom,tdm-sync-src:
+	Usage: required for tdm interface
+	Value type: <prop-encoded-array>
+	Definition: Synchronization source.
+		0 - External source
+		1 - Internal source
+
+ - qcom,tdm-data-out:
+	Usage: required for tdm interface
+	Value type: <prop-encoded-array>
+	Definition: Data out signal to drive with other masters.
+		0 - Disable
+		1 - Enable
+
+ - qcom,tdm-invert-sync:
+	Usage: required for tdm interface
+	Value type: <prop-encoded-array>
+	Definition: Invert the sync.
+		0 - Normal
+		1 - Invert
+
+ - qcom,tdm-data-delay:
+	Usage: required for tdm interface
+	Value type: <prop-encoded-array>
+	Definition: Number of bit clock to delay data
+		with respect to sync edge.
+		0 - 0 bit clock cycle
+		1 - 1 bit clock cycle
+		2 - 2 bit clock cycle
+
+ - qcom,tdm-data-align:
+	Usage: required for tdm interface
+	Value type: <prop-encoded-array>
+	Definition: Indicate how data is packed
+		within the slot. For example, 32 slot width in case of
+		sample bit width is 24.
+		0 - MSB
+		1 - LSB
+
+= EXAMPLE
+
+q6afe@4 {
+	compatible = "qcom,q6afe";
+	reg = <APR_SVC_AFE>;
+
+	dais {
+		#sound-dai-cells = <1>;
+		#address-cells = <1>;
+		#size-cells = <0>;
+
+		hdmi@1 {
+			reg = <1>;
+		};
+
+		tdm@24 {
+			reg = <24>;
+			qcom,tdm-sync-mode = <1>:
+			qcom,tdm-sync-src = <1>;
+			qcom,tdm-data-out = <0>;
+			qcom,tdm-invert-sync = <1>;
+			qcom,tdm-data-delay = <1>;
+			qcom,tdm-data-align = <0>;
+
+		};
+
+		tdm@25 {
+			reg = <25>;
+			qcom,tdm-sync-mode = <1>:
+			qcom,tdm-sync-src = <1>;
+			qcom,tdm-data-out = <0>;
+			qcom,tdm-invert-sync = <1>;
+			qcom,tdm-data-delay <1>:
+			qcom,tdm-data-align = <0>;
+		};
+
+		prim-mi2s-rx@16 {
+			reg = <16>;
+			qcom,sd-lines = <1 3>;
+		};
+
+		prim-mi2s-tx@17 {
+			reg = <17>;
+			qcom,sd-lines = <2>;
+		};
+
+		sec-mi2s-rx@18 {
+			reg = <18>;
+			qcom,sd-lines = <1 4>;
+		};
+
+		sec-mi2s-tx@19 {
+			reg = <19>;
+			qcom,sd-lines = <2>;
+		};
+
+		tert-mi2s-rx@20 {
+			reg = <20>;
+			qcom,sd-lines = <2 4>;
+		};
+
+		tert-mi2s-tx@21 {
+			reg = <21>;
+			qcom,sd-lines = <1>;
+		};
+
+		quat-mi2s-rx@22 {
+			reg = <22>;
+			qcom,sd-lines = <1>;
+		};
+
+		quat-mi2s-tx@23 {
+			reg = <23>;
+			qcom,sd-lines = <2>;
+		};
+	};
+};

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

+Qualcomm Audio Stream Manager (Q6ASM) binding
+
+Q6ASM is one of the APR audio service on Q6DSP.
+Please refer to qcom,apr.txt for details of the common apr service bindings
+used by the apr service device.
+
+- but must contain the following property:
+
+- compatible:
+	Usage: required
+	Value type: <stringlist>
+	Definition: must be "qcom,q6asm-v<MAJOR-NUMBER>.<MINOR-NUMBER>".
+		    Or "qcom,q6asm" where the version number can be queried
+		    from DSP.
+		    example "qcom,q6asm-v2.0"
+
+= ASM DAIs (Digial Audio Interface)
+"dais" subnode of the ASM node represents dai specific configuration
+
+- #sound-dai-cells
+	Usage: required
+	Value type: <u32>
+	Definition: Must be 1
+
+= EXAMPLE
+
+q6asm@7 {
+	compatible = "qcom,q6asm";
+	reg = <APR_SVC_ASM>;
+	q6asmdai: dais {
+		#sound-dai-cells = <1>;
+	};
+};

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

+Qualcomm ADSP Core service binding
+
+Q6CORE is one of the APR audio service on Q6DSP.
+Please refer to qcom,apr.txt for details of the common apr service bindings
+used by the apr service device.
+
+- but must contain the following property:
+
+- compatible:
+	Usage: required
+	Value type: <stringlist>
+	Definition: must be "qcom,q6core-v<MAJOR-NUMBER>.<MINOR-NUMBER>".
+		   Or "qcom,q6core" where the version number can be queried
+		   from DSP.
+		   example "qcom,q6core-v2.0"
+
+= EXAMPLE
+q6core@3 {
+	compatible = "qcom,q6core";
+	reg = <APR_SVC_ADSP_CORE>;
+};

Documentation/devicetree/bindings/sound/rt274.txt

@@ -26,7 +26,7 @@ Pins on the device (for linking into audio routes) for RT274:
 
 Example:
 
-codec: rt274@1c {
+rt274: codec@1c {
 	compatible = "realtek,rt274";
 	reg = <0x1c>;
 	interrupts = <7 IRQ_TYPE_EDGE_FALLING>;

Documentation/devicetree/bindings/sound/rt5514.txt

@@ -32,7 +32,7 @@ Pins on the device (for linking into audio routes) for I2C:
 
 Example:
 
-codec: rt5514@57 {
+rt5514: codec@57 {
 	compatible = "realtek,rt5514";
 	reg = <0x57>;
 };

Documentation/devicetree/bindings/sound/rt5616.txt

@@ -26,7 +26,7 @@ Pins on the device (for linking into audio routes) for RT5616:
 
 Example:
 
-codec: rt5616@1b {
+rt5616: codec@1b {
 	compatible = "realtek,rt5616";
 	reg = <0x1b>;
 };

Documentation/devicetree/bindings/sound/rt5640.txt

@@ -22,6 +22,41 @@ Optional properties:
 
 - realtek,ldo1-en-gpios : The GPIO that controls the CODEC's LDO1_EN pin.
 
+- realtek,dmic1-data-pin
+  0: dmic1 is not used
+  1: using IN1P pin as dmic1 data pin
+  2: using GPIO3 pin as dmic1 data pin
+
+- realtek,dmic2-data-pin
+  0: dmic2 is not used
+  1: using IN1N pin as dmic2 data pin
+  2: using GPIO4 pin as dmic2 data pin
+
+- realtek,jack-detect-source
+  u32. Valid values:
+  0: jack-detect is not used
+  1: Use GPIO1 for jack-detect
+  2: Use JD1_IN4P for jack-detect
+  3: Use JD2_IN4N for jack-detect
+  4: Use GPIO2 for jack-detect
+  5: Use GPIO3 for jack-detect
+  6: Use GPIO4 for jack-detect
+
+- realtek,jack-detect-not-inverted
+  bool. Normal jack-detect switches give an inverted signal, set this bool
+  in the rare case you've a jack-detect switch which is not inverted.
+
+- realtek,over-current-threshold-microamp
+  u32, micbias over-current detection threshold in µA, valid values are
+  600, 1500 and 2000µA.
+
+- realtek,over-current-scale-factor
+  u32, micbias over-current detection scale-factor, valid values are:
+  0: Scale current by 0.5
+  1: Scale current by 0.75
+  2: Scale current by 1.0
+  3: Scale current by 1.5
+
 Pins on the device (for linking into audio routes) for RT5639/RT5640:
 
   * DMIC1

Documentation/devicetree/bindings/sound/rt5645.txt

@@ -69,4 +69,4 @@ codec: rt5650@1a {
 	realtek,dmic-en = "true";
 	realtek,en-jd-func = "true";
 	realtek,jd-mode = <3>;
-};
\ No newline at end of file
+};

Documentation/devicetree/bindings/sound/rt5651.txt

@@ -50,7 +50,7 @@ Pins on the device (for linking into audio routes) for RT5651:
 
 Example:
 
-codec: rt5651@1a {
+rt5651: codec@1a {
 	compatible = "realtek,rt5651";
 	reg = <0x1a>;
 	realtek,dmic-en = "true";

Documentation/devicetree/bindings/sound/rt5663.txt

@@ -47,7 +47,7 @@ Pins on the device (for linking into audio routes) for RT5663:
 
 Example:
 
-codec: rt5663@12 {
+rt5663: codec@12 {
 	compatible = "realtek,rt5663";
 	reg = <0x12>;
 	interrupts = <7 IRQ_TYPE_EDGE_FALLING>;

Documentation/devicetree/bindings/sound/rt5668.txt

+RT5668B audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+- compatible : "realtek,rt5668b"
+
+- 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 RT5668B:
+
+  * DMIC L1
+  * DMIC R1
+  * IN1P
+  * HPOL
+  * HPOR
+
+Example:
+
+rt5668 {
+	compatible = "realtek,rt5668b";
+	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

@@ -39,7 +39,7 @@ VDDIO		1.8V		2.5V		3.3V
 
 Example:
 
-codec: sgtl5000@a {
+sgtl5000: codec@a {
 	compatible = "fsl,sgtl5000";
 	reg = <0x0a>;
 	#sound-dai-cells = <0>;

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

@@ -86,6 +86,11 @@ Optional CPU/CODEC subnodes properties:
 					  in dai startup() and disabled with
 					  clk_disable_unprepare() in dai
 					  shutdown().
+					  If a clock is specified and a
+					  multiplication factor is given with
+					  mclk-fs, the clock will be set to the
+					  calculated mclk frequency when the
+					  stream starts.
 - system-clock-direction-out		: specifies clock direction as 'out' on
 					  initialization. It is useful for some aCPUs with
 					  fixed clocks.

Documentation/devicetree/bindings/sound/ti,tas6424.txt

@@ -6,6 +6,8 @@ Required properties:
 	- compatible: "ti,tas6424" - TAS6424
 	- reg: I2C slave address
 	- sound-dai-cells: must be equal to 0
+	- standby-gpios: GPIO used to shut the TAS6424 down.
+	- mute-gpios: GPIO used to mute all the outputs
 
 Example:
 

Documentation/devicetree/bindings/sound/tscs42xx.txt

@@ -8,9 +8,15 @@ Required Properties:
 	- reg : 	<0x71> for analog mic
 			<0x69> for digital mic
 
+	- clock-names:	Must one of  the following "mclk1", "xtal", "mclk2"
+
+	- clocks:	phandle of the clock that provides the codec sysclk
+
 Example:
 
 wookie: codec@69 {
 	compatible = "tempo,tscs42A2";
 	reg = <0x69>;
+	clock-names = "xtal";
+	clocks = <&audio_xtal>;
 };

Documentation/devicetree/bindings/sound/tscs454.txt

+TSCS454 Audio CODEC
+
+Required Properties:
+
+	- compatible :	"tempo,tscs454"
+
+	- reg : 	<0x69>
+
+	- clock-names:	Must one of  the following "xtal", "mclk1", "mclk2"
+
+	- clocks:	phandle of the clock that provides the codec sysclk
+
+	Note: If clock is not provided then bit clock is assumed
+
+Example:
+
+redwood: codec@69 {
+	#sound-dai-cells = <1>;
+	compatible = "tempo,tscs454";
+	reg = <0x69>;
+	clock-names = "mclk1";
+	clocks = <&audio_mclk>;
+};

Documentation/devicetree/bindings/sound/wm8510.txt

@@ -12,7 +12,7 @@ Required properties:
 
 Example:
 
-codec: wm8510@1a {
+wm8510: codec@1a {
 	compatible = "wlf,wm8510";
 	reg = <0x1a>;
 };

Documentation/devicetree/bindings/sound/wm8523.txt

@@ -10,7 +10,7 @@ Required properties:
 
 Example:
 
-codec: wm8523@1a {
+wm8523: codec@1a {
 	compatible = "wlf,wm8523";
 	reg = <0x1a>;
 };

Documentation/devicetree/bindings/sound/wm8524.txt

@@ -10,7 +10,7 @@ Required properties:
 
 Example:
 
-codec: wm8524 {
+wm8524: codec {
 	compatible = "wlf,wm8524";
 	wlf,mute-gpios = <&gpio1 8 GPIO_ACTIVE_LOW>;
 };

Documentation/devicetree/bindings/sound/wm8580.txt

@@ -10,7 +10,7 @@ Required properties:
 
 Example:
 
-codec: wm8580@1a {
+wm8580: codec@1a {
 	compatible = "wlf,wm8580";
 	reg = <0x1a>;
 };

Documentation/devicetree/bindings/sound/wm8711.txt

@@ -12,7 +12,7 @@ Required properties:
 
 Example:
 
-codec: wm8711@1a {
+wm8711: codec@1a {
 	compatible = "wlf,wm8711";
 	reg = <0x1a>;
 };

Documentation/devicetree/bindings/sound/wm8728.txt

@@ -12,7 +12,7 @@ Required properties:
 
 Example:
 
-codec: wm8728@1a {
+wm8728: codec@1a {
 	compatible = "wlf,wm8728";
 	reg = <0x1a>;
 };

Documentation/devicetree/bindings/sound/wm8731.txt

@@ -12,7 +12,7 @@ Required properties:
 
 Example:
 
-codec: wm8731@1a {
+wm8731: codec@1a {
 	compatible = "wlf,wm8731";
 	reg = <0x1a>;
 };

Documentation/devicetree/bindings/sound/wm8737.txt

@@ -12,7 +12,7 @@ Required properties:
 
 Example:
 
-codec: wm8737@1a {
+wm8737: codec@1a {
 	compatible = "wlf,wm8737";
 	reg = <0x1a>;
 };

Documentation/devicetree/bindings/sound/wm8741.txt

@@ -21,7 +21,7 @@ Optional properties:
 
 Example:
 
-codec: wm8741@1a {
+wm8741: codec@1a {
 	compatible = "wlf,wm8741";
 	reg = <0x1a>;
 

Documentation/devicetree/bindings/sound/wm8750.txt

@@ -12,7 +12,7 @@ Required properties:
 
 Example:
 
-codec: wm8750@1a {
+wm8750: codec@1a {
 	compatible = "wlf,wm8750";
 	reg = <0x1a>;
 };

Documentation/devicetree/bindings/sound/wm8753.txt

@@ -34,7 +34,7 @@ Pins on the device (for linking into audio routes):
 
 Example:
 
-codec: wm8753@1a {
+wm8753: codec@1a {
 	compatible = "wlf,wm8753";
 	reg = <0x1a>;
 };

Documentation/devicetree/bindings/sound/wm8770.txt

@@ -10,7 +10,7 @@ Required properties:
 
 Example:
 
-codec: wm8770@1 {
+wm8770: codec@1 {
 	compatible = "wlf,wm8770";
 	reg = <1>;
 };

Documentation/devicetree/bindings/sound/wm8776.txt

@@ -12,7 +12,7 @@ Required properties:
 
 Example:
 
-codec: wm8776@1a {
+wm8776: codec@1a {
 	compatible = "wlf,wm8776";
 	reg = <0x1a>;
 };

Documentation/devicetree/bindings/sound/wm8804.txt

@@ -19,7 +19,7 @@ Optional properties:
 
 Example:
 
-codec: wm8804@1a {
+wm8804: codec@1a {
 	compatible = "wlf,wm8804";
 	reg = <0x1a>;
 };

Documentation/devicetree/bindings/sound/wm8903.txt

@@ -57,7 +57,7 @@ Pins on the device (for linking into audio routes):
 
 Example:
 
-codec: wm8903@1a {
+wm8903: codec@1a {
 	compatible = "wlf,wm8903";
 	reg = <0x1a>;
 	interrupts = < 347 >;

Documentation/devicetree/bindings/sound/wm8960.txt

@@ -23,7 +23,7 @@ Optional properties:
 
 Example:
 
-codec: wm8960@1a {
+wm8960: codec@1a {
 	compatible = "wlf,wm8960";
 	reg = <0x1a>;
 

Documentation/devicetree/bindings/sound/wm8962.txt

@@ -24,7 +24,7 @@ Optional properties:
 
 Example:
 
-codec: wm8962@1a {
+wm8962: codec@1a {
 	compatible = "wlf,wm8962";
 	reg = <0x1a>;
 

Documentation/devicetree/bindings/sound/wm8994.txt

@@ -59,7 +59,7 @@ Optional properties:
 
 Example:
 
-codec: wm8994@1a {
+wm8994: codec@1a {
 	compatible = "wlf,wm8994";
 	reg = <0x1a>;
 

Documentation/sound/alsa-configuration.rst

@@ -2224,6 +2224,13 @@ quirk_alias
     Quirk alias list, pass strings like ``0123abcd:5678beef``, which
     applies the existing quirk for the device 5678:beef to a new
     device 0123:abcd.
+use_vmalloc
+    Use vmalloc() for allocations of the PCM buffers (default: yes).
+    For architectures with non-coherent memory like ARM or MIPS, the
+    mmap access may give inconsistent results with vmalloc'ed
+    buffers.  If mmap is used on such architectures, turn off this
+    option, so that the DMA-coherent buffers are allocated and used
+    instead.
 
 This module supports multiple devices, autoprobe and hotplugging.
 

Documentation/sound/hd-audio/models.rst

@@ -263,6 +263,8 @@ hp-dock
     HP dock support
 mute-led-gpio
     Mute LED control via GPIO
+hp-mic-fix
+    Fix for headset mic pin on HP boxes
 
 STAC9200
 ========

Documentation/sound/soc/codec.rst

@@ -179,12 +179,12 @@ i.e.
 
   static int wm8974_mute(struct snd_soc_dai *dai, int mute)
   {
-	struct snd_soc_codec *codec = dai->codec;
-	u16 mute_reg = snd_soc_read(codec, WM8974_DAC) & 0xffbf;
+	struct snd_soc_component *component = dai->component;
+	u16 mute_reg = snd_soc_component_read32(component, WM8974_DAC) & 0xffbf;
 
 	if (mute)
-		snd_soc_write(codec, WM8974_DAC, mute_reg | 0x40);
+		snd_soc_component_write(component, WM8974_DAC, mute_reg | 0x40);
 	else
-		snd_soc_write(codec, WM8974_DAC, mute_reg);
+		snd_soc_component_write(component, WM8974_DAC, mute_reg);
 	return 0;
   }

Documentation/sound/soc/platform.rst

@@ -23,30 +23,26 @@ The platform DMA driver optionally supports the following ALSA operations:-
   };
 
 The platform driver exports its DMA functionality via struct
-snd_soc_platform_driver:-
+snd_soc_component_driver:-
 ::
 
-  struct snd_soc_platform_driver {
-	char *name;
+  struct snd_soc_component_driver {
+	const char *name;
 
-	int (*probe)(struct platform_device *pdev);
-	int (*remove)(struct platform_device *pdev);
-	int (*suspend)(struct platform_device *pdev, struct snd_soc_cpu_dai *cpu_dai);
-	int (*resume)(struct platform_device *pdev, struct snd_soc_cpu_dai *cpu_dai);
+	...
+	int (*probe)(struct snd_soc_component *);
+	void (*remove)(struct snd_soc_component *);
+	int (*suspend)(struct snd_soc_component *);
+	int (*resume)(struct snd_soc_component *);
 
 	/* pcm creation and destruction */
-	int (*pcm_new)(struct snd_card *, struct snd_soc_codec_dai *, struct snd_pcm *);
+	int (*pcm_new)(struct snd_soc_pcm_runtime *);
 	void (*pcm_free)(struct snd_pcm *);
 
-	/*
-	 * For platform caused delay reporting.
-	 * Optional.
-	 */
-	snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *,
-		struct snd_soc_dai *);
-
-	/* platform stream ops */
-	struct snd_pcm_ops *pcm_ops;
+	...
+	const struct snd_pcm_ops *ops;
+	const struct snd_compr_ops *compr_ops;
+	...
   };
 
 Please refer to the ALSA driver documentation for details of audio DMA.

MAINTAINERS

@@ -7082,14 +7082,13 @@ L:	linux-fbdev@vger.kernel.org
 S:	Maintained
 F:	drivers/video/fbdev/i810/
 
-INTEL ASoC BDW/HSW DRIVERS
+INTEL ASoC DRIVERS
+M:	Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
+M:	Liam Girdwood <liam.r.girdwood@linux.intel.com>
 M:	Jie Yang <yang.jie@linux.intel.com>
 L:	alsa-devel@alsa-project.org (moderated for non-subscribers)
 S:	Supported
-F:	sound/soc/intel/common/sst-dsp*
-F:	sound/soc/intel/common/sst-firmware.c
-F:	sound/soc/intel/boards/broadwell.c
-F:	sound/soc/intel/haswell/
+F:	sound/soc/intel/
 
 INTEL C600 SERIES SAS CONTROLLER DRIVER
 M:	Intel SCU Linux support <intel-linux-scu@intel.com>
@@ -15542,6 +15541,13 @@ S:	Supported
 F:	arch/x86/xen/*swiotlb*
 F:	drivers/xen/*swiotlb*
 
+XEN SOUND FRONTEND DRIVER
+M:	Oleksandr Andrushchenko <oleksandr_andrushchenko@epam.com>
+L:	xen-devel@lists.xenproject.org (moderated for non-subscribers)
+L:	alsa-devel@alsa-project.org (moderated for non-subscribers)
+S:	Supported
+F:	sound/xen/*
+
 XFS FILESYSTEM
 M:	Darrick J. Wong <darrick.wong@oracle.com>
 M:	linux-xfs@vger.kernel.org

arch/arm/mach-ep93xx/core.c

@@ -635,6 +635,7 @@ EXPORT_SYMBOL(ep93xx_keypad_release_gpio);
  *************************************************************************/
 static struct resource ep93xx_i2s_resource[] = {
 	DEFINE_RES_MEM(EP93XX_I2S_PHYS_BASE, 0x100),
+	DEFINE_RES_IRQ(IRQ_EP93XX_SAI),
 };
 
 static struct platform_device ep93xx_i2s_device = {

drivers/media/i2c/tda1997x.c

@@ -2444,7 +2444,7 @@ static int tda1997x_pcm_startup(struct snd_pcm_substream *substream,
 				struct snd_soc_dai *dai)
 {
 	struct tda1997x_state *state = snd_soc_dai_get_drvdata(dai);
-	struct snd_soc_codec *codec = dai->codec;
+	struct snd_soc_component *component = dai->component;
 	struct snd_pcm_runtime *rtd = substream->runtime;
 	int rate, err;
 
@@ -2452,11 +2452,11 @@ static int tda1997x_pcm_startup(struct snd_pcm_substream *substream,
 	err = snd_pcm_hw_constraint_minmax(rtd, SNDRV_PCM_HW_PARAM_RATE,
 					   rate, rate);
 	if (err < 0) {
-		dev_err(codec->dev, "failed to constrain samplerate to %dHz\n",
+		dev_err(component->dev, "failed to constrain samplerate to %dHz\n",
 			rate);
 		return err;
 	}
-	dev_info(codec->dev, "set samplerate constraint to %dHz\n", rate);
+	dev_info(component->dev, "set samplerate constraint to %dHz\n", rate);
 
 	return 0;
 }
@@ -2479,20 +2479,22 @@ static struct snd_soc_dai_driver tda1997x_audio_dai = {
 	.ops = &tda1997x_dai_ops,
 };
 
-static int tda1997x_codec_probe(struct snd_soc_codec *codec)
+static int tda1997x_codec_probe(struct snd_soc_component *component)
 {
 	return 0;
 }
 
-static int tda1997x_codec_remove(struct snd_soc_codec *codec)
+static void tda1997x_codec_remove(struct snd_soc_component *component)
 {
-	return 0;
 }
 
-static struct snd_soc_codec_driver tda1997x_codec_driver = {
-	.probe = tda1997x_codec_probe,
-	.remove = tda1997x_codec_remove,
-	.reg_word_size = sizeof(u16),
+static struct snd_soc_component_driver tda1997x_codec_driver = {
+	.probe			= tda1997x_codec_probe,
+	.remove			= tda1997x_codec_remove,
+	.idle_bias_on		= 1,
+	.use_pmdown_time	= 1,
+	.endianness		= 1,
+	.non_legacy_dai_naming	= 1,
 };
 
 static int tda1997x_probe(struct i2c_client *client,
@@ -2737,7 +2739,7 @@ static int tda1997x_probe(struct i2c_client *client,
 		else
 			formats = SNDRV_PCM_FMTBIT_S16_LE;
 		tda1997x_audio_dai.capture.formats = formats;
-		ret = snd_soc_register_codec(&state->client->dev,
+		ret = devm_snd_soc_register_component(&state->client->dev,
 					     &tda1997x_codec_driver,
 					     &tda1997x_audio_dai, 1);
 		if (ret) {
@@ -2782,7 +2784,6 @@ static int tda1997x_remove(struct i2c_client *client)
 	struct tda1997x_platform_data *pdata = &state->pdata;
 
 	if (pdata->audout_format) {
-		snd_soc_unregister_codec(&client->dev);
 		mutex_destroy(&state->audio_lock);
 	}
 

drivers/soc/qcom/Kconfig

@@ -108,4 +108,13 @@ config QCOM_WCNSS_CTRL
 	  Client driver for the WCNSS_CTRL SMD channel, used to download nv
 	  firmware to a newly booted WCNSS chip.
 
+config QCOM_APR
+	tristate "Qualcomm APR Bus (Asynchronous Packet Router)"
+	depends on ARCH_QCOM
+	depends on RPMSG
+	help
+          Enable APR IPC protocol support between
+          application processor and QDSP6. APR is
+          used by audio driver to configure QDSP6
+          ASM, ADM and AFE modules.
 endmenu

drivers/soc/qcom/Makefile

@@ -12,3 +12,4 @@ obj-$(CONFIG_QCOM_SMEM_STATE) += smem_state.o
 obj-$(CONFIG_QCOM_SMP2P)	+= smp2p.o
 obj-$(CONFIG_QCOM_SMSM)	+= smsm.o
 obj-$(CONFIG_QCOM_WCNSS_CTRL) += wcnss_ctrl.o
+obj-$(CONFIG_QCOM_APR) += apr.o

drivers/soc/qcom/apr.c

+// SPDX-License-Identifier: GPL-2.0
+// Copyright (c) 2011-2017, The Linux Foundation. All rights reserved.
+// Copyright (c) 2018, Linaro Limited
+
+#include <linux/kernel.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/spinlock.h>
+#include <linux/idr.h>
+#include <linux/slab.h>
+#include <linux/of_device.h>
+#include <linux/soc/qcom/apr.h>
+#include <linux/rpmsg.h>
+#include <linux/of.h>
+
+struct apr {
+	struct rpmsg_endpoint *ch;
+	struct device *dev;
+	spinlock_t svcs_lock;
+	struct idr svcs_idr;
+	int dest_domain_id;
+};
+
+/**
+ * apr_send_pkt() - Send a apr message from apr device
+ *
+ * @adev: Pointer to previously registered apr device.
+ * @pkt: Pointer to apr packet to send
+ *
+ * Return: Will be an negative on packet size on success.
+ */
+int apr_send_pkt(struct apr_device *adev, struct apr_pkt *pkt)
+{
+	struct apr *apr = dev_get_drvdata(adev->dev.parent);
+	struct apr_hdr *hdr;
+	unsigned long flags;
+	int ret;
+
+	spin_lock_irqsave(&adev->lock, flags);
+
+	hdr = &pkt->hdr;
+	hdr->src_domain = APR_DOMAIN_APPS;
+	hdr->src_svc = adev->svc_id;
+	hdr->dest_domain = adev->domain_id;
+	hdr->dest_svc = adev->svc_id;
+
+	ret = rpmsg_trysend(apr->ch, pkt, hdr->pkt_size);
+	spin_unlock_irqrestore(&adev->lock, flags);
+
+	return ret ? ret : hdr->pkt_size;
+}
+EXPORT_SYMBOL_GPL(apr_send_pkt);
+
+static void apr_dev_release(struct device *dev)
+{
+	struct apr_device *adev = to_apr_device(dev);
+
+	kfree(adev);
+}
+
+static int apr_callback(struct rpmsg_device *rpdev, void *buf,
+				  int len, void *priv, u32 addr)
+{
+	struct apr *apr = dev_get_drvdata(&rpdev->dev);
+	uint16_t hdr_size, msg_type, ver, svc_id;
+	struct apr_device *svc = NULL;
+	struct apr_driver *adrv = NULL;
+	struct apr_resp_pkt resp;
+	struct apr_hdr *hdr;
+	unsigned long flags;
+
+	if (len <= APR_HDR_SIZE) {
+		dev_err(apr->dev, "APR: Improper apr pkt received:%p %d\n",
+			buf, len);
+		return -EINVAL;
+	}
+
+	hdr = buf;
+	ver = APR_HDR_FIELD_VER(hdr->hdr_field);
+	if (ver > APR_PKT_VER + 1)
+		return -EINVAL;
+
+	hdr_size = APR_HDR_FIELD_SIZE_BYTES(hdr->hdr_field);
+	if (hdr_size < APR_HDR_SIZE) {
+		dev_err(apr->dev, "APR: Wrong hdr size:%d\n", hdr_size);
+		return -EINVAL;
+	}
+
+	if (hdr->pkt_size < APR_HDR_SIZE || hdr->pkt_size != len) {
+		dev_err(apr->dev, "APR: Wrong paket size\n");
+		return -EINVAL;
+	}
+
+	msg_type = APR_HDR_FIELD_MT(hdr->hdr_field);
+	if (msg_type >= APR_MSG_TYPE_MAX) {
+		dev_err(apr->dev, "APR: Wrong message type: %d\n", msg_type);
+		return -EINVAL;
+	}
+
+	if (hdr->src_domain >= APR_DOMAIN_MAX ||
+			hdr->dest_domain >= APR_DOMAIN_MAX ||
+			hdr->src_svc >= APR_SVC_MAX ||
+			hdr->dest_svc >= APR_SVC_MAX) {
+		dev_err(apr->dev, "APR: Wrong APR header\n");
+		return -EINVAL;
+	}
+
+	svc_id = hdr->dest_svc;
+	spin_lock_irqsave(&apr->svcs_lock, flags);
+	svc = idr_find(&apr->svcs_idr, svc_id);
+	if (svc && svc->dev.driver)
+		adrv = to_apr_driver(svc->dev.driver);
+	spin_unlock_irqrestore(&apr->svcs_lock, flags);
+
+	if (!adrv) {
+		dev_err(apr->dev, "APR: service is not registered\n");
+		return -EINVAL;
+	}
+
+	resp.hdr = *hdr;
+	resp.payload_size = hdr->pkt_size - hdr_size;
+
+	/*
+	 * NOTE: hdr_size is not same as APR_HDR_SIZE as remote can include
+	 * optional headers in to apr_hdr which should be ignored
+	 */
+	if (resp.payload_size > 0)
+		resp.payload = buf + hdr_size;
+
+	adrv->callback(svc, &resp);
+
+	return 0;
+}
+
+static int apr_device_match(struct device *dev, struct device_driver *drv)
+{
+	struct apr_device *adev = to_apr_device(dev);
+	struct apr_driver *adrv = to_apr_driver(drv);
+	const struct apr_device_id *id = adrv->id_table;
+
+	/* Attempt an OF style match first */
+	if (of_driver_match_device(dev, drv))
+		return 1;
+
+	if (!id)
+		return 0;
+
+	while (id->domain_id != 0 || id->svc_id != 0) {
+		if (id->domain_id == adev->domain_id &&
+		    id->svc_id == adev->svc_id)
+			return 1;
+		id++;
+	}
+
+	return 0;
+}
+
+static int apr_device_probe(struct device *dev)
+{
+	struct apr_device *adev = to_apr_device(dev);
+	struct apr_driver *adrv = to_apr_driver(dev->driver);
+
+	return adrv->probe(adev);
+}
+
+static int apr_device_remove(struct device *dev)
+{
+	struct apr_device *adev = to_apr_device(dev);
+	struct apr_driver *adrv;
+	struct apr *apr = dev_get_drvdata(adev->dev.parent);
+
+	if (dev->driver) {
+		adrv = to_apr_driver(dev->driver);
+		if (adrv->remove)
+			adrv->remove(adev);
+		spin_lock(&apr->svcs_lock);
+		idr_remove(&apr->svcs_idr, adev->svc_id);
+		spin_unlock(&apr->svcs_lock);
+	}
+
+	return 0;
+}
+
+static int apr_uevent(struct device *dev, struct kobj_uevent_env *env)
+{
+	struct apr_device *adev = to_apr_device(dev);
+	int ret;
+
+	ret = of_device_uevent_modalias(dev, env);
+	if (ret != -ENODEV)
+		return ret;
+
+	return add_uevent_var(env, "MODALIAS=apr:%s", adev->name);
+}
+
+struct bus_type aprbus = {
+	.name		= "aprbus",
+	.match		= apr_device_match,
+	.probe		= apr_device_probe,
+	.uevent		= apr_uevent,
+	.remove		= apr_device_remove,
+};
+EXPORT_SYMBOL_GPL(aprbus);
+
+static int apr_add_device(struct device *dev, struct device_node *np,
+			  const struct apr_device_id *id)
+{
+	struct apr *apr = dev_get_drvdata(dev);
+	struct apr_device *adev = NULL;
+	int ret;
+
+	adev = kzalloc(sizeof(*adev), GFP_KERNEL);
+	if (!adev)
+		return -ENOMEM;
+
+	spin_lock_init(&adev->lock);
+
+	adev->svc_id = id->svc_id;
+	adev->domain_id = id->domain_id;
+	adev->version = id->svc_version;
+	if (np)
+		strncpy(adev->name, np->name, APR_NAME_SIZE);
+	else
+		strncpy(adev->name, id->name, APR_NAME_SIZE);
+
+	dev_set_name(&adev->dev, "aprsvc:%s:%x:%x", adev->name,
+		     id->domain_id, id->svc_id);
+
+	adev->dev.bus = &aprbus;
+	adev->dev.parent = dev;
+	adev->dev.of_node = np;
+	adev->dev.release = apr_dev_release;
+	adev->dev.driver = NULL;
+
+	spin_lock(&apr->svcs_lock);
+	idr_alloc(&apr->svcs_idr, adev, id->svc_id,
+		  id->svc_id + 1, GFP_ATOMIC);
+	spin_unlock(&apr->svcs_lock);
+
+	dev_info(dev, "Adding APR dev: %s\n", dev_name(&adev->dev));
+
+	ret = device_register(&adev->dev);
+	if (ret) {
+		dev_err(dev, "device_register failed: %d\n", ret);
+		put_device(&adev->dev);
+	}
+
+	return ret;
+}
+
+static void of_register_apr_devices(struct device *dev)
+{
+	struct apr *apr = dev_get_drvdata(dev);
+	struct device_node *node;
+
+	for_each_child_of_node(dev->of_node, node) {
+		struct apr_device_id id = { {0} };
+
+		if (of_property_read_u32(node, "reg", &id.svc_id))
+			continue;
+
+		id.domain_id = apr->dest_domain_id;
+
+		if (apr_add_device(dev, node, &id))
+			dev_err(dev, "Failed to add apr %d svc\n", id.svc_id);
+	}
+}
+
+static int apr_probe(struct rpmsg_device *rpdev)
+{
+	struct device *dev = &rpdev->dev;
+	struct apr *apr;
+	int ret;
+
+	apr = devm_kzalloc(dev, sizeof(*apr), GFP_KERNEL);
+	if (!apr)
+		return -ENOMEM;
+
+	ret = of_property_read_u32(dev->of_node, "reg", &apr->dest_domain_id);
+	if (ret) {
+		dev_err(dev, "APR Domain ID not specified in DT\n");
+		return ret;
+	}
+
+	dev_set_drvdata(dev, apr);
+	apr->ch = rpdev->ept;
+	apr->dev = dev;
+	spin_lock_init(&apr->svcs_lock);
+	idr_init(&apr->svcs_idr);
+	of_register_apr_devices(dev);
+
+	return 0;
+}
+
+static int apr_remove_device(struct device *dev, void *null)
+{
+	struct apr_device *adev = to_apr_device(dev);
+
+	device_unregister(&adev->dev);
+
+	return 0;
+}
+
+static void apr_remove(struct rpmsg_device *rpdev)
+{
+	device_for_each_child(&rpdev->dev, NULL, apr_remove_device);
+}
+
+/*
+ * __apr_driver_register() - Client driver registration with aprbus
+ *
+ * @drv:Client driver to be associated with client-device.
+ * @owner: owning module/driver
+ *
+ * This API will register the client driver with the aprbus
+ * It is called from the driver's module-init function.
+ */
+int __apr_driver_register(struct apr_driver *drv, struct module *owner)
+{
+	drv->driver.bus = &aprbus;
+	drv->driver.owner = owner;
+
+	return driver_register(&drv->driver);
+}
+EXPORT_SYMBOL_GPL(__apr_driver_register);
+
+/*
+ * apr_driver_unregister() - Undo effect of apr_driver_register
+ *
+ * @drv: Client driver to be unregistered
+ */
+void apr_driver_unregister(struct apr_driver *drv)
+{
+	driver_unregister(&drv->driver);
+}
+EXPORT_SYMBOL_GPL(apr_driver_unregister);
+
+static const struct of_device_id apr_of_match[] = {
+	{ .compatible = "qcom,apr"},
+	{ .compatible = "qcom,apr-v2"},
+	{}
+};
+MODULE_DEVICE_TABLE(of, apr_of_match);
+
+static struct rpmsg_driver apr_driver = {
+	.probe = apr_probe,
+	.remove = apr_remove,
+	.callback = apr_callback,
+	.drv = {
+		.name = "qcom,apr",
+		.of_match_table = apr_of_match,
+	},
+};
+
+static int __init apr_init(void)
+{
+	int ret;
+
+	ret = bus_register(&aprbus);
+	if (!ret)
+		ret = register_rpmsg_driver(&apr_driver);
+	else
+		bus_unregister(&aprbus);
+
+	return ret;
+}
+
+static void __exit apr_exit(void)
+{
+	bus_unregister(&aprbus);
+	unregister_rpmsg_driver(&apr_driver);
+}
+
+subsys_initcall(apr_init);
+module_exit(apr_exit);
+
+MODULE_LICENSE("GPL v2");
+MODULE_DESCRIPTION("Qualcomm APR Bus");

include/dt-bindings/soc/qcom,apr.h

+/* SPDX-License-Identifier: GPL-2.0 */
+#ifndef __DT_BINDINGS_QCOM_APR_H
+#define __DT_BINDINGS_QCOM_APR_H
+
+/* Domain IDs */
+#define APR_DOMAIN_SIM		0x1
+#define APR_DOMAIN_PC		0x2
+#define APR_DOMAIN_MODEM	0x3
+#define APR_DOMAIN_ADSP		0x4
+#define APR_DOMAIN_APPS		0x5
+#define APR_DOMAIN_MAX		0x6
+
+/* ADSP service IDs */
+#define APR_SVC_ADSP_CORE	0x3
+#define APR_SVC_AFE		0x4
+#define APR_SVC_VSM		0x5
+#define APR_SVC_VPM		0x6
+#define APR_SVC_ASM		0x7
+#define APR_SVC_ADM		0x8
+#define APR_SVC_ADSP_MVM	0x09
+#define APR_SVC_ADSP_CVS	0x0A
+#define APR_SVC_ADSP_CVP	0x0B
+#define APR_SVC_USM		0x0C
+#define APR_SVC_LSM		0x0D
+#define APR_SVC_VIDC		0x16
+#define APR_SVC_MAX		0x17
+
+#endif /* __DT_BINDINGS_QCOM_APR_H */

include/dt-bindings/sound/fsl-imx-audmux.h

@@ -25,6 +25,13 @@
 #define MX51_AUDMUX_PORT6		5
 #define MX51_AUDMUX_PORT7		6
 
+/*
+ * TFCSEL/RFCSEL (i.MX27) or TFSEL/TCSEL/RFSEL/RCSEL (i.MX31/51/53/6Q)
+ * can be sourced from Rx/Tx.
+ */
+#define IMX_AUDMUX_RXFS			0x8
+#define IMX_AUDMUX_RXCLK		0x8
+
 /* Register definitions for the i.MX21/27 Digital Audio Multiplexer */
 #define IMX_AUDMUX_V1_PCR_INMMASK(x)	((x) & 0xff)
 #define IMX_AUDMUX_V1_PCR_INMEN		(1 << 8)

include/dt-bindings/sound/qcom,q6afe.h

+/* SPDX-License-Identifier: GPL-2.0 */
+#ifndef __DT_BINDINGS_Q6_AFE_H__
+#define __DT_BINDINGS_Q6_AFE_H__
+
+/* Audio Front End (AFE) virtual ports IDs */
+#define HDMI_RX		1
+#define SLIMBUS_0_RX    2
+#define SLIMBUS_0_TX    3
+#define SLIMBUS_1_RX    4
+#define SLIMBUS_1_TX    5
+#define SLIMBUS_2_RX    6
+#define SLIMBUS_2_TX    7
+#define SLIMBUS_3_RX    8
+#define SLIMBUS_3_TX    9
+#define SLIMBUS_4_RX    10
+#define SLIMBUS_4_TX    11
+#define SLIMBUS_5_RX    12
+#define SLIMBUS_5_TX    13
+#define SLIMBUS_6_RX    14
+#define SLIMBUS_6_TX    15
+#define PRIMARY_MI2S_RX		16
+#define PRIMARY_MI2S_TX		17
+#define SECONDARY_MI2S_RX	18
+#define SECONDARY_MI2S_TX	19
+#define TERTIARY_MI2S_RX	20
+#define TERTIARY_MI2S_TX	21
+#define QUATERNARY_MI2S_RX	22
+#define QUATERNARY_MI2S_TX	23
+#define PRIMARY_TDM_RX_0	24
+#define PRIMARY_TDM_TX_0	25
+#define PRIMARY_TDM_RX_1	26
+#define PRIMARY_TDM_TX_1	27
+#define PRIMARY_TDM_RX_2	28
+#define PRIMARY_TDM_TX_2	29
+#define PRIMARY_TDM_RX_3	30
+#define PRIMARY_TDM_TX_3	31
+#define PRIMARY_TDM_RX_4	32
+#define PRIMARY_TDM_TX_4	33
+#define PRIMARY_TDM_RX_5	34
+#define PRIMARY_TDM_TX_5	35
+#define PRIMARY_TDM_RX_6	36
+#define PRIMARY_TDM_TX_6	37
+#define PRIMARY_TDM_RX_7	38
+#define PRIMARY_TDM_TX_7	39
+#define SECONDARY_TDM_RX_0	40
+#define SECONDARY_TDM_TX_0	41
+#define SECONDARY_TDM_RX_1	42
+#define SECONDARY_TDM_TX_1	43
+#define SECONDARY_TDM_RX_2	44
+#define SECONDARY_TDM_TX_2	45
+#define SECONDARY_TDM_RX_3	46
+#define SECONDARY_TDM_TX_3	47
+#define SECONDARY_TDM_RX_4	48
+#define SECONDARY_TDM_TX_4	49
+#define SECONDARY_TDM_RX_5	50
+#define SECONDARY_TDM_TX_5	51
+#define SECONDARY_TDM_RX_6	52
+#define SECONDARY_TDM_TX_6	53
+#define SECONDARY_TDM_RX_7	54
+#define SECONDARY_TDM_TX_7	55
+#define TERTIARY_TDM_RX_0	56
+#define TERTIARY_TDM_TX_0	57
+#define TERTIARY_TDM_RX_1	58
+#define TERTIARY_TDM_TX_1	59
+#define TERTIARY_TDM_RX_2	60
+#define TERTIARY_TDM_TX_2	61
+#define TERTIARY_TDM_RX_3	62
+#define TERTIARY_TDM_TX_3	63
+#define TERTIARY_TDM_RX_4	64
+#define TERTIARY_TDM_TX_4	65
+#define TERTIARY_TDM_RX_5	66
+#define TERTIARY_TDM_TX_5	67
+#define TERTIARY_TDM_RX_6	68
+#define TERTIARY_TDM_TX_6	69
+#define TERTIARY_TDM_RX_7	70
+#define TERTIARY_TDM_TX_7	71
+#define QUATERNARY_TDM_RX_0	72
+#define QUATERNARY_TDM_TX_0	73
+#define QUATERNARY_TDM_RX_1	74
+#define QUATERNARY_TDM_TX_1	75
+#define QUATERNARY_TDM_RX_2	76
+#define QUATERNARY_TDM_TX_2	77
+#define QUATERNARY_TDM_RX_3	78
+#define QUATERNARY_TDM_TX_3	79
+#define QUATERNARY_TDM_RX_4	80
+#define QUATERNARY_TDM_TX_4	81
+#define QUATERNARY_TDM_RX_5	82
+#define QUATERNARY_TDM_TX_5	83
+#define QUATERNARY_TDM_RX_6	84
+#define QUATERNARY_TDM_TX_6	85
+#define QUATERNARY_TDM_RX_7	86
+#define QUATERNARY_TDM_TX_7	87
+#define QUINARY_TDM_RX_0	88
+#define QUINARY_TDM_TX_0	89
+#define QUINARY_TDM_RX_1	90
+#define QUINARY_TDM_TX_1	91
+#define QUINARY_TDM_RX_2	92
+#define QUINARY_TDM_TX_2	93
+#define QUINARY_TDM_RX_3	94
+#define QUINARY_TDM_TX_3	95
+#define QUINARY_TDM_RX_4	96
+#define QUINARY_TDM_TX_4	97
+#define QUINARY_TDM_RX_5	98
+#define QUINARY_TDM_TX_5	99
+#define QUINARY_TDM_RX_6	100
+#define QUINARY_TDM_TX_6	101
+#define QUINARY_TDM_RX_7	102
+#define QUINARY_TDM_TX_7	103
+
+#endif /* __DT_BINDINGS_Q6_AFE_H__ */
+

include/dt-bindings/sound/qcom,q6asm.h

+/* SPDX-License-Identifier: GPL-2.0 */
+#ifndef __DT_BINDINGS_Q6_ASM_H__
+#define __DT_BINDINGS_Q6_ASM_H__
+
+#define	MSM_FRONTEND_DAI_MULTIMEDIA1	0
+#define	MSM_FRONTEND_DAI_MULTIMEDIA2	1
+#define	MSM_FRONTEND_DAI_MULTIMEDIA3	2
+#define	MSM_FRONTEND_DAI_MULTIMEDIA4	3
+#define	MSM_FRONTEND_DAI_MULTIMEDIA5	4
+#define	MSM_FRONTEND_DAI_MULTIMEDIA6	5
+#define	MSM_FRONTEND_DAI_MULTIMEDIA7	6
+#define	MSM_FRONTEND_DAI_MULTIMEDIA8	7
+#define	MSM_FRONTEND_DAI_MULTIMEDIA9	8
+#define	MSM_FRONTEND_DAI_MULTIMEDIA10	9
+#define	MSM_FRONTEND_DAI_MULTIMEDIA11	10
+#define	MSM_FRONTEND_DAI_MULTIMEDIA12	11
+#define	MSM_FRONTEND_DAI_MULTIMEDIA13	12
+#define	MSM_FRONTEND_DAI_MULTIMEDIA14	13
+#define	MSM_FRONTEND_DAI_MULTIMEDIA15	14
+#define	MSM_FRONTEND_DAI_MULTIMEDIA16	15
+
+#endif /* __DT_BINDINGS_Q6_ASM_H__ */

include/dt-bindings/sound/rt5640.h

+/* SPDX-License-Identifier: GPL-2.0 */
+#ifndef __DT_RT5640_H
+#define __DT_RT5640_H
+
+#define RT5640_DMIC1_DATA_PIN_NONE	0
+#define RT5640_DMIC1_DATA_PIN_IN1P	1
+#define RT5640_DMIC1_DATA_PIN_GPIO3	2
+
+#define RT5640_DMIC2_DATA_PIN_NONE	0
+#define RT5640_DMIC2_DATA_PIN_IN1N	1
+#define RT5640_DMIC2_DATA_PIN_GPIO4	2
+
+#define RT5640_JD_SRC_GPIO1		1
+#define RT5640_JD_SRC_JD1_IN4P		2
+#define RT5640_JD_SRC_JD2_IN4N		3
+#define RT5640_JD_SRC_GPIO2		4
+#define RT5640_JD_SRC_GPIO3		5
+#define RT5640_JD_SRC_GPIO4		6
+
+#define RT5640_OVCD_SF_0P5		0
+#define RT5640_OVCD_SF_0P75		1
+#define RT5640_OVCD_SF_1P0		2
+#define RT5640_OVCD_SF_1P5		3
+
+#endif /* __DT_RT5640_H */

include/linux/mfd/wm8350/audio.h

@@ -617,11 +617,8 @@ struct wm8350_audio_platform_data {
 	u32 codec_current_charge:2;	/* codec current @ vmid charge */
 };
 
-struct snd_soc_codec;
-
 struct wm8350_codec {
 	struct platform_device *pdev;
-	struct snd_soc_codec *codec;
 	struct wm8350_audio_platform_data *platform_data;
 };
 

include/linux/mod_devicetable.h

@@ -471,6 +471,17 @@ struct slim_device_id {
 	kernel_ulong_t driver_data;
 };
 
+#define APR_NAME_SIZE	32
+#define APR_MODULE_PREFIX "apr:"
+
+struct apr_device_id {
+	char name[APR_NAME_SIZE];
+	__u32 domain_id;
+	__u32 svc_id;
+	__u32 svc_version;
+	kernel_ulong_t driver_data;	/* Data private to the driver */
+};
+
 #define SPMI_NAME_SIZE	32
 #define SPMI_MODULE_PREFIX "spmi:"
 

include/linux/soc/qcom/apr.h

+/* SPDX-License-Identifier: GPL-2.0 */
+
+#ifndef __QCOM_APR_H_
+#define __QCOM_APR_H_
+
+#include <linux/spinlock.h>
+#include <linux/device.h>
+#include <linux/mod_devicetable.h>
+#include <dt-bindings/soc/qcom,apr.h>
+
+extern struct bus_type aprbus;
+
+#define APR_HDR_LEN(hdr_len) ((hdr_len)/4)
+
+/*
+ * HEADER field
+ * version:0:3
+ * header_size : 4:7
+ * message_type : 8:9
+ * reserved: 10:15
+ */
+#define APR_HDR_FIELD(msg_type, hdr_len, ver)\
+	(((msg_type & 0x3) << 8) | ((hdr_len & 0xF) << 4) | (ver & 0xF))
+
+#define APR_HDR_SIZE sizeof(struct apr_hdr)
+#define APR_SEQ_CMD_HDR_FIELD APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD, \
+					    APR_HDR_LEN(APR_HDR_SIZE), \
+					    APR_PKT_VER)
+/* Version */
+#define APR_PKT_VER		0x0
+
+/* Command and Response Types */
+#define APR_MSG_TYPE_EVENT	0x0
+#define APR_MSG_TYPE_CMD_RSP	0x1
+#define APR_MSG_TYPE_SEQ_CMD	0x2
+#define APR_MSG_TYPE_NSEQ_CMD	0x3
+#define APR_MSG_TYPE_MAX	0x04
+
+/* APR Basic Response Message */
+#define APR_BASIC_RSP_RESULT 0x000110E8
+#define APR_RSP_ACCEPTED     0x000100BE
+
+struct aprv2_ibasic_rsp_result_t {
+	uint32_t opcode;
+	uint32_t status;
+};
+
+/* hdr field Ver [0:3], Size [4:7], Message type [8:10] */
+#define APR_HDR_FIELD_VER(h)		(h & 0x000F)
+#define APR_HDR_FIELD_SIZE(h)		((h & 0x00F0) >> 4)
+#define APR_HDR_FIELD_SIZE_BYTES(h)	(((h & 0x00F0) >> 4) * 4)
+#define APR_HDR_FIELD_MT(h)		((h & 0x0300) >> 8)
+
+struct apr_hdr {
+	uint16_t hdr_field;
+	uint16_t pkt_size;
+	uint8_t src_svc;
+	uint8_t src_domain;
+	uint16_t src_port;
+	uint8_t dest_svc;
+	uint8_t dest_domain;
+	uint16_t dest_port;
+	uint32_t token;
+	uint32_t opcode;
+} __packed;
+
+struct apr_pkt {
+	struct apr_hdr hdr;
+	uint8_t payload[];
+};
+
+struct apr_resp_pkt {
+	struct apr_hdr hdr;
+	void *payload;
+	int payload_size;
+};
+
+/* Bits 0 to 15 -- Minor version,  Bits 16 to 31 -- Major version */
+#define APR_SVC_MAJOR_VERSION(v)	((v >> 16) & 0xFF)
+#define APR_SVC_MINOR_VERSION(v)	(v & 0xFF)
+
+struct apr_device {
+	struct device	dev;
+	uint16_t	svc_id;
+	uint16_t	domain_id;
+	uint32_t	version;
+	char name[APR_NAME_SIZE];
+	spinlock_t	lock;
+	struct list_head node;
+};
+
+#define to_apr_device(d) container_of(d, struct apr_device, dev)
+
+struct apr_driver {
+	int	(*probe)(struct apr_device *sl);
+	int	(*remove)(struct apr_device *sl);
+	int	(*callback)(struct apr_device *a,
+			    struct apr_resp_pkt *d);
+	struct device_driver		driver;
+	const struct apr_device_id	*id_table;
+};
+
+#define to_apr_driver(d) container_of(d, struct apr_driver, driver)
+
+/*
+ * use a macro to avoid include chaining to get THIS_MODULE
+ */
+#define apr_driver_register(drv) __apr_driver_register(drv, THIS_MODULE)
+
+int __apr_driver_register(struct apr_driver *drv, struct module *owner);
+void apr_driver_unregister(struct apr_driver *drv);
+
+/**
+ * module_apr_driver() - Helper macro for registering a aprbus driver
+ * @__aprbus_driver: aprbus_driver struct
+ *
+ * Helper macro for aprbus drivers which do not do anything special in
+ * module init/exit. This eliminates a lot of boilerplate. Each module
+ * may only use this macro once, and calling it replaces module_init()
+ * and module_exit()
+ */
+#define module_apr_driver(__apr_driver) \
+	module_driver(__apr_driver, apr_driver_register, \
+			apr_driver_unregister)
+
+int apr_send_pkt(struct apr_device *adev, struct apr_pkt *pkt);
+
+#endif /* __QCOM_APR_H_ */

include/linux/usb/audio-v2.h

@@ -189,6 +189,13 @@ struct uac2_iso_endpoint_descriptor {
 #define UAC2_CONTROL_DATA_OVERRUN	(3 << 2)
 #define UAC2_CONTROL_DATA_UNDERRUN	(3 << 4)
 
+/* 5.2.5.4.2 Connector Control Parameter Block */
+struct uac2_connectors_ctl_blk {
+	__u8 bNrChannels;
+	__le32 bmChannelConfig;
+	__u8 iChannelNames;
+} __attribute__((packed));
+
 /* 6.1 Interrupt Data Message */
 
 #define UAC2_INTERRUPT_DATA_MSG_VENDOR	(1 << 0)

include/linux/usb/audio-v3.h

@@ -221,6 +221,12 @@ struct uac3_iso_endpoint_descriptor {
 	__le16 wLockDelay;
 } __attribute__((packed));
 
+/* 5.2.1.6.1 INSERTION CONTROL PARAMETER BLOCK */
+struct uac3_insertion_ctl_blk {
+	__u8 bSize;
+	__u8 bmConInserted;
+} __attribute__ ((packed));
+
 /* 6.1 INTERRUPT DATA MESSAGE */
 struct uac3_interrupt_data_msg {
 	__u8 bInfo;
@@ -392,4 +398,38 @@ struct uac3_interrupt_data_msg {
 #define UAC3_AC_ACTIVE_INTERFACE_CONTROL	0x01
 #define UAC3_AC_POWER_DOMAIN_CONTROL		0x02
 
+/* A.23.5 TERMINAL CONTROL SELECTORS */
+#define UAC3_TE_UNDEFINED			0x00
+#define UAC3_TE_INSERTION			0x01
+#define UAC3_TE_OVERLOAD			0x02
+#define UAC3_TE_UNDERFLOW			0x03
+#define UAC3_TE_OVERFLOW			0x04
+#define UAC3_TE_LATENCY 			0x05
+
+/* 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 */
+#define UAC3_BADD_OT_ID3	3  /* Output Terminal ID3: bTerminalID = 3 */
+#define UAC3_BADD_IT_ID4	4  /* Input Terminal ID4: bTerminalID = 4 */
+#define UAC3_BADD_FU_ID5	5  /* Feature Unit ID5: bUnitID = 5 */
+#define UAC3_BADD_OT_ID6	6  /* Output Terminal ID6: bTerminalID = 6 */
+#define UAC3_BADD_FU_ID7	7  /* Feature Unit ID7: bUnitID = 7 */
+#define UAC3_BADD_MU_ID8	8  /* Mixer Unit ID8: bUnitID = 8 */
+#define UAC3_BADD_CS_ID9	9  /* Clock Source Entity ID9: bClockID = 9 */
+#define UAC3_BADD_PD_ID10	10 /* Power Domain ID10: bPowerDomainID = 10 */
+#define UAC3_BADD_PD_ID11	11 /* Power Domain ID11: bPowerDomainID = 11 */
+
+/* BADD wMaxPacketSize of AS endpoints */
+#define UAC3_BADD_EP_MAXPSIZE_SYNC_MONO_16		0x0060
+#define UAC3_BADD_EP_MAXPSIZE_ASYNC_MONO_16		0x0062
+#define UAC3_BADD_EP_MAXPSIZE_SYNC_MONO_24		0x0090
+#define UAC3_BADD_EP_MAXPSIZE_ASYNC_MONO_24		0x0093
+#define UAC3_BADD_EP_MAXPSIZE_SYNC_STEREO_16		0x00C0
+#define UAC3_BADD_EP_MAXPSIZE_ASYNC_STEREO_16		0x00C4
+#define UAC3_BADD_EP_MAXPSIZE_SYNC_STEREO_24		0x0120
+#define UAC3_BADD_EP_MAXPSIZE_ASYNC_STEREO_24		0x0126
+
+/* BADD sample rate is always fixed to 48kHz */
+#define UAC3_BADD_SAMPLING_RATE				48000
+
 #endif /* __LINUX_USB_AUDIO_V3_H */

include/sound/core.h

@@ -51,7 +51,6 @@ struct completion;
  */
 enum snd_device_type {
 	SNDRV_DEV_LOWLEVEL,
-	SNDRV_DEV_CONTROL,
 	SNDRV_DEV_INFO,
 	SNDRV_DEV_BUS,
 	SNDRV_DEV_CODEC,
@@ -62,6 +61,7 @@ enum snd_device_type {
 	SNDRV_DEV_SEQUENCER,
 	SNDRV_DEV_HWDEP,
 	SNDRV_DEV_JACK,
+	SNDRV_DEV_CONTROL,	/* NOTE: this must be the last one */
 };
 
 enum snd_device_state {

include/sound/emu10k1.h

@@ -1610,7 +1610,7 @@ struct snd_emu10k1_fx8010_pcm {
 	struct snd_pcm_indirect pcm_rec;
 	unsigned int tram_pos;
 	unsigned int tram_shift;
-	struct snd_emu10k1_fx8010_irq *irq;
+	struct snd_emu10k1_fx8010_irq irq;
 };
 
 struct snd_emu10k1_fx8010 {
@@ -1902,7 +1902,7 @@ int snd_emu10k1_fx8010_register_irq_handler(struct snd_emu10k1 *emu,
 					    snd_fx8010_irq_handler_t *handler,
 					    unsigned char gpr_running,
 					    void *private_data,
-					    struct snd_emu10k1_fx8010_irq **r_irq);
+					    struct snd_emu10k1_fx8010_irq *irq);
 int snd_emu10k1_fx8010_unregister_irq_handler(struct snd_emu10k1 *emu,
 					      struct snd_emu10k1_fx8010_irq *irq);
 

include/sound/hdaudio.h

@@ -571,4 +571,9 @@ static inline unsigned int snd_array_index(struct snd_array *array, void *ptr)
 	return (unsigned long)(ptr - array->list) / array->elem_size;
 }
 
+/* a helper macro to iterate for each snd_array element */
+#define snd_array_for_each(array, idx, ptr) \
+	for ((idx) = 0, (ptr) = (array)->list; (idx) < (array)->used; \
+	     (ptr) = snd_array_elem(array, ++(idx)))
+
 #endif /* __SOUND_HDAUDIO_H */

include/sound/memalloc.h

@@ -34,11 +34,9 @@ struct snd_dma_device {
 	struct device *dev;		/* generic device */
 };
 
-#ifndef snd_dma_pci_data
 #define snd_dma_pci_data(pci)	(&(pci)->dev)
 #define snd_dma_isa_data()	NULL
 #define snd_dma_continuous_data(x)	((struct device *)(__force unsigned long)(x))
-#endif
 
 
 /*

include/sound/omap-pcm.h

-/*
- * omap-pcm.h - OMAP PCM driver
- *
- * Copyright (C) 2014 Texas Instruments, Inc.
- *
- * Author: Peter Ujfalusi <peter.ujfalusi@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.
- *
- * 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.
- */
-
-#ifndef __OMAP_PCM_H__
-#define __OMAP_PCM_H__
-
-#if IS_ENABLED(CONFIG_SND_OMAP_SOC)
-int omap_pcm_platform_register(struct device *dev);
-#else
-static inline int omap_pcm_platform_register(struct device *dev)
-{
-	return 0;
-}
-#endif /* CONFIG_SND_OMAP_SOC */
-
-#endif /* __OMAP_PCM_H__ */

include/sound/rt5640.h → include/sound/rt5668.h

 /*
- * linux/sound/rt5640.h -- Platform data for RT5640
+ * linux/sound/rt5668.h -- Platform data for RT5668
  *
- * Copyright 2011 Realtek Microelectronics
+ * 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_RT5640_H
-#define __LINUX_SND_RT5640_H
+#ifndef __LINUX_SND_RT5668_H
+#define __LINUX_SND_RT5668_H
 
-struct rt5640_platform_data {
-	/* IN1 & IN2 & IN3 can optionally be differential */
-	bool in1_diff;
-	bool in2_diff;
-	bool in3_diff;
+enum rt5668_dmic1_data_pin {
+	RT5668_DMIC1_NULL,
+	RT5668_DMIC1_DATA_GPIO2,
+	RT5668_DMIC1_DATA_GPIO5,
+};
+
+enum rt5668_dmic1_clk_pin {
+	RT5668_DMIC1_CLK_GPIO1,
+	RT5668_DMIC1_CLK_GPIO3,
+};
 
-	bool dmic_en;
-	bool dmic1_data_pin; /* 0 = IN1P; 1 = GPIO3 */
-	bool dmic2_data_pin; /* 0 = IN1N; 1 = GPIO4 */
+enum rt5668_jd_src {
+	RT5668_JD_NULL,
+	RT5668_JD1,
+};
+
+struct rt5668_platform_data {
 
 	int ldo1_en; /* GPIO for LDO1_EN */
+
+	enum rt5668_dmic1_data_pin dmic1_data_pin;
+	enum rt5668_dmic1_clk_pin dmic1_clk_pin;
+	enum rt5668_jd_src jd_src;
 };
 
 #endif
+

include/sound/soc-dai.h

@@ -296,8 +296,8 @@ struct snd_soc_dai {
 	struct snd_soc_dai_driver *driver;
 
 	/* DAI runtime info */
-	unsigned int capture_active:1;		/* stream is in use */
-	unsigned int playback_active:1;		/* stream is in use */
+	unsigned int capture_active;		/* stream usage count */
+	unsigned int playback_active;		/* stream usage count */
 	unsigned int probed:1;
 
 	unsigned int active;
@@ -315,7 +315,6 @@ struct snd_soc_dai {
 	unsigned int sample_bits;
 
 	/* parent platform/codec */
-	struct snd_soc_codec *codec;
 	struct snd_soc_component *component;
 
 	/* CODEC TDM slot masks and params (for fixup) */

include/trace/events/asoc.h

@@ -12,7 +12,6 @@
 #define DAPM_ARROW(dir) (((dir) == SND_SOC_DAPM_DIR_OUT) ? "->" : "<-")
 
 struct snd_soc_jack;
-struct snd_soc_codec;
 struct snd_soc_card;
 struct snd_soc_dapm_widget;
 struct snd_soc_dapm_path;

include/uapi/linux/usb/audio.h

@@ -285,9 +285,22 @@ static inline __u8 uac_mixer_unit_iChannelNames(struct uac_mixer_unit_descriptor
 static inline __u8 *uac_mixer_unit_bmControls(struct uac_mixer_unit_descriptor *desc,
 					      int protocol)
 {
-	return (protocol == UAC_VERSION_1) ?
-		&desc->baSourceID[desc->bNrInPins + 4] :
-		&desc->baSourceID[desc->bNrInPins + 6];
+	switch (protocol) {
+	case UAC_VERSION_1:
+		return &desc->baSourceID[desc->bNrInPins + 4];
+	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 __u16 uac3_mixer_unit_wClusterDescrID(struct uac_mixer_unit_descriptor *desc)
+{
+	return (desc->baSourceID[desc->bNrInPins + 1] << 8) |
+		desc->baSourceID[desc->bNrInPins];
 }
 
 static inline __u8 uac_mixer_unit_iMixer(struct uac_mixer_unit_descriptor *desc)

include/uapi/sound/asoc.h

@@ -139,6 +139,15 @@
 #define SND_SOC_TPLG_DAI_FLGBIT_SYMMETRIC_CHANNELS      (1 << 1)
 #define SND_SOC_TPLG_DAI_FLGBIT_SYMMETRIC_SAMPLEBITS    (1 << 2)
 
+/* DAI clock gating */
+#define SND_SOC_TPLG_DAI_CLK_GATE_UNDEFINED	0
+#define SND_SOC_TPLG_DAI_CLK_GATE_GATED	1
+#define SND_SOC_TPLG_DAI_CLK_GATE_CONT		2
+
+/* DAI mclk_direction */
+#define SND_SOC_TPLG_MCLK_CO            0 /* for codec, mclk is output */
+#define SND_SOC_TPLG_MCLK_CI            1 /* for codec, mclk is input */
+
 /* DAI physical PCM data formats.
  * Add new formats to the end of the list.
  */
@@ -160,6 +169,18 @@
 #define SND_SOC_TPLG_LNK_FLGBIT_SYMMETRIC_SAMPLEBITS    (1 << 2)
 #define SND_SOC_TPLG_LNK_FLGBIT_VOICE_WAKEUP            (1 << 3)
 
+/* DAI topology BCLK parameter
+ * For the backwards capability, by default codec is bclk master
+ */
+#define SND_SOC_TPLG_BCLK_CM         0 /* codec is bclk master */
+#define SND_SOC_TPLG_BCLK_CS         1 /* codec is bclk slave */
+
+/* DAI topology FSYNC parameter
+ * For the backwards capability, by default codec is fsync master
+ */
+#define SND_SOC_TPLG_FSYNC_CM         0 /* codec is fsync master */
+#define SND_SOC_TPLG_FSYNC_CS         1 /* codec is fsync slave */
+
 /*
  * Block Header.
  * This header precedes all object and object arrays below.
@@ -312,12 +333,12 @@ struct snd_soc_tplg_hw_config {
 	__le32 size;            /* in bytes of this structure */
 	__le32 id;		/* unique ID - - used to match */
 	__le32 fmt;		/* SND_SOC_DAI_FORMAT_ format value */
-	__u8 clock_gated;	/* 1 if clock can be gated to save power */
+	__u8 clock_gated;	/* SND_SOC_TPLG_DAI_CLK_GATE_ value */
 	__u8 invert_bclk;	/* 1 for inverted BCLK, 0 for normal */
 	__u8 invert_fsync;	/* 1 for inverted frame clock, 0 for normal */
-	__u8 bclk_master;	/* 1 for master of BCLK, 0 for slave */
-	__u8 fsync_master;	/* 1 for master of FSYNC, 0 for slave */
-	__u8 mclk_direction;    /* 0 for input, 1 for output */
+	__u8 bclk_master;	/* SND_SOC_TPLG_BCLK_ value */
+	__u8 fsync_master;	/* SND_SOC_TPLG_FSYNC_ value */
+	__u8 mclk_direction;    /* SND_SOC_TPLG_MCLK_ value */
 	__le16 reserved;	/* for 32bit alignment */
 	__le32 mclk_rate;	/* MCLK or SYSCLK freqency in Hz */
 	__le32 bclk_rate;	/* BCLK freqency in Hz */
@@ -552,4 +573,61 @@ struct snd_soc_tplg_dai {
 	__le32 flags;           /* SND_SOC_TPLG_DAI_FLGBIT_* */
 	struct snd_soc_tplg_private priv;
 } __attribute__((packed));
+
+/*
+ * Old version of ABI structs, supported for backward compatibility.
+ */
+
+/* Manifest v4 */
+struct snd_soc_tplg_manifest_v4 {
+	__le32 size;		/* in bytes of this structure */
+	__le32 control_elems;	/* number of control elements */
+	__le32 widget_elems;	/* number of widget elements */
+	__le32 graph_elems;	/* number of graph elements */
+	__le32 pcm_elems;	/* number of PCM elements */
+	__le32 dai_link_elems;	/* number of DAI link elements */
+	struct snd_soc_tplg_private priv;
+} __packed;
+
+/* Stream Capabilities v4 */
+struct snd_soc_tplg_stream_caps_v4 {
+	__le32 size;		/* in bytes of this structure */
+	char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
+	__le64 formats;	/* supported formats SNDRV_PCM_FMTBIT_* */
+	__le32 rates;		/* supported rates SNDRV_PCM_RATE_* */
+	__le32 rate_min;	/* min rate */
+	__le32 rate_max;	/* max rate */
+	__le32 channels_min;	/* min channels */
+	__le32 channels_max;	/* max channels */
+	__le32 periods_min;	/* min number of periods */
+	__le32 periods_max;	/* max number of periods */
+	__le32 period_size_min;	/* min period size bytes */
+	__le32 period_size_max;	/* max period size bytes */
+	__le32 buffer_size_min;	/* min buffer size bytes */
+	__le32 buffer_size_max;	/* max buffer size bytes */
+} __packed;
+
+/* PCM v4 */
+struct snd_soc_tplg_pcm_v4 {
+	__le32 size;		/* in bytes of this structure */
+	char pcm_name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
+	char dai_name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
+	__le32 pcm_id;		/* unique ID - used to match with DAI link */
+	__le32 dai_id;		/* unique ID - used to match */
+	__le32 playback;	/* supports playback mode */
+	__le32 capture;		/* supports capture mode */
+	__le32 compress;	/* 1 = compressed; 0 = PCM */
+	struct snd_soc_tplg_stream stream[SND_SOC_TPLG_STREAM_CONFIG_MAX]; /* for DAI link */
+	__le32 num_streams;	/* number of streams */
+	struct snd_soc_tplg_stream_caps_v4 caps[2]; /* playback and capture for DAI */
+} __packed;
+
+/* Physical link config v4 */
+struct snd_soc_tplg_link_config_v4 {
+	__le32 size;            /* in bytes of this structure */
+	__le32 id;              /* unique ID - used to match */
+	struct snd_soc_tplg_stream stream[SND_SOC_TPLG_STREAM_CONFIG_MAX]; /* supported configs playback and captrure */
+	__le32 num_streams;     /* number of streams */
+} __packed;
+
 #endif

sound/soc/intel/skylake/skl-tplg-interface.h → include/uapi/sound/skl-tplg-interface.h

+/* SPDX-License-Identifier: GPL-2.0 */
 /*
  * skl-tplg-interface.h - Intel DSP FW private data interface
  *
  * Copyright (C) 2015 Intel Corp
  * Author: Jeeja KP <jeeja.kp@intel.com>
  *	    Nilofer, Samreen <samreen.nilofer@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 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.
  */
 
 #ifndef __HDA_TPLG_INTERFACE_H__
@@ -169,4 +160,78 @@ enum skl_tuple_type {
 	SKL_TYPE_DATA
 };
 
+/* v4 configuration data */
+
+struct skl_dfw_v4_module_pin {
+	u16 module_id;
+	u16 instance_id;
+} __packed;
+
+struct skl_dfw_v4_module_fmt {
+	u32 channels;
+	u32 freq;
+	u32 bit_depth;
+	u32 valid_bit_depth;
+	u32 ch_cfg;
+	u32 interleaving_style;
+	u32 sample_type;
+	u32 ch_map;
+} __packed;
+
+struct skl_dfw_v4_module_caps {
+	u32 set_params:2;
+	u32 rsvd:30;
+	u32 param_id;
+	u32 caps_size;
+	u32 caps[HDA_SST_CFG_MAX];
+} __packed;
+
+struct skl_dfw_v4_pipe {
+	u8 pipe_id;
+	u8 pipe_priority;
+	u16 conn_type:4;
+	u16 rsvd:4;
+	u16 memory_pages:8;
+} __packed;
+
+struct skl_dfw_v4_module {
+	char uuid[SKL_UUID_STR_SZ];
+
+	u16 module_id;
+	u16 instance_id;
+	u32 max_mcps;
+	u32 mem_pages;
+	u32 obs;
+	u32 ibs;
+	u32 vbus_id;
+
+	u32 max_in_queue:8;
+	u32 max_out_queue:8;
+	u32 time_slot:8;
+	u32 core_id:4;
+	u32 rsvd1:4;
+
+	u32 module_type:8;
+	u32 conn_type:4;
+	u32 dev_type:4;
+	u32 hw_conn_type:4;
+	u32 rsvd2:12;
+
+	u32 params_fixup:8;
+	u32 converter:8;
+	u32 input_pin_type:1;
+	u32 output_pin_type:1;
+	u32 is_dynamic_in_pin:1;
+	u32 is_dynamic_out_pin:1;
+	u32 is_loadable:1;
+	u32 rsvd3:11;
+
+	struct skl_dfw_v4_pipe pipe;
+	struct skl_dfw_v4_module_fmt in_fmt[MAX_IN_QUEUE];
+	struct skl_dfw_v4_module_fmt out_fmt[MAX_OUT_QUEUE];
+	struct skl_dfw_v4_module_pin in_pin[MAX_IN_QUEUE];
+	struct skl_dfw_v4_module_pin out_pin[MAX_OUT_QUEUE];
+	struct skl_dfw_v4_module_caps caps;
+} __packed;
+
 #endif

include/uapi/sound/tlv.h

@@ -42,6 +42,10 @@
 #define SNDRV_CTL_TLVD_LENGTH(...) \
 	((unsigned int)sizeof((const unsigned int[]) { __VA_ARGS__ }))
 
+/* Accessor offsets for TLV data items */
+#define SNDRV_CTL_TLVO_TYPE		0
+#define SNDRV_CTL_TLVO_LEN		1
+
 #define SNDRV_CTL_TLVD_CONTAINER_ITEM(...) \
 	SNDRV_CTL_TLVD_ITEM(SNDRV_CTL_TLVT_CONTAINER, __VA_ARGS__)
 #define SNDRV_CTL_TLVD_DECLARE_CONTAINER(name, ...) \
@@ -61,6 +65,10 @@
 		SNDRV_CTL_TLVD_DB_SCALE_ITEM(min, step, mute) \
 	}
 
+/* Accessor offsets for min, mute and step items in dB scale type TLV */
+#define SNDRV_CTL_TLVO_DB_SCALE_MIN		2
+#define SNDRV_CTL_TLVO_DB_SCALE_MUTE_AND_STEP	3
+
 /* dB scale specified with min/max values instead of step */
 #define SNDRV_CTL_TLVD_DB_MINMAX_ITEM(min_dB, max_dB) \
 	SNDRV_CTL_TLVD_ITEM(SNDRV_CTL_TLVT_DB_MINMAX, (min_dB), (max_dB))
@@ -75,6 +83,10 @@
 		SNDRV_CTL_TLVD_DB_MINMAX_MUTE_ITEM(min_dB, max_dB) \
 	}
 
+/* Accessor offsets for min, max items in db-minmax types of TLV. */
+#define SNDRV_CTL_TLVO_DB_MINMAX_MIN	2
+#define SNDRV_CTL_TLVO_DB_MINMAX_MAX	3
+
 /* linear volume between min_dB and max_dB (.01dB unit) */
 #define SNDRV_CTL_TLVD_DB_LINEAR_ITEM(min_dB, max_dB) \
 	SNDRV_CTL_TLVD_ITEM(SNDRV_CTL_TLVT_DB_LINEAR, (min_dB), (max_dB))
@@ -83,6 +95,10 @@
 		SNDRV_CTL_TLVD_DB_LINEAR_ITEM(min_dB, max_dB) \
 	}
 
+/* Accessor offsets for min, max items in db-linear type of TLV. */
+#define SNDRV_CTL_TLVO_DB_LINEAR_MIN	2
+#define SNDRV_CTL_TLVO_DB_LINEAR_MAX	3
+
 /* dB range container:
  * Items in dB range container must be ordered by their values and by their
  * dB values. This implies that larger values must correspond with larger

sound/Kconfig

@@ -96,6 +96,8 @@ source "sound/x86/Kconfig"
 
 source "sound/synth/Kconfig"
 
+source "sound/xen/Kconfig"
+
 endif # SND
 
 endif # !UML

sound/Makefile

@@ -5,7 +5,7 @@
 obj-$(CONFIG_SOUND) += soundcore.o
 obj-$(CONFIG_DMASOUND) += oss/dmasound/
 obj-$(CONFIG_SND) += core/ i2c/ drivers/ isa/ pci/ ppc/ arm/ sh/ synth/ usb/ \
-	firewire/ sparc/ spi/ parisc/ pcmcia/ mips/ soc/ atmel/ hda/ x86/
+	firewire/ sparc/ spi/ parisc/ pcmcia/ mips/ soc/ atmel/ hda/ x86/ xen/
 obj-$(CONFIG_SND_AOA) += aoa/
 
 # This one must be compilable even if sound is configured out

sound/core/compress_offload.c

@@ -1001,7 +1001,7 @@ static int snd_compress_proc_init(struct snd_compr *compr)
 					   compr->card->proc_root);
 	if (!entry)
 		return -ENOMEM;
-	entry->mode = S_IFDIR | S_IRUGO | S_IXUGO;
+	entry->mode = S_IFDIR | 0555;
 	if (snd_info_register(entry) < 0) {
 		snd_info_free_entry(entry);
 		return -ENOMEM;

sound/core/device.c

@@ -240,6 +240,15 @@ void snd_device_free_all(struct snd_card *card)
 
 	if (snd_BUG_ON(!card))
 		return;
+	list_for_each_entry_safe_reverse(dev, next, &card->devices, list) {
+		/* exception: free ctl and lowlevel stuff later */
+		if (dev->type == SNDRV_DEV_CONTROL ||
+		    dev->type == SNDRV_DEV_LOWLEVEL)
+			continue;
+		__snd_device_free(dev);
+	}
+
+	/* free all */
 	list_for_each_entry_safe_reverse(dev, next, &card->devices, list)
 		__snd_device_free(dev);
 }

sound/core/info.c

@@ -454,7 +454,7 @@ static struct snd_info_entry *create_subdir(struct module *mod,
 	entry = snd_info_create_module_entry(mod, name, NULL);
 	if (!entry)
 		return NULL;
-	entry->mode = S_IFDIR | S_IRUGO | S_IXUGO;
+	entry->mode = S_IFDIR | 0555;
 	if (snd_info_register(entry) < 0) {
 		snd_info_free_entry(entry);
 		return NULL;
@@ -470,7 +470,7 @@ int __init snd_info_init(void)
 	snd_proc_root = snd_info_create_entry("asound", NULL);
 	if (!snd_proc_root)
 		return -ENOMEM;
-	snd_proc_root->mode = S_IFDIR | S_IRUGO | S_IXUGO;
+	snd_proc_root->mode = S_IFDIR | 0555;
 	snd_proc_root->p = proc_mkdir("asound", NULL);
 	if (!snd_proc_root->p)
 		goto error;
@@ -716,7 +716,7 @@ snd_info_create_entry(const char *name, struct snd_info_entry *parent)
 		kfree(entry);
 		return NULL;
 	}
-	entry->mode = S_IFREG | S_IRUGO;
+	entry->mode = S_IFREG | 0444;
 	entry->content = SNDRV_INFO_CONTENT_TEXT;
 	mutex_init(&entry->access);
 	INIT_LIST_HEAD(&entry->children);

sound/core/init.c

@@ -703,7 +703,7 @@ card_id_store_attr(struct device *dev, struct device_attribute *attr,
 	return count;
 }
 
-static DEVICE_ATTR(id, S_IRUGO | S_IWUSR, card_id_show_attr, card_id_store_attr);
+static DEVICE_ATTR(id, 0644, card_id_show_attr, card_id_store_attr);
 
 static ssize_t
 card_number_show_attr(struct device *dev,
@@ -713,7 +713,7 @@ card_number_show_attr(struct device *dev,
 	return scnprintf(buf, PAGE_SIZE, "%i\n", card->number);
 }
 
-static DEVICE_ATTR(number, S_IRUGO, card_number_show_attr, NULL);
+static DEVICE_ATTR(number, 0444, card_number_show_attr, NULL);
 
 static struct attribute *card_dev_attrs[] = {
 	&dev_attr_id.attr,

sound/core/oss/mixer_oss.c

@@ -1247,7 +1247,7 @@ static void snd_mixer_oss_proc_init(struct snd_mixer_oss *mixer)
 	if (! entry)
 		return;
 	entry->content = SNDRV_INFO_CONTENT_TEXT;
-	entry->mode = S_IFREG | S_IRUGO | S_IWUSR;
+	entry->mode = S_IFREG | 0644;
 	entry->c.text.read = snd_mixer_oss_proc_read;
 	entry->c.text.write = snd_mixer_oss_proc_write;
 	entry->private_data = mixer;

sound/core/oss/pcm_oss.c

@@ -3045,7 +3045,7 @@ static void snd_pcm_oss_proc_init(struct snd_pcm *pcm)
 			continue;
 		if ((entry = snd_info_create_card_entry(pcm->card, "oss", pstr->proc_root)) != NULL) {
 			entry->content = SNDRV_INFO_CONTENT_TEXT;
-			entry->mode = S_IFREG | S_IRUGO | S_IWUSR;
+			entry->mode = S_IFREG | 0644;
 			entry->c.text.read = snd_pcm_oss_proc_read;
 			entry->c.text.write = snd_pcm_oss_proc_write;
 			entry->private_data = pstr;

sound/core/pcm.c

@@ -530,7 +530,7 @@ static int snd_pcm_stream_proc_init(struct snd_pcm_str *pstr)
 					   pcm->card->proc_root);
 	if (!entry)
 		return -ENOMEM;
-	entry->mode = S_IFDIR | S_IRUGO | S_IXUGO;
+	entry->mode = S_IFDIR | 0555;
 	if (snd_info_register(entry) < 0) {
 		snd_info_free_entry(entry);
 		return -ENOMEM;
@@ -552,7 +552,7 @@ static int snd_pcm_stream_proc_init(struct snd_pcm_str *pstr)
 	if (entry) {
 		entry->c.text.read = snd_pcm_xrun_debug_read;
 		entry->c.text.write = snd_pcm_xrun_debug_write;
-		entry->mode |= S_IWUSR;
+		entry->mode |= 0200;
 		entry->private_data = pstr;
 		if (snd_info_register(entry) < 0) {
 			snd_info_free_entry(entry);
@@ -590,7 +590,7 @@ static int snd_pcm_substream_proc_init(struct snd_pcm_substream *substream)
 					   substream->pstr->proc_root);
 	if (!entry)
 		return -ENOMEM;
-	entry->mode = S_IFDIR | S_IRUGO | S_IXUGO;
+	entry->mode = S_IFDIR | 0555;
 	if (snd_info_register(entry) < 0) {
 		snd_info_free_entry(entry);
 		return -ENOMEM;
@@ -647,7 +647,7 @@ static int snd_pcm_substream_proc_init(struct snd_pcm_substream *substream)
 		entry->private_data = substream;
 		entry->c.text.read = NULL;
 		entry->c.text.write = snd_pcm_xrun_injection_write;
-		entry->mode = S_IFREG | S_IWUSR;
+		entry->mode = S_IFREG | 0200;
 		if (snd_info_register(entry) < 0) {
 			snd_info_free_entry(entry);
 			entry = NULL;
@@ -1087,7 +1087,7 @@ static ssize_t show_pcm_class(struct device *dev,
         return snprintf(buf, PAGE_SIZE, "%s\n", str);
 }
 
-static DEVICE_ATTR(pcm_class, S_IRUGO, show_pcm_class, NULL);
+static DEVICE_ATTR(pcm_class, 0444, show_pcm_class, NULL);
 static struct attribute *pcm_dev_attrs[] = {
 	&dev_attr_pcm_class.attr,
 	NULL

sound/core/pcm_compat.c

@@ -45,10 +45,7 @@ static int snd_pcm_ioctl_rewind_compat(struct snd_pcm_substream *substream,
 
 	if (get_user(frames, src))
 		return -EFAULT;
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		err = snd_pcm_playback_rewind(substream, frames);
-	else
-		err = snd_pcm_capture_rewind(substream, frames);
+	err = snd_pcm_rewind(substream, frames);
 	if (put_user(err, src))
 		return -EFAULT;
 	return err < 0 ? err : 0;
@@ -62,10 +59,7 @@ static int snd_pcm_ioctl_forward_compat(struct snd_pcm_substream *substream,
 
 	if (get_user(frames, src))
 		return -EFAULT;
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		err = snd_pcm_playback_forward(substream, frames);
-	else
-		err = snd_pcm_capture_forward(substream, frames);
+	err = snd_pcm_forward(substream, frames);
 	if (put_user(err, src))
 		return -EFAULT;
 	return err < 0 ? err : 0;

sound/core/pcm_lib.c

@@ -191,10 +191,7 @@ int snd_pcm_update_state(struct snd_pcm_substream *substream,
 {
 	snd_pcm_uframes_t avail;
 
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		avail = snd_pcm_playback_avail(runtime);
-	else
-		avail = snd_pcm_capture_avail(runtime);
+	avail = snd_pcm_avail(substream);
 	if (avail > runtime->avail_max)
 		runtime->avail_max = avail;
 	if (runtime->status->state == SNDRV_PCM_STATE_DRAINING) {
@@ -1856,10 +1853,7 @@ static int wait_for_avail(struct snd_pcm_substream *substream,
 		 * This check must happen after been added to the waitqueue
 		 * and having current state be INTERRUPTIBLE.
 		 */
-		if (is_playback)
-			avail = snd_pcm_playback_avail(runtime);
-		else
-			avail = snd_pcm_capture_avail(runtime);
+		avail = snd_pcm_avail(substream);
 		if (avail >= runtime->twake)
 			break;
 		snd_pcm_stream_unlock_irq(substream);
@@ -2175,10 +2169,7 @@ snd_pcm_sframes_t __snd_pcm_lib_xfer(struct snd_pcm_substream *substream,
 	runtime->twake = runtime->control->avail_min ? : 1;
 	if (runtime->status->state == SNDRV_PCM_STATE_RUNNING)
 		snd_pcm_update_hw_ptr(substream);
-	if (is_playback)
-		avail = snd_pcm_playback_avail(runtime);
-	else
-		avail = snd_pcm_capture_avail(runtime);
+	avail = snd_pcm_avail(substream);
 	while (size > 0) {
 		snd_pcm_uframes_t frames, appl_ptr, appl_ofs;
 		snd_pcm_uframes_t cont;

sound/core/pcm_local.h

@@ -36,6 +36,24 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream);
 void snd_pcm_playback_silence(struct snd_pcm_substream *substream,
 			      snd_pcm_uframes_t new_hw_ptr);
 
+static inline snd_pcm_uframes_t
+snd_pcm_avail(struct snd_pcm_substream *substream)
+{
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		return snd_pcm_playback_avail(substream->runtime);
+	else
+		return snd_pcm_capture_avail(substream->runtime);
+}
+
+static inline snd_pcm_uframes_t
+snd_pcm_hw_avail(struct snd_pcm_substream *substream)
+{
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		return snd_pcm_playback_hw_avail(substream->runtime);
+	else
+		return snd_pcm_capture_hw_avail(substream->runtime);
+}
+
 #ifdef CONFIG_SND_PCM_TIMER
 void snd_pcm_timer_resolution_change(struct snd_pcm_substream *substream);
 void snd_pcm_timer_init(struct snd_pcm_substream *substream);

sound/core/pcm_memory.c

@@ -201,7 +201,7 @@ static inline void preallocate_info_init(struct snd_pcm_substream *substream)
 	if ((entry = snd_info_create_card_entry(substream->pcm->card, "prealloc", substream->proc_root)) != NULL) {
 		entry->c.text.read = snd_pcm_lib_preallocate_proc_read;
 		entry->c.text.write = snd_pcm_lib_preallocate_proc_write;
-		entry->mode |= S_IWUSR;
+		entry->mode |= 0200;
 		entry->private_data = substream;
 		if (snd_info_register(entry) < 0) {
 			snd_info_free_entry(entry);

sound/core/pcm_native.c

@@ -99,6 +99,57 @@ static inline void down_write_nonblock(struct rw_semaphore *lock)
 		cond_resched();
 }
 
+#define PCM_LOCK_DEFAULT	0
+#define PCM_LOCK_IRQ	1
+#define PCM_LOCK_IRQSAVE	2
+
+static unsigned long __snd_pcm_stream_lock_mode(struct snd_pcm_substream *substream,
+						unsigned int mode)
+{
+	unsigned long flags = 0;
+	if (substream->pcm->nonatomic) {
+		down_read_nested(&snd_pcm_link_rwsem, SINGLE_DEPTH_NESTING);
+		mutex_lock(&substream->self_group.mutex);
+	} else {
+		switch (mode) {
+		case PCM_LOCK_DEFAULT:
+			read_lock(&snd_pcm_link_rwlock);
+			break;
+		case PCM_LOCK_IRQ:
+			read_lock_irq(&snd_pcm_link_rwlock);
+			break;
+		case PCM_LOCK_IRQSAVE:
+			read_lock_irqsave(&snd_pcm_link_rwlock, flags);
+			break;
+		}
+		spin_lock(&substream->self_group.lock);
+	}
+	return flags;
+}
+
+static void __snd_pcm_stream_unlock_mode(struct snd_pcm_substream *substream,
+					 unsigned int mode, unsigned long flags)
+{
+	if (substream->pcm->nonatomic) {
+		mutex_unlock(&substream->self_group.mutex);
+		up_read(&snd_pcm_link_rwsem);
+	} else {
+		spin_unlock(&substream->self_group.lock);
+
+		switch (mode) {
+		case PCM_LOCK_DEFAULT:
+			read_unlock(&snd_pcm_link_rwlock);
+			break;
+		case PCM_LOCK_IRQ:
+			read_unlock_irq(&snd_pcm_link_rwlock);
+			break;
+		case PCM_LOCK_IRQSAVE:
+			read_unlock_irqrestore(&snd_pcm_link_rwlock, flags);
+			break;
+		}
+	}
+}
+
 /**
  * snd_pcm_stream_lock - Lock the PCM stream
  * @substream: PCM substream
@@ -109,13 +160,7 @@ static inline void down_write_nonblock(struct rw_semaphore *lock)
  */
 void snd_pcm_stream_lock(struct snd_pcm_substream *substream)
 {
-	if (substream->pcm->nonatomic) {
-		down_read_nested(&snd_pcm_link_rwsem, SINGLE_DEPTH_NESTING);
-		mutex_lock(&substream->self_group.mutex);
-	} else {
-		read_lock(&snd_pcm_link_rwlock);
-		spin_lock(&substream->self_group.lock);
-	}
+	__snd_pcm_stream_lock_mode(substream, PCM_LOCK_DEFAULT);
 }
 EXPORT_SYMBOL_GPL(snd_pcm_stream_lock);
 
@@ -127,13 +172,7 @@ EXPORT_SYMBOL_GPL(snd_pcm_stream_lock);
  */
 void snd_pcm_stream_unlock(struct snd_pcm_substream *substream)
 {
-	if (substream->pcm->nonatomic) {
-		mutex_unlock(&substream->self_group.mutex);
-		up_read(&snd_pcm_link_rwsem);
-	} else {
-		spin_unlock(&substream->self_group.lock);
-		read_unlock(&snd_pcm_link_rwlock);
-	}
+	__snd_pcm_stream_unlock_mode(substream, PCM_LOCK_DEFAULT, 0);
 }
 EXPORT_SYMBOL_GPL(snd_pcm_stream_unlock);
 
@@ -147,9 +186,7 @@ EXPORT_SYMBOL_GPL(snd_pcm_stream_unlock);
  */
 void snd_pcm_stream_lock_irq(struct snd_pcm_substream *substream)
 {
-	if (!substream->pcm->nonatomic)
-		local_irq_disable();
-	snd_pcm_stream_lock(substream);
+	__snd_pcm_stream_lock_mode(substream, PCM_LOCK_IRQ);
 }
 EXPORT_SYMBOL_GPL(snd_pcm_stream_lock_irq);
 
@@ -161,19 +198,13 @@ EXPORT_SYMBOL_GPL(snd_pcm_stream_lock_irq);
  */
 void snd_pcm_stream_unlock_irq(struct snd_pcm_substream *substream)
 {
-	snd_pcm_stream_unlock(substream);
-	if (!substream->pcm->nonatomic)
-		local_irq_enable();
+	__snd_pcm_stream_unlock_mode(substream, PCM_LOCK_IRQ, 0);
 }
 EXPORT_SYMBOL_GPL(snd_pcm_stream_unlock_irq);
 
 unsigned long _snd_pcm_stream_lock_irqsave(struct snd_pcm_substream *substream)
 {
-	unsigned long flags = 0;
-	if (!substream->pcm->nonatomic)
-		local_irq_save(flags);
-	snd_pcm_stream_lock(substream);
-	return flags;
+	return __snd_pcm_stream_lock_mode(substream, PCM_LOCK_IRQSAVE);
 }
 EXPORT_SYMBOL_GPL(_snd_pcm_stream_lock_irqsave);
 
@@ -187,9 +218,7 @@ EXPORT_SYMBOL_GPL(_snd_pcm_stream_lock_irqsave);
 void snd_pcm_stream_unlock_irqrestore(struct snd_pcm_substream *substream,
 				      unsigned long flags)
 {
-	snd_pcm_stream_unlock(substream);
-	if (!substream->pcm->nonatomic)
-		local_irq_restore(flags);
+	__snd_pcm_stream_unlock_mode(substream, PCM_LOCK_IRQSAVE, flags);
 }
 EXPORT_SYMBOL_GPL(snd_pcm_stream_unlock_irqrestore);
 
@@ -857,6 +886,18 @@ static int snd_pcm_sw_params_user(struct snd_pcm_substream *substream,
 	return err;
 }
 
+static inline snd_pcm_uframes_t
+snd_pcm_calc_delay(struct snd_pcm_substream *substream)
+{
+	snd_pcm_uframes_t delay;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		delay = snd_pcm_playback_hw_avail(substream->runtime);
+	else
+		delay = snd_pcm_capture_avail(substream->runtime);
+	return delay + substream->runtime->delay;
+}
+
 int snd_pcm_status(struct snd_pcm_substream *substream,
 		   struct snd_pcm_status *status)
 {
@@ -908,21 +949,9 @@ int snd_pcm_status(struct snd_pcm_substream *substream,
  _tstamp_end:
 	status->appl_ptr = runtime->control->appl_ptr;
 	status->hw_ptr = runtime->status->hw_ptr;
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		status->avail = snd_pcm_playback_avail(runtime);
-		if (runtime->status->state == SNDRV_PCM_STATE_RUNNING ||
-		    runtime->status->state == SNDRV_PCM_STATE_DRAINING) {
-			status->delay = runtime->buffer_size - status->avail;
-			status->delay += runtime->delay;
-		} else
-			status->delay = 0;
-	} else {
-		status->avail = snd_pcm_capture_avail(runtime);
-		if (runtime->status->state == SNDRV_PCM_STATE_RUNNING)
-			status->delay = status->avail + runtime->delay;
-		else
-			status->delay = 0;
-	}
+	status->avail = snd_pcm_avail(substream);
+	status->delay = snd_pcm_running(substream) ?
+		snd_pcm_calc_delay(substream) : 0;
 	status->avail_max = runtime->avail_max;
 	status->overrange = runtime->overrange;
 	runtime->avail_max = 0;
@@ -2610,10 +2639,9 @@ static snd_pcm_sframes_t rewind_appl_ptr(struct snd_pcm_substream *substream,
 	return ret < 0 ? 0 : frames;
 }
 
-static snd_pcm_sframes_t snd_pcm_playback_rewind(struct snd_pcm_substream *substream,
-						 snd_pcm_uframes_t frames)
+static snd_pcm_sframes_t snd_pcm_rewind(struct snd_pcm_substream *substream,
+					snd_pcm_uframes_t frames)
 {
-	struct snd_pcm_runtime *runtime = substream->runtime;
 	snd_pcm_sframes_t ret;
 
 	if (frames == 0)
@@ -2623,33 +2651,14 @@ static snd_pcm_sframes_t snd_pcm_playback_rewind(struct snd_pcm_substream *subst
 	ret = do_pcm_hwsync(substream);
 	if (!ret)
 		ret = rewind_appl_ptr(substream, frames,
-				      snd_pcm_playback_hw_avail(runtime));
+				      snd_pcm_hw_avail(substream));
 	snd_pcm_stream_unlock_irq(substream);
 	return ret;
 }
 
-static snd_pcm_sframes_t snd_pcm_capture_rewind(struct snd_pcm_substream *substream,
-						snd_pcm_uframes_t frames)
+static snd_pcm_sframes_t snd_pcm_forward(struct snd_pcm_substream *substream,
+					 snd_pcm_uframes_t frames)
 {
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	snd_pcm_sframes_t ret;
-
-	if (frames == 0)
-		return 0;
-
-	snd_pcm_stream_lock_irq(substream);
-	ret = do_pcm_hwsync(substream);
-	if (!ret)
-		ret = rewind_appl_ptr(substream, frames,
-				      snd_pcm_capture_hw_avail(runtime));
-	snd_pcm_stream_unlock_irq(substream);
-	return ret;
-}
-
-static snd_pcm_sframes_t snd_pcm_playback_forward(struct snd_pcm_substream *substream,
-						  snd_pcm_uframes_t frames)
-{
-	struct snd_pcm_runtime *runtime = substream->runtime;
 	snd_pcm_sframes_t ret;
 
 	if (frames == 0)
@@ -2659,25 +2668,7 @@ static snd_pcm_sframes_t snd_pcm_playback_forward(struct snd_pcm_substream *subs
 	ret = do_pcm_hwsync(substream);
 	if (!ret)
 		ret = forward_appl_ptr(substream, frames,
-				       snd_pcm_playback_avail(runtime));
-	snd_pcm_stream_unlock_irq(substream);
-	return ret;
-}
-
-static snd_pcm_sframes_t snd_pcm_capture_forward(struct snd_pcm_substream *substream,
-						 snd_pcm_uframes_t frames)
-{
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	snd_pcm_sframes_t ret;
-
-	if (frames == 0)
-		return 0;
-
-	snd_pcm_stream_lock_irq(substream);
-	ret = do_pcm_hwsync(substream);
-	if (!ret)
-		ret = forward_appl_ptr(substream, frames,
-				       snd_pcm_capture_avail(runtime));
+				       snd_pcm_avail(substream));
 	snd_pcm_stream_unlock_irq(substream);
 	return ret;
 }
@@ -2695,19 +2686,13 @@ static int snd_pcm_hwsync(struct snd_pcm_substream *substream)
 static int snd_pcm_delay(struct snd_pcm_substream *substream,
 			 snd_pcm_sframes_t *delay)
 {
-	struct snd_pcm_runtime *runtime = substream->runtime;
 	int err;
 	snd_pcm_sframes_t n = 0;
 
 	snd_pcm_stream_lock_irq(substream);
 	err = do_pcm_hwsync(substream);
-	if (!err) {
-		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-			n = snd_pcm_playback_hw_avail(runtime);
-		else
-			n = snd_pcm_capture_avail(runtime);
-		n += runtime->delay;
-	}
+	if (!err)
+		n = snd_pcm_calc_delay(substream);
 	snd_pcm_stream_unlock_irq(substream);
 	if (!err)
 		*delay = n;
@@ -2834,10 +2819,7 @@ static int snd_pcm_rewind_ioctl(struct snd_pcm_substream *substream,
 		return -EFAULT;
 	if (put_user(0, _frames))
 		return -EFAULT;
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		result = snd_pcm_playback_rewind(substream, frames);
-	else
-		result = snd_pcm_capture_rewind(substream, frames);
+	result = snd_pcm_rewind(substream, frames);
 	__put_user(result, _frames);
 	return result < 0 ? result : 0;
 }
@@ -2852,10 +2834,7 @@ static int snd_pcm_forward_ioctl(struct snd_pcm_substream *substream,
 		return -EFAULT;
 	if (put_user(0, _frames))
 		return -EFAULT;
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		result = snd_pcm_playback_forward(substream, frames);
-	else
-		result = snd_pcm_capture_forward(substream, frames);
+	result = snd_pcm_forward(substream, frames);
 	__put_user(result, _frames);
 	return result < 0 ? result : 0;
 }
@@ -2998,7 +2977,7 @@ int snd_pcm_kernel_ioctl(struct snd_pcm_substream *substream,
 		/* provided only for OSS; capture-only and no value returned */
 		if (substream->stream != SNDRV_PCM_STREAM_CAPTURE)
 			return -EINVAL;
-		result = snd_pcm_capture_forward(substream, *frames);
+		result = snd_pcm_forward(substream, *frames);
 		return result < 0 ? result : 0;
 	}
 	case SNDRV_PCM_IOCTL_HW_PARAMS:
@@ -3140,82 +3119,46 @@ static ssize_t snd_pcm_writev(struct kiocb *iocb, struct iov_iter *from)
 	return result;
 }
 
-static __poll_t snd_pcm_playback_poll(struct file *file, poll_table * wait)
+static __poll_t snd_pcm_poll(struct file *file, poll_table *wait)
 {
 	struct snd_pcm_file *pcm_file;
 	struct snd_pcm_substream *substream;
 	struct snd_pcm_runtime *runtime;
-        __poll_t mask;
+	__poll_t mask, ok;
 	snd_pcm_uframes_t avail;
 
 	pcm_file = file->private_data;
 
 	substream = pcm_file->substream;
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		ok = EPOLLOUT | EPOLLWRNORM;
+	else
+		ok = EPOLLIN | EPOLLRDNORM;
 	if (PCM_RUNTIME_CHECK(substream))
-		return EPOLLOUT | EPOLLWRNORM | EPOLLERR;
-	runtime = substream->runtime;
-
-	poll_wait(file, &runtime->sleep, wait);
+		return ok | EPOLLERR;
 
-	snd_pcm_stream_lock_irq(substream);
-	avail = snd_pcm_playback_avail(runtime);
-	switch (runtime->status->state) {
-	case SNDRV_PCM_STATE_RUNNING:
-	case SNDRV_PCM_STATE_PREPARED:
-	case SNDRV_PCM_STATE_PAUSED:
-		if (avail >= runtime->control->avail_min) {
-			mask = EPOLLOUT | EPOLLWRNORM;
-			break;
-		}
-		/* Fall through */
-	case SNDRV_PCM_STATE_DRAINING:
-		mask = 0;
-		break;
-	default:
-		mask = EPOLLOUT | EPOLLWRNORM | EPOLLERR;
-		break;
-	}
-	snd_pcm_stream_unlock_irq(substream);
-	return mask;
-}
-
-static __poll_t snd_pcm_capture_poll(struct file *file, poll_table * wait)
-{
-	struct snd_pcm_file *pcm_file;
-	struct snd_pcm_substream *substream;
-	struct snd_pcm_runtime *runtime;
-        __poll_t mask;
-	snd_pcm_uframes_t avail;
-
-	pcm_file = file->private_data;
-
-	substream = pcm_file->substream;
-	if (PCM_RUNTIME_CHECK(substream))
-		return EPOLLIN | EPOLLRDNORM | EPOLLERR;
 	runtime = substream->runtime;
-
 	poll_wait(file, &runtime->sleep, wait);
 
+	mask = 0;
 	snd_pcm_stream_lock_irq(substream);
-	avail = snd_pcm_capture_avail(runtime);
+	avail = snd_pcm_avail(substream);
 	switch (runtime->status->state) {
 	case SNDRV_PCM_STATE_RUNNING:
 	case SNDRV_PCM_STATE_PREPARED:
 	case SNDRV_PCM_STATE_PAUSED:
-		if (avail >= runtime->control->avail_min) {
-			mask = EPOLLIN | EPOLLRDNORM;
-			break;
-		}
-		mask = 0;
+		if (avail >= runtime->control->avail_min)
+			mask = ok;
 		break;
 	case SNDRV_PCM_STATE_DRAINING:
-		if (avail > 0) {
-			mask = EPOLLIN | EPOLLRDNORM;
-			break;
+		if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+			mask = ok;
+			if (!avail)
+				mask |= EPOLLERR;
 		}
-		/* Fall through */
+		break;
 	default:
-		mask = EPOLLIN | EPOLLRDNORM | EPOLLERR;
+		mask = ok | EPOLLERR;
 		break;
 	}
 	snd_pcm_stream_unlock_irq(substream);
@@ -3707,7 +3650,7 @@ const struct file_operations snd_pcm_f_ops[2] = {
 		.open =			snd_pcm_playback_open,
 		.release =		snd_pcm_release,
 		.llseek =		no_llseek,
-		.poll =			snd_pcm_playback_poll,
+		.poll =			snd_pcm_poll,
 		.unlocked_ioctl =	snd_pcm_ioctl,
 		.compat_ioctl = 	snd_pcm_ioctl_compat,
 		.mmap =			snd_pcm_mmap,
@@ -3721,7 +3664,7 @@ const struct file_operations snd_pcm_f_ops[2] = {
 		.open =			snd_pcm_capture_open,
 		.release =		snd_pcm_release,
 		.llseek =		no_llseek,
-		.poll =			snd_pcm_capture_poll,
+		.poll =			snd_pcm_poll,
 		.unlocked_ioctl =	snd_pcm_ioctl,
 		.compat_ioctl = 	snd_pcm_ioctl_compat,
 		.mmap =			snd_pcm_mmap,

sound/core/seq/seq_ports.c

@@ -669,7 +669,7 @@ int snd_seq_event_port_attach(int client,
 	/* Set up the port */
 	memset(&portinfo, 0, sizeof(portinfo));
 	portinfo.addr.client = client;
-	strlcpy(portinfo.name, portname ? portname : "Unamed port",
+	strlcpy(portinfo.name, portname ? portname : "Unnamed port",
 		sizeof(portinfo.name));
 
 	portinfo.capability = cap;

sound/core/seq/seq_timer.c

@@ -371,9 +371,7 @@ static int initialize_timer(struct snd_seq_timer *tmr)
 
 	tmr->ticks = 1;
 	if (!(t->hw.flags & SNDRV_TIMER_HW_SLAVE)) {
-		unsigned long r = t->hw.resolution;
-		if (! r && t->hw.c_resolution)
-			r = t->hw.c_resolution(t);
+		unsigned long r = snd_timer_resolution(tmr->timeri);
 		if (r) {
 			tmr->ticks = (unsigned int)(1000000000uL / (r * freq));
 			if (! tmr->ticks)

sound/core/timer.c

@@ -427,25 +427,35 @@ int snd_timer_close(struct snd_timer_instance *timeri)
 }
 EXPORT_SYMBOL(snd_timer_close);
 
+static unsigned long snd_timer_hw_resolution(struct snd_timer *timer)
+{
+	if (timer->hw.c_resolution)
+		return timer->hw.c_resolution(timer);
+	else
+		return timer->hw.resolution;
+}
+
 unsigned long snd_timer_resolution(struct snd_timer_instance *timeri)
 {
 	struct snd_timer * timer;
+	unsigned long ret = 0;
+	unsigned long flags;
 
 	if (timeri == NULL)
 		return 0;
 	timer = timeri->timer;
 	if (timer) {
-		if (timer->hw.c_resolution)
-			return timer->hw.c_resolution(timer);
-		return timer->hw.resolution;
+		spin_lock_irqsave(&timer->lock, flags);
+		ret = snd_timer_hw_resolution(timer);
+		spin_unlock_irqrestore(&timer->lock, flags);
 	}
-	return 0;
+	return ret;
 }
 EXPORT_SYMBOL(snd_timer_resolution);
 
 static void snd_timer_notify1(struct snd_timer_instance *ti, int event)
 {
-	struct snd_timer *timer;
+	struct snd_timer *timer = ti->timer;
 	unsigned long resolution = 0;
 	struct snd_timer_instance *ts;
 	struct timespec tstamp;
@@ -457,14 +467,14 @@ static void snd_timer_notify1(struct snd_timer_instance *ti, int event)
 	if (snd_BUG_ON(event < SNDRV_TIMER_EVENT_START ||
 		       event > SNDRV_TIMER_EVENT_PAUSE))
 		return;
-	if (event == SNDRV_TIMER_EVENT_START ||
-	    event == SNDRV_TIMER_EVENT_CONTINUE)
-		resolution = snd_timer_resolution(ti);
+	if (timer &&
+	    (event == SNDRV_TIMER_EVENT_START ||
+	     event == SNDRV_TIMER_EVENT_CONTINUE))
+		resolution = snd_timer_hw_resolution(timer);
 	if (ti->ccallback)
 		ti->ccallback(ti, event, &tstamp, resolution);
 	if (ti->flags & SNDRV_TIMER_IFLG_SLAVE)
 		return;
-	timer = ti->timer;
 	if (timer == NULL)
 		return;
 	if (timer->hw.flags & SNDRV_TIMER_HW_SLAVE)
@@ -771,10 +781,7 @@ void snd_timer_interrupt(struct snd_timer * timer, unsigned long ticks_left)
 	spin_lock_irqsave(&timer->lock, flags);
 
 	/* remember the current resolution */
-	if (timer->hw.c_resolution)
-		resolution = timer->hw.c_resolution(timer);
-	else
-		resolution = timer->hw.resolution;
+	resolution = snd_timer_hw_resolution(timer);
 
 	/* loop for all active instances
 	 * Here we cannot use list_for_each_entry because the active_list of a
@@ -1014,12 +1021,8 @@ void snd_timer_notify(struct snd_timer *timer, int event, struct timespec *tstam
 	spin_lock_irqsave(&timer->lock, flags);
 	if (event == SNDRV_TIMER_EVENT_MSTART ||
 	    event == SNDRV_TIMER_EVENT_MCONTINUE ||
-	    event == SNDRV_TIMER_EVENT_MRESUME) {
-		if (timer->hw.c_resolution)
-			resolution = timer->hw.c_resolution(timer);
-		else
-			resolution = timer->hw.resolution;
-	}
+	    event == SNDRV_TIMER_EVENT_MRESUME)
+		resolution = snd_timer_hw_resolution(timer);
 	list_for_each_entry(ti, &timer->active_list_head, active_list) {
 		if (ti->ccallback)
 			ti->ccallback(ti, event, tstamp, resolution);
@@ -1656,10 +1659,8 @@ static int snd_timer_user_gstatus(struct file *file,
 	mutex_lock(&register_mutex);
 	t = snd_timer_find(&tid);
 	if (t != NULL) {
-		if (t->hw.c_resolution)
-			gstatus.resolution = t->hw.c_resolution(t);
-		else
-			gstatus.resolution = t->hw.resolution;
+		spin_lock_irq(&t->lock);
+		gstatus.resolution = snd_timer_hw_resolution(t);
 		if (t->hw.precise_resolution) {
 			t->hw.precise_resolution(t, &gstatus.resolution_num,
 						 &gstatus.resolution_den);
@@ -1667,6 +1668,7 @@ static int snd_timer_user_gstatus(struct file *file,
 			gstatus.resolution_num = gstatus.resolution;
 			gstatus.resolution_den = 1000000000uL;
 		}
+		spin_unlock_irq(&t->lock);
 	} else {
 		err = -ENODEV;
 	}

sound/core/vmaster.c

@@ -421,13 +421,15 @@ struct snd_kcontrol *snd_ctl_make_virtual_master(char *name,
 	kctl->private_free = master_free;
 
 	/* additional (constant) TLV read */
-	if (tlv &&
-	    (tlv[0] == SNDRV_CTL_TLVT_DB_SCALE ||
-	     tlv[0] == SNDRV_CTL_TLVT_DB_MINMAX ||
-	     tlv[0] == SNDRV_CTL_TLVT_DB_MINMAX_MUTE)) {
-		kctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ;
-		memcpy(master->tlv, tlv, sizeof(master->tlv));
-		kctl->tlv.p = master->tlv;
+	if (tlv) {
+		unsigned int type = tlv[SNDRV_CTL_TLVO_TYPE];
+		if (type == SNDRV_CTL_TLVT_DB_SCALE ||
+		    type == SNDRV_CTL_TLVT_DB_MINMAX ||
+		    type == SNDRV_CTL_TLVT_DB_MINMAX_MUTE) {
+			kctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ;
+			memcpy(master->tlv, tlv, sizeof(master->tlv));
+			kctl->tlv.p = master->tlv;
+		}
 	}
 
 	return kctl;