Merge branch 'asoc-4.18' into asoc-next

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/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/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.

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 = {
+static struct snd_soc_component_driver tda1997x_codec_driver = {
 	.probe			= tda1997x_codec_probe,
 	.remove			= tda1997x_codec_remove,
-	.reg_word_size = sizeof(u16),
+	.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/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

@@ -313,7 +313,6 @@ 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/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

sound/soc/Makefile

 # SPDX-License-Identifier: GPL-2.0
-snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o
+snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-utils.o
 snd-soc-core-objs += soc-pcm.o soc-io.o soc-devres.o soc-ops.o
 snd-soc-core-$(CONFIG_SND_SOC_COMPRESS) += soc-compress.o
 

sound/soc/amd/acp-da7219-max98357a.c

@@ -33,17 +33,19 @@
 #include <linux/gpio.h>
 #include <linux/module.h>
 #include <linux/i2c.h>
+#include <linux/input.h>
 #include <linux/acpi.h>
 
+#include "acp.h"
 #include "../codecs/da7219.h"
 #include "../codecs/da7219-aad.h"
 
-#define CZ_PLAT_CLK 24000000
-#define MCLK_RATE 24576000
+#define CZ_PLAT_CLK 25000000
 #define DUAL_CHANNEL		2
 
 static struct snd_soc_jack cz_jack;
 static struct clk *da7219_dai_clk;
+extern int bt_uart_enable;
 
 static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd)
 {
@@ -62,7 +64,7 @@ static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd)
 	}
 
 	ret = snd_soc_dai_set_pll(codec_dai, 0, DA7219_SYSCLK_PLL,
-				  CZ_PLAT_CLK, MCLK_RATE);
+				  CZ_PLAT_CLK, DA7219_PLL_FREQ_OUT_98304);
 	if (ret < 0) {
 		dev_err(rtd->dev, "can't set codec pll: %d\n", ret);
 		return ret;
@@ -80,13 +82,17 @@ static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd)
 		return ret;
 	}
 
+	snd_jack_set_key(cz_jack.jack, SND_JACK_BTN_0, KEY_PLAYPAUSE);
+	snd_jack_set_key(cz_jack.jack, SND_JACK_BTN_1, KEY_VOLUMEUP);
+	snd_jack_set_key(cz_jack.jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN);
+	snd_jack_set_key(cz_jack.jack, SND_JACK_BTN_3, KEY_VOICECOMMAND);
+
 	da7219_aad_jack_det(component, &cz_jack);
 
 	return 0;
 }
 
-static int cz_da7219_hw_params(struct snd_pcm_substream *substream,
-			     struct snd_pcm_hw_params *params)
+static int da7219_clk_enable(struct snd_pcm_substream *substream)
 {
 	int ret = 0;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
@@ -100,11 +106,9 @@ static int cz_da7219_hw_params(struct snd_pcm_substream *substream,
 	return ret;
 }
 
-static int cz_da7219_hw_free(struct snd_pcm_substream *substream)
+static void da7219_clk_disable(void)
 {
 	clk_disable_unprepare(da7219_dai_clk);
-
-	return 0;
 }
 
 static const unsigned int channels[] = {
@@ -127,9 +131,12 @@ static const struct snd_pcm_hw_constraint_list constraints_channels = {
 	.mask = 0,
 };
 
-static int cz_fe_startup(struct snd_pcm_substream *substream)
+static int cz_da7219_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_card *card = rtd->card;
+	struct acp_platform_info *machine = snd_soc_card_get_drvdata(card);
 
 	/*
 	 * On this platform for PCM device we support stereo
@@ -141,23 +148,58 @@ static int cz_fe_startup(struct snd_pcm_substream *substream)
 	snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
 				   &constraints_rates);
 
-	return 0;
+	machine->i2s_instance = I2S_BT_INSTANCE;
+	return da7219_clk_enable(substream);
+}
+
+static void cz_da7219_shutdown(struct snd_pcm_substream *substream)
+{
+	da7219_clk_disable();
+}
+
+static int cz_max_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_card *card = rtd->card;
+	struct acp_platform_info *machine = snd_soc_card_get_drvdata(card);
+
+	machine->i2s_instance = I2S_SP_INSTANCE;
+	return da7219_clk_enable(substream);
+}
+
+static void cz_max_shutdown(struct snd_pcm_substream *substream)
+{
+	da7219_clk_disable();
+}
+
+static int cz_dmic_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_card *card = rtd->card;
+	struct acp_platform_info *machine = snd_soc_card_get_drvdata(card);
+
+	machine->i2s_instance = I2S_SP_INSTANCE;
+	return da7219_clk_enable(substream);
+}
+
+static void cz_dmic_shutdown(struct snd_pcm_substream *substream)
+{
+	da7219_clk_disable();
 }
 
-static struct snd_soc_ops cz_da7219_cap_ops = {
-	.hw_params = cz_da7219_hw_params,
-	.hw_free = cz_da7219_hw_free,
-	.startup = cz_fe_startup,
+static const struct snd_soc_ops cz_da7219_cap_ops = {
+	.startup = cz_da7219_startup,
+	.shutdown = cz_da7219_shutdown,
 };
 
-static struct snd_soc_ops cz_max_play_ops = {
-	.hw_params = cz_da7219_hw_params,
-	.hw_free = cz_da7219_hw_free,
+static const struct snd_soc_ops cz_max_play_ops = {
+	.startup = cz_max_startup,
+	.shutdown = cz_max_shutdown,
 };
 
-static struct snd_soc_ops cz_dmic_cap_ops = {
-	.hw_params = cz_da7219_hw_params,
-	.hw_free = cz_da7219_hw_free,
+static const struct snd_soc_ops cz_dmic_cap_ops = {
+	.startup = cz_dmic_startup,
+	.shutdown = cz_dmic_shutdown,
 };
 
 static struct snd_soc_dai_link cz_dai_7219_98357[] = {
@@ -240,10 +282,16 @@ static int cz_probe(struct platform_device *pdev)
 {
 	int ret;
 	struct snd_soc_card *card;
+	struct acp_platform_info *machine;
 
+	machine = devm_kzalloc(&pdev->dev, sizeof(struct acp_platform_info),
+			       GFP_KERNEL);
+	if (!machine)
+		return -ENOMEM;
 	card = &cz_card;
 	cz_card.dev = &pdev->dev;
 	platform_set_drvdata(pdev, card);
+	snd_soc_card_set_drvdata(card, machine);
 	ret = devm_snd_soc_register_card(&pdev->dev, &cz_card);
 	if (ret) {
 		dev_err(&pdev->dev,
@@ -251,6 +299,8 @@ static int cz_probe(struct platform_device *pdev)
 				cz_card.name, ret);
 		return ret;
 	}
+	bt_uart_enable = !device_property_read_bool(&pdev->dev,
+						    "bt-pad-enable");
 	return 0;
 }
 

sound/soc/amd/acp.h

@@ -10,17 +10,30 @@
 #define ACP_PLAYBACK_PTE_OFFSET			10
 #define ACP_CAPTURE_PTE_OFFSET			0
 
+/* Playback and Capture Offset for Stoney */
+#define ACP_ST_PLAYBACK_PTE_OFFSET	0x04
+#define ACP_ST_CAPTURE_PTE_OFFSET	0x00
+#define ACP_ST_BT_PLAYBACK_PTE_OFFSET	0x08
+#define ACP_ST_BT_CAPTURE_PTE_OFFSET	0x0c
+
 #define ACP_GARLIC_CNTL_DEFAULT			0x00000FB4
 #define ACP_ONION_CNTL_DEFAULT			0x00000FB4
 
 #define ACP_PHYSICAL_BASE			0x14000
 
-/* Playback SRAM address (as a destination in dma descriptor) */
-#define ACP_SHARED_RAM_BANK_1_ADDRESS		0x4002000
-
-/* Capture SRAM address (as a source in dma descriptor) */
-#define ACP_SHARED_RAM_BANK_5_ADDRESS		0x400A000
-#define ACP_SHARED_RAM_BANK_3_ADDRESS		0x4006000
+/*
+ * In case of I2S SP controller instance, Stoney uses SRAM bank 1 for
+ * playback and SRAM Bank 2 for capture where as in case of BT I2S
+ * Instance, Stoney uses SRAM Bank 3 for playback & SRAM Bank 4 will
+ * be used for capture. Carrizo uses I2S SP controller instance. SRAM Banks
+ * 1, 2, 3, 4 will be used for playback & SRAM Banks 5, 6, 7, 8 will be used
+ * for capture scenario.
+ */
+#define ACP_SRAM_BANK_1_ADDRESS		0x4002000
+#define ACP_SRAM_BANK_2_ADDRESS		0x4004000
+#define ACP_SRAM_BANK_3_ADDRESS		0x4006000
+#define ACP_SRAM_BANK_4_ADDRESS		0x4008000
+#define ACP_SRAM_BANK_5_ADDRESS		0x400A000
 
 #define ACP_DMA_RESET_TIME			10000
 #define ACP_CLOCK_EN_TIME_OUT_VALUE		0x000000FF
@@ -35,8 +48,13 @@
 
 #define TO_ACP_I2S_1   0x2
 #define TO_ACP_I2S_2   0x4
+#define TO_BLUETOOTH   0x3
 #define FROM_ACP_I2S_1 0xa
 #define FROM_ACP_I2S_2 0xb
+#define FROM_BLUETOOTH 0xb
+
+#define I2S_SP_INSTANCE                 0x01
+#define I2S_BT_INSTANCE                 0x02
 
 #define ACP_TILE_ON_MASK                0x03
 #define ACP_TILE_OFF_MASK               0x02
@@ -57,6 +75,14 @@
 #define ACP_TO_SYSRAM_CH_NUM 14
 #define I2S_TO_ACP_DMA_CH_NUM 15
 
+/* Playback DMA Channels for I2S BT instance */
+#define SYSRAM_TO_ACP_BT_INSTANCE_CH_NUM  8
+#define ACP_TO_I2S_DMA_BT_INSTANCE_CH_NUM 9
+
+/* Capture DMA Channels for I2S BT Instance */
+#define ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM 10
+#define I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM 11
+
 #define NUM_DSCRS_PER_CHANNEL 2
 
 #define PLAYBACK_START_DMA_DESCR_CH12 0
@@ -69,9 +95,23 @@
 #define CAPTURE_START_DMA_DESCR_CH15 6
 #define CAPTURE_END_DMA_DESCR_CH15 7
 
+/* I2S BT Instance DMA Descriptors */
+#define PLAYBACK_START_DMA_DESCR_CH8 8
+#define PLAYBACK_END_DMA_DESCR_CH8 9
+#define PLAYBACK_START_DMA_DESCR_CH9 10
+#define PLAYBACK_END_DMA_DESCR_CH9 11
+
+#define CAPTURE_START_DMA_DESCR_CH10 12
+#define CAPTURE_END_DMA_DESCR_CH10 13
+#define CAPTURE_START_DMA_DESCR_CH11 14
+#define CAPTURE_END_DMA_DESCR_CH11 15
+
 #define mmACP_I2S_16BIT_RESOLUTION_EN       0x5209
 #define ACP_I2S_MIC_16BIT_RESOLUTION_EN 0x01
 #define ACP_I2S_SP_16BIT_RESOLUTION_EN	0x02
+#define ACP_I2S_BT_16BIT_RESOLUTION_EN	0x04
+#define ACP_BT_UART_PAD_SELECT_MASK	0x1
+
 enum acp_dma_priority_level {
 	/* 0x0 Specifies the DMA channel is given normal priority */
 	ACP_DMA_PRIORITY_LEVEL_NORMAL = 0x0,
@@ -84,20 +124,39 @@ struct audio_substream_data {
 	struct page *pg;
 	unsigned int order;
 	u16 num_of_pages;
+	u16 i2s_instance;
 	u16 direction;
+	u16 ch1;
+	u16 ch2;
+	u16 destination;
+	u16 dma_dscr_idx_1;
+	u16 dma_dscr_idx_2;
+	u32 pte_offset;
+	u32 sram_bank;
+	u32 byte_cnt_high_reg_offset;
+	u32 byte_cnt_low_reg_offset;
 	uint64_t size;
-	u64 i2ssp_renderbytescount;
-	u64 i2ssp_capturebytescount;
+	u64 bytescount;
 	void __iomem *acp_mmio;
 };
 
 struct audio_drv_data {
 	struct snd_pcm_substream *play_i2ssp_stream;
 	struct snd_pcm_substream *capture_i2ssp_stream;
+	struct snd_pcm_substream *play_i2sbt_stream;
+	struct snd_pcm_substream *capture_i2sbt_stream;
 	void __iomem *acp_mmio;
 	u32 asic_type;
 };
 
+/*
+ * this structure used for platform data transfer between machine driver
+ * and dma driver
+ */
+struct acp_platform_info {
+	u16 i2s_instance;
+};
+
 union acp_dma_count {
 	struct {
 	u32 low;
@@ -115,21 +174,23 @@ enum {
 };
 
 enum {
-	ACP_DMA_ATTRIBUTES_SHAREDMEM_TO_DAGB_ONION = 0x0,
-	ACP_DMA_ATTRIBUTES_SHARED_MEM_TO_DAGB_GARLIC = 0x1,
-	ACP_DMA_ATTRIBUTES_DAGB_ONION_TO_SHAREDMEM = 0x8,
-	ACP_DMA_ATTRIBUTES_DAGB_GARLIC_TO_SHAREDMEM = 0x9,
-	ACP_DMA_ATTRIBUTES_FORCE_SIZE = 0xF
+	ACP_DMA_ATTR_SHAREDMEM_TO_DAGB_ONION = 0x0,
+	ACP_DMA_ATTR_SHARED_MEM_TO_DAGB_GARLIC = 0x1,
+	ACP_DMA_ATTR_DAGB_ONION_TO_SHAREDMEM = 0x8,
+	ACP_DMA_ATTR_DAGB_GARLIC_TO_SHAREDMEM = 0x9,
+	ACP_DMA_ATTR_FORCE_SIZE = 0xF
 };
 
 typedef struct acp_dma_dscr_transfer {
 	/* Specifies the source memory location for the DMA data transfer. */
 	u32 src;
-	/* Specifies the destination memory location to where the data will
+	/*
+	 * Specifies the destination memory location to where the data will
 	 * be transferred.
 	 */
 	u32 dest;
-	/* Specifies the number of bytes need to be transferred
+	/*
+	 * Specifies the number of bytes need to be transferred
 	 * from source to destination memory.Transfer direction & IOC enable
 	 */
 	u32 xfer_val;

sound/soc/atmel/Kconfig

@@ -88,4 +88,13 @@ config SND_ATMEL_SOC_TSE850_PCM5142
 	help
 	  Say Y if you want to add support for the ASoC driver for the
 	  Axentia TSE-850 with a PCM5142 codec.
+
+config SND_ATMEL_SOC_I2S
+	tristate "Atmel ASoC driver for boards using I2S"
+	depends on OF && (ARCH_AT91 || COMPILE_TEST)
+	select SND_SOC_GENERIC_DMAENGINE_PCM
+	select REGMAP_MMIO
+	help
+	  Say Y or M if you want to add support for Atmel ASoc driver for boards
+	  using I2S.
 endif

sound/soc/atmel/Makefile

@@ -3,10 +3,12 @@
 snd-soc-atmel-pcm-pdc-objs := atmel-pcm-pdc.o
 snd-soc-atmel-pcm-dma-objs := atmel-pcm-dma.o
 snd-soc-atmel_ssc_dai-objs := atmel_ssc_dai.o
+snd-soc-atmel-i2s-objs := atmel-i2s.o
 
 obj-$(CONFIG_SND_ATMEL_SOC_PDC) += snd-soc-atmel-pcm-pdc.o
 obj-$(CONFIG_SND_ATMEL_SOC_DMA) += snd-soc-atmel-pcm-dma.o
 obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel_ssc_dai.o
+obj-$(CONFIG_SND_ATMEL_SOC_I2S) += snd-soc-atmel-i2s.o
 
 # AT91 Machine Support
 snd-soc-sam9g20-wm8731-objs := sam9g20_wm8731.o

sound/soc/atmel/atmel_ssc_dai.c

@@ -820,7 +820,7 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
 		if (ret < 0) {
 			printk(KERN_WARNING
 					"atmel_ssc_dai: request_irq failure\n");
-			pr_debug("Atmel_ssc_dai: Stoping clock\n");
+			pr_debug("Atmel_ssc_dai: Stopping clock\n");
 			clk_disable(ssc_p->ssc->clk);
 			return ret;
 		}
@@ -1002,8 +1002,7 @@ static const struct snd_soc_component_driver atmel_ssc_component = {
 
 static int asoc_ssc_init(struct device *dev)
 {
-	struct platform_device *pdev = to_platform_device(dev);
-	struct ssc_device *ssc = platform_get_drvdata(pdev);
+	struct ssc_device *ssc = dev_get_drvdata(dev);
 	int ret;
 
 	ret = snd_soc_register_component(dev, &atmel_ssc_component,
@@ -1033,8 +1032,7 @@ static int asoc_ssc_init(struct device *dev)
 
 static void asoc_ssc_exit(struct device *dev)
 {
-	struct platform_device *pdev = to_platform_device(dev);
-	struct ssc_device *ssc = platform_get_drvdata(pdev);
+	struct ssc_device *ssc = dev_get_drvdata(dev);
 
 	if (ssc->pdata->use_dma)
 		atmel_pcm_dma_platform_unregister(dev);

sound/soc/bcm/Kconfig

@@ -11,7 +11,6 @@ config SND_BCM2835_SOC_I2S
 config SND_SOC_CYGNUS
 	tristate "SoC platform audio for Broadcom Cygnus chips"
 	depends on ARCH_BCM_CYGNUS || COMPILE_TEST
-	depends on HAS_DMA
 	help
 	  Say Y if you want to add support for ASoC audio on Broadcom
 	  Cygnus chips (bcm958300, bcm958305, bcm911360)

sound/soc/cirrus/Kconfig

@@ -9,6 +9,23 @@ config SND_EP93XX_SOC
 config SND_EP93XX_SOC_I2S
 	tristate
 
+if SND_EP93XX_SOC_I2S
+
+config SND_EP93XX_SOC_I2S_WATCHDOG
+	bool "IRQ based underflow watchdog workaround"
+	default y
+	help
+	  I2S controller on EP93xx seems to have undocumented HW issue.
+	  Underflow of internal I2S controller FIFO could confuse the
+	  state machine and the whole stream can be shifted by one byte
+	  until I2S is disabled. This option enables IRQ based watchdog
+	  which disables and re-enables I2S in case of underflow and
+	  fills FIFO with zeroes.
+
+	  If you are unsure how to answer this question, answer Y.
+
+endif # if SND_EP93XX_SOC_I2S
+
 config SND_EP93XX_SOC_AC97
 	tristate
 	select AC97_BUS

sound/soc/cirrus/ep93xx-i2s.c

@@ -35,8 +35,12 @@
 
 #define EP93XX_I2S_TXCLKCFG		0x00
 #define EP93XX_I2S_RXCLKCFG		0x04
+#define EP93XX_I2S_GLSTS		0x08
 #define EP93XX_I2S_GLCTRL		0x0C
 
+#define EP93XX_I2S_I2STX0LFT		0x10
+#define EP93XX_I2S_I2STX0RT		0x14
+
 #define EP93XX_I2S_TXLINCTRLDATA	0x28
 #define EP93XX_I2S_TXCTRL		0x2C
 #define EP93XX_I2S_TXWRDLEN		0x30
@@ -55,12 +59,22 @@
 
 #define EP93XX_I2S_TXLINCTRLDATA_R_JUST	BIT(2) /* Right justify */
 
+/*
+ * Transmit empty interrupt level select:
+ * 0 - Generate interrupt when FIFO is half empty
+ * 1 - Generate interrupt when FIFO is empty
+ */
+#define EP93XX_I2S_TXCTRL_TXEMPTY_LVL	BIT(0)
+#define EP93XX_I2S_TXCTRL_TXUFIE	BIT(1) /* Transmit interrupt enable */
+
 #define EP93XX_I2S_CLKCFG_LRS		(1 << 0) /* lrclk polarity */
 #define EP93XX_I2S_CLKCFG_CKP		(1 << 1) /* Bit clock polarity */
 #define EP93XX_I2S_CLKCFG_REL		(1 << 2) /* First bit transition */
 #define EP93XX_I2S_CLKCFG_MASTER	(1 << 3) /* Master mode */
 #define EP93XX_I2S_CLKCFG_NBCG		(1 << 4) /* Not bit clock gating */
 
+#define EP93XX_I2S_GLSTS_TX0_FIFO_FULL	BIT(12)
+
 struct ep93xx_i2s_info {
 	struct clk			*mclk;
 	struct clk			*sclk;
@@ -98,7 +112,6 @@ static inline unsigned ep93xx_i2s_read_reg(struct ep93xx_i2s_info *info,
 static void ep93xx_i2s_enable(struct ep93xx_i2s_info *info, int stream)
 {
 	unsigned base_reg;
-	int i;
 
 	if ((ep93xx_i2s_read_reg(info, EP93XX_I2S_TX0EN) & 0x1) == 0 &&
 	    (ep93xx_i2s_read_reg(info, EP93XX_I2S_RX0EN) & 0x1) == 0) {
@@ -111,27 +124,36 @@ static void ep93xx_i2s_enable(struct ep93xx_i2s_info *info, int stream)
 		ep93xx_i2s_write_reg(info, EP93XX_I2S_GLCTRL, 1);
 	}
 
-	/* Enable fifos */
+	/* Enable fifo */
 	if (stream == SNDRV_PCM_STREAM_PLAYBACK)
 		base_reg = EP93XX_I2S_TX0EN;
 	else
 		base_reg = EP93XX_I2S_RX0EN;
-	for (i = 0; i < 3; i++)
-		ep93xx_i2s_write_reg(info, base_reg + (i * 4), 1);
+	ep93xx_i2s_write_reg(info, base_reg, 1);
+
+	/* Enable TX IRQs (FIFO empty or underflow) */
+	if (IS_ENABLED(CONFIG_SND_EP93XX_SOC_I2S_WATCHDOG) &&
+	    stream == SNDRV_PCM_STREAM_PLAYBACK)
+		ep93xx_i2s_write_reg(info, EP93XX_I2S_TXCTRL,
+				     EP93XX_I2S_TXCTRL_TXEMPTY_LVL |
+				     EP93XX_I2S_TXCTRL_TXUFIE);
 }
 
 static void ep93xx_i2s_disable(struct ep93xx_i2s_info *info, int stream)
 {
 	unsigned base_reg;
-	int i;
 
-	/* Disable fifos */
+	/* Disable IRQs */
+	if (IS_ENABLED(CONFIG_SND_EP93XX_SOC_I2S_WATCHDOG) &&
+	    stream == SNDRV_PCM_STREAM_PLAYBACK)
+		ep93xx_i2s_write_reg(info, EP93XX_I2S_TXCTRL, 0);
+
+	/* Disable fifo */
 	if (stream == SNDRV_PCM_STREAM_PLAYBACK)
 		base_reg = EP93XX_I2S_TX0EN;
 	else
 		base_reg = EP93XX_I2S_RX0EN;
-	for (i = 0; i < 3; i++)
-		ep93xx_i2s_write_reg(info, base_reg + (i * 4), 0);
+	ep93xx_i2s_write_reg(info, base_reg, 0);
 
 	if ((ep93xx_i2s_read_reg(info, EP93XX_I2S_TX0EN) & 0x1) == 0 &&
 	    (ep93xx_i2s_read_reg(info, EP93XX_I2S_RX0EN) & 0x1) == 0) {
@@ -145,6 +167,37 @@ static void ep93xx_i2s_disable(struct ep93xx_i2s_info *info, int stream)
 	}
 }
 
+/*
+ * According to documentation I2S controller can handle underflow conditions
+ * just fine, but in reality the state machine is sometimes confused so that
+ * the whole stream is shifted by one byte. The watchdog below disables the TX
+ * FIFO, fills the buffer with zeroes and re-enables the FIFO. State machine
+ * is being reset and by filling the buffer we get some time before next
+ * underflow happens.
+ */
+static irqreturn_t ep93xx_i2s_interrupt(int irq, void *dev_id)
+{
+	struct ep93xx_i2s_info *info = dev_id;
+
+	/* Disable FIFO */
+	ep93xx_i2s_write_reg(info, EP93XX_I2S_TX0EN, 0);
+	/*
+	 * Fill TX FIFO with zeroes, this way we can defer next IRQs as much as
+	 * possible and get more time for DMA to catch up. Actually there are
+	 * only 8 samples in this FIFO, so even on 8kHz maximum deferral here is
+	 * 1ms.
+	 */
+	while (!(ep93xx_i2s_read_reg(info, EP93XX_I2S_GLSTS) &
+		 EP93XX_I2S_GLSTS_TX0_FIFO_FULL)) {
+		ep93xx_i2s_write_reg(info, EP93XX_I2S_I2STX0LFT, 0);
+		ep93xx_i2s_write_reg(info, EP93XX_I2S_I2STX0RT, 0);
+	}
+	/* Re-enable FIFO */
+	ep93xx_i2s_write_reg(info, EP93XX_I2S_TX0EN, 1);
+
+	return IRQ_HANDLED;
+}
+
 static int ep93xx_i2s_dai_probe(struct snd_soc_dai *dai)
 {
 	struct ep93xx_i2s_info *info = snd_soc_dai_get_drvdata(dai);
@@ -394,6 +447,17 @@ static int ep93xx_i2s_probe(struct platform_device *pdev)
 	if (IS_ERR(info->regs))
 		return PTR_ERR(info->regs);
 
+	if (IS_ENABLED(CONFIG_SND_EP93XX_SOC_I2S_WATCHDOG)) {
+		int irq = platform_get_irq(pdev, 0);
+		if (irq <= 0)
+			return irq < 0 ? irq : -ENODEV;
+
+		err = devm_request_irq(&pdev->dev, irq, ep93xx_i2s_interrupt, 0,
+				       pdev->name, info);
+		if (err)
+			return err;
+	}
+
 	info->mclk = clk_get(&pdev->dev, "mclk");
 	if (IS_ERR(info->mclk)) {
 		err = PTR_ERR(info->mclk);

sound/soc/codecs/adau17x1.c

@@ -843,6 +843,15 @@ int adau17x1_setup_firmware(struct snd_soc_component *component,
 	struct adau *adau = snd_soc_component_get_drvdata(component);
 	struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component);
 
+	/* Check if sample rate is the same as before. If it is there is no
+	 * point in performing the below steps as the call to
+	 * sigmadsp_setup(...) will return directly when it finds the sample
+	 * rate to be the same as before. By checking this we can prevent an
+	 * audiable popping noise which occours when toggling DSP_RUN.
+	 */
+	if (adau->sigmadsp->current_samplerate == rate)
+		return 0;
+
 	snd_soc_dapm_mutex_lock(dapm);
 
 	ret = regmap_read(adau->regmap, ADAU17X1_DSP_SAMPLING_RATE, &dspsr);

sound/soc/codecs/max98088.c

@@ -1382,15 +1382,12 @@ static const char *eq_mode_name[] = {"EQ1 Mode", "EQ2 Mode"};
 
 static int max98088_get_channel(struct snd_soc_component *component, const char *name)
 {
-	int i;
-
-	for (i = 0; i < ARRAY_SIZE(eq_mode_name); i++)
-		if (strcmp(name, eq_mode_name[i]) == 0)
-			return i;
+	int ret;
 
-	/* Shouldn't happen */
+	ret = match_string(eq_mode_name, ARRAY_SIZE(eq_mode_name), name);
+	if (ret < 0)
 		dev_err(component->dev, "Bad EQ channel name '%s'\n", name);
-	return -EINVAL;
+	return ret;
 }
 
 static void max98088_setup_eq1(struct snd_soc_component *component)

sound/soc/codecs/max98095.c

@@ -1634,15 +1634,12 @@ static const char *bq_mode_name[] = {"Biquad1 Mode", "Biquad2 Mode"};
 static int max98095_get_bq_channel(struct snd_soc_component *component,
 				   const char *name)
 {
-	int i;
-
-	for (i = 0; i < ARRAY_SIZE(bq_mode_name); i++)
-		if (strcmp(name, bq_mode_name[i]) == 0)
-			return i;
+	int ret;
 
-	/* Shouldn't happen */
+	ret = match_string(bq_mode_name, ARRAY_SIZE(bq_mode_name), name);
+	if (ret < 0)
 		dev_err(component->dev, "Bad biquad channel name '%s'\n", name);
-	return -EINVAL;
+	return ret;
 }
 
 static int max98095_put_bq_enum(struct snd_kcontrol *kcontrol,