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

Pull sound updates from Takashi Iwai: "It's been busy summer weeks and hence lots of changes, partly for a few new drivers and partly for a wide range of fixes. Here are highlights: ALSA Core: - Fix rawmidi buffer management, code cleanup / refactoring - Fix the SG-buffer page handling with incorrect fallback size - Fix the stall at virmidi trigger callback with a large buffer; also offloading and code-refactoring along with it - Various ALSA sequencer code cleanups ASoC: - Deploy the standard snd_pcm_stop_xrun() helper in several drivers - Support for providing name prefixes to generic component nodes - Quite a few fixes for DPCM as it gains a bit wider use and more robust testing - Generalization of the DIO2125 support to a simple amplifier driver - Accessory detection support for the audio graph card - DT support for PXA AC'97 devices - Quirks for a number of new x86 systems - Support for AM Logic Meson, Everest ES7154, Intel systems with RT5682, Qualcomm QDSP6 and WCD9335, Realtek RT5682 and TI TAS5707 HD-audio: - Code refactoring in HD-audio ext codec codes to drop own classes; preliminary works for the upcoming legacy codec support - Generalized DRM audio component for the upcoming radeon / amdgpu support - Unification of mic mute-LED and GPIO support for various codecs - Further improvement of CA0132 codec support including Recon3D - Proper vga_switcheroo handling for AMD i-GPU - Update of model list in documentation - Fixups for another HP Spectre x360, Conexant codecs, power-save blacklist update USB-audio: - Fix the invalid sample rate setup with external clock - Support of UAC3 selector units and processing units - Basic UAC3 power-domain support - Support for Encore mDSD and Thesycon-based DSD devices - Preparation for future complete callback changes Firewire: - Add support for MOTU Traveler Misc: - The endianess notation fixes in various drivers - Add fall-through comment in lots of drivers - Various sparse warning fixes, e.g. about PCM format types" * tag 'sound-4.19-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (529 commits) ASoC: adav80x: mark expected switch fall-through ASoC: da7219: Add delays to capture path to remove DC offset noise ALSA: usb-audio: Mark expected switch fall-through ALSA: mixart: Mark expected switch fall-through ALSA: opl3: Mark expected switch fall-through ALSA: hda/ca0132 - Add exit commands for Recon3D ALSA: hda/ca0132 - Change mixer controls for Recon3D ALSA: hda/ca0132 - Add Recon3D input and output select commands ALSA: hda/ca0132 - Add DSP setup defaults for Recon3D ALSA: hda/ca0132 - Add Recon3D startup functions and setup ALSA: hda/ca0132 - Add bool variable to enable/disable pci region2 mmio ALSA: hda/ca0132 - Add Recon3D pincfg ALSA: hda/ca0132 - Add quirk ID and enum for Recon3D ALSA: hda/ca0132 - Add alt_functions unsolicited response ALSA: hda/ca0132 - Clean up ca0132_init function. ALSA: hda/ca0132 - Create mmio gpio function to make code clearer ASoC: wm_adsp: Make DSP name configurable by codec driver ASoC: wm_adsp: Declare firmware controls from codec driver ASoC: max98373: Added software reset register to readable registers ASoC: wm_adsp: Correct DSP pointer for preloader control ...
2026-05-01 15:00:59 -07:00 · 2018-08-14 14:10:30 -07:00
parent 2c20443ec2 f5b6c1fcb4
commit 747f62305d
498 changed files with 19446 additions and 6562 deletions
@@ -0,0 +1,32 @@
+Generic AC97 Device Properties
+
+This documents describes the devicetree bindings for an ac97 controller child
+node describing ac97 codecs.
+
+Required properties:
+-compatible : Must be "ac97,vendor_id1,vendor_id2
+	      The ids shall be the 4 characters hexadecimal encoding, such as
+	      given by "%04x" formatting of printf
+-reg	    : Must be the ac97 codec number, between 0 and 3
+
+Example:
+ac97: sound@40500000 {
+	compatible = "marvell,pxa270-ac97";
+	reg = < 0x40500000 0x1000 >;
+	interrupts = <14>;
+	reset-gpios = <&gpio 95 GPIO_ACTIVE_HIGH>;
+	#sound-dai-cells = <1>;
+	pinctrl-names = "default";
+	pinctrl-0 = < &pinctrl_ac97_default >;
+	clocks = <&clks CLK_AC97>, <&clks CLK_AC97CONF>;
+	clock-names = "AC97CLK", "AC97CONFCLK";
+
+	#address-cells = <1>;
+	#size-cells = <0>;
+	audio-codec@0 {
+		reg = <0>;
+		compatible = "ac97,574d,4c13";
+		clocks = <&fixed_wm9713_clock>;
+		clock-names = "ac97_clk";
+	}
+};
@@ -0,0 +1,23 @@
+* Amlogic Audio FIFO controllers
+
+Required properties:
+- compatible: 'amlogic,axg-toddr' or
+	      'amlogic,axg-frddr'
+- reg: physical base address of the controller and length of memory
+       mapped region.
+- interrupts: interrupt specifier for the fifo.
+- clocks: phandle to the fifo peripheral clock provided by the audio
+	  clock controller.
+- resets: phandle to memory ARB line provided by the arb reset controller.
+- #sound-dai-cells: must be 0.
+
+Example of FRDDR A on the A113 SoC:
+
+frddr_a: audio-controller@1c0 {
+	compatible = "amlogic,axg-frddr";
+	reg = <0x0 0x1c0 0x0 0x1c>;
+	#sound-dai-cells = <0>;
+	interrupts = <GIC_SPI 88 IRQ_TYPE_EDGE_RISING>;
+	clocks = <&clkc_audio AUD_CLKID_FRDDR_A>;
+	resets = <&arb AXG_ARB_FRDDR_A>;
+};
@@ -0,0 +1,124 @@
+Amlogic AXG sound card:
+
+Required properties:
+
+- compatible: "amlogic,axg-sound-card"
+- model : User specified audio sound card name, one string
+
+Optional properties:
+
+- audio-aux-devs : List of phandles pointing to auxiliary devices
+- audio-widgets : Please refer to widgets.txt.
+- audio-routing : A list of the connections between audio components.
+
+Subnodes:
+
+- dai-link: Container for dai-link level properties and the CODEC
+	    sub-nodes. There should be at least one (and probably more)
+	    subnode of this type.
+
+Required dai-link properties:
+
+- sound-dai: phandle and port of the CPU DAI.
+
+Required TDM Backend dai-link properties:
+- dai-format : CPU/CODEC common audio format
+
+Optional TDM Backend dai-link properties:
+- dai-tdm-slot-rx-mask-{0,1,2,3}: Receive direction slot masks
+- dai-tdm-slot-tx-mask-{0,1,2,3}: Transmit direction slot masks
+				  When omitted, mask is assumed to have to no
+				  slots. A valid must have at one slot, so at
+				  least one these mask should be provided with
+				  an enabled slot.
+- dai-tdm-slot-num : Please refer to tdm-slot.txt.
+		     If omitted, slot number is set to accommodate the largest
+		     mask provided.
+- dai-tdm-slot-width : Please refer to tdm-slot.txt. default to 32 if omitted.
+- mclk-fs : Multiplication factor between stream rate and mclk
+
+Backend dai-link subnodes:
+
+- codec: dai-link representing backend links should have at least one subnode.
+	 One subnode for each codec of the dai-link.
+	 dai-link representing frontend links have no codec, therefore have no
+	 subnodes
+
+Required codec subnodes properties:
+
+- sound-dai: phandle and port of the CODEC DAI.
+
+Optional codec subnodes properties:
+
+- dai-tdm-slot-tx-mask : Please refer to tdm-slot.txt.
+- dai-tdm-slot-rx-mask : Please refer to tdm-slot.txt.
+
+Example:
+
+sound {
+	compatible = "amlogic,axg-sound-card";
+	model = "AXG-S420";
+	audio-aux-devs = <&tdmin_a>, <&tdmout_c>;
+	audio-widgets = "Line", "Lineout",
+			"Line", "Linein",
+			"Speaker", "Speaker1 Left",
+			"Speaker", "Speaker1 Right";
+			"Speaker", "Speaker2 Left",
+			"Speaker", "Speaker2 Right";
+	audio-routing = "TDMOUT_C IN 0", "FRDDR_A OUT 2",
+			"SPDIFOUT IN 0", "FRDDR_A OUT 3",
+			"TDM_C Playback", "TDMOUT_C OUT",
+			"TDMIN_A IN 2", "TDM_C Capture",
+			"TDMIN_A IN 5", "TDM_C Loopback",
+			"TODDR_A IN 0", "TDMIN_A OUT",
+			"Lineout", "Lineout AOUTL",
+			"Lineout", "Lineout AOUTR",
+			"Speaker1 Left", "SPK1 OUT_A",
+			"Speaker2 Left", "SPK2 OUT_A",
+			"Speaker1 Right", "SPK1 OUT_B",
+			"Speaker2 Right", "SPK2 OUT_B",
+			"Linein AINL", "Linein",
+			"Linein AINR", "Linein";
+
+	dai-link@0 {
+		sound-dai = <&frddr_a>;
+	};
+
+	dai-link@1 {
+		sound-dai = <&toddr_a>;
+	};
+
+	dai-link@2 {
+		sound-dai = <&tdmif_c>;
+		dai-format = "i2s";
+		dai-tdm-slot-tx-mask-2 = <1 1>;
+		dai-tdm-slot-tx-mask-3 = <1 1>;
+		dai-tdm-slot-rx-mask-1 = <1 1>;
+		mclk-fs = <256>;
+
+		codec@0 {
+			sound-dai = <&lineout>;
+		};
+
+		codec@1 {
+			sound-dai = <&speaker_amp1>;
+		};
+
+		codec@2 {
+			sound-dai = <&speaker_amp2>;
+		};
+
+		codec@3 {
+			sound-dai = <&linein>;
+		};
+
+	};
+
+	dai-link@3 {
+		sound-dai = <&spdifout>;
+
+		codec {
+			sound-dai = <&spdif_dit>;
+		};
+	};
+};
@@ -0,0 +1,20 @@
+* Amlogic Audio SPDIF Output
+
+Required properties:
+- compatible: 'amlogic,axg-spdifout'
+- clocks: list of clock phandle, one for each entry clock-names.
+- clock-names: should contain the following:
+  * "pclk" : peripheral clock.
+  * "mclk" : master clock
+- #sound-dai-cells: must be 0.
+
+Example on the A113 SoC:
+
+spdifout: audio-controller@480 {
+	compatible = "amlogic,axg-spdifout";
+	reg = <0x0 0x480 0x0 0x50>;
+	#sound-dai-cells = <0>;
+	clocks = <&clkc_audio AUD_CLKID_SPDIFOUT>,
+		 <&clkc_audio AUD_CLKID_SPDIFOUT_CLK>;
+	clock-names = "pclk", "mclk";
+};
@@ -0,0 +1,28 @@
+* Amlogic Audio TDM formatters
+
+Required properties:
+- compatible: 'amlogic,axg-tdmin' or
+	      'amlogic,axg-tdmout'
+- reg: physical base address of the controller and length of memory
+       mapped region.
+- clocks: list of clock phandle, one for each entry clock-names.
+- clock-names: should contain the following:
+  * "pclk"     : peripheral clock.
+  * "sclk"     : bit clock.
+  * "sclk_sel" : bit clock input multiplexer.
+  * "lrclk"    : sample clock
+  * "lrclk_sel": sample clock input multiplexer
+
+Example of TDMOUT_A on the A113 SoC:
+
+tdmout_a: audio-controller@500 {
+	compatible = "amlogic,axg-tdmout";
+	reg = <0x0 0x500 0x0 0x40>;
+	clocks = <&clkc_audio AUD_CLKID_TDMOUT_A>,
+		 <&clkc_audio AUD_CLKID_TDMOUT_A_SCLK>,
+		 <&clkc_audio AUD_CLKID_TDMOUT_A_SCLK_SEL>,
+		 <&clkc_audio AUD_CLKID_TDMOUT_A_LRCLK>,
+		 <&clkc_audio AUD_CLKID_TDMOUT_A_LRCLK>;
+	clock-names = "pclk", "sclk", "sclk_sel",
+		      "lrclk", "lrclk_sel";
+};
@@ -0,0 +1,22 @@
+* Amlogic Audio TDM Interfaces
+
+Required properties:
+- compatible: 'amlogic,axg-tdm-iface'
+- clocks: list of clock phandle, one for each entry clock-names.
+- clock-names: should contain the following:
+  * "sclk" : bit clock.
+  * "lrclk": sample clock
+  * "mclk" : master clock
+	     -> optional if the interface is in clock slave mode.
+- #sound-dai-cells: must be 0.
+
+Example of TDM_A on the A113 SoC:
+
+tdmif_a: audio-controller@0 {
+	compatible = "amlogic,axg-tdm-iface";
+	#sound-dai-cells = <0>;
+	clocks = <&clkc_audio AUD_CLKID_MST_A_MCLK>,
+		 <&clkc_audio AUD_CLKID_MST_A_SCLK>,
+		 <&clkc_audio AUD_CLKID_MST_A_LRCLK>;
+	clock-names = "mclk", "sclk", "lrclk";
+};
@@ -15,7 +15,6 @@ Required properties:
 - clock-names:    Should be one of each entry matching the clocks phandles list:
                  - "pclk" (peripheral clock) Required.
                  - "gclk" (generated clock) Optional (1).
-                  - "aclk" (Audio PLL clock) Optional (1).
                  - "muxclk" (I2S mux clock) Optional (1).

 Optional properties:
@@ -23,9 +22,9 @@ Optional properties:
 - princtrl-names: Should contain only one value - "default".


-(1) : Only the peripheral clock is required. The generated clock, the Audio
-      PLL clock adn the I2S mux clock are optional and should only be set
-      together, when Master Mode is required.
+(1) : Only the peripheral clock is required. The generated clock and the I2S
+      mux clock are optional and should only be set together, when Master Mode
+      is required.

 Example:

@@ -40,8 +39,8 @@ Example:
 			(AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1) |
 			 AT91_XDMAC_DT_PERID(32))>;
 		dma-names = "tx", "rx";
-		clocks = <&i2s0_clk>, <&i2s0_gclk>, <&audio_pll_pmc>, <&i2s0muxck>;
-		clock-names = "pclk", "gclk", "aclk", "muxclk";
+		clocks = <&i2s0_clk>, <&i2s0_gclk>, <&i2s0muxck>;
+		clock-names = "pclk", "gclk", "muxclk";
 		pinctrl-names = "default";
 		pinctrl-0 = <&pinctrl_i2s0_default>;
 	};
@@ -18,6 +18,8 @@ Below are same as Simple-Card.
 - bitclock-inversion
 - frame-inversion
 - mclk-fs
+- hp-det-gpio
+- mic-det-gpio
 - dai-tdm-slot-num
 - dai-tdm-slot-width
 - clocks / system-clock-frequency
@@ -1,12 +0,0 @@
-DIO2125 Audio Driver
-
-Required properties:
- compatible : "dioo,dio2125"
- enable-gpios : the gpio connected to the enable pin of the dio2125
-
-Example:
-
-amp: analog-amplifier {
-	compatible = "dioo,dio2125";
-	enable-gpios = <&gpio GPIOH_3 0>;
-};
@@ -1,10 +1,15 @@
 ES7134 i2s DA converter

 Required properties:
- compatible : "everest,es7134" or "everest,es7144"
+- compatible : "everest,es7134" or
+               "everest,es7144" or
+	       "everest,es7154"
+- VDD-supply : regulator phandle for the VDD supply
+- PVDD-supply: regulator phandle for the PVDD supply for the es7154

 Example:

 i2s_codec: external-codec {
 	compatible = "everest,es7134";
+	VDD-supply = <&vcc_5v>;
 };
@@ -0,0 +1,28 @@
+ES7241 i2s AD converter
+
+Required properties:
+- compatible : "everest,es7241"
+- VDDP-supply: regulator phandle for the VDDA supply
+- VDDA-supply: regulator phandle for the VDDP supply
+- VDDD-supply: regulator phandle for the VDDD supply
+
+Optional properties:
+- reset-gpios: gpio connected to the reset pin
+- m0-gpios   : gpio connected to the m0 pin
+- m1-gpios   : gpio connected to the m1 pin
+- everest,sdout-pull-down:
+   Format used by the serial interface is controlled by pulling
+   the sdout. If the sdout is pulled down, leftj format is used.
+   If this property is not provided, sdout is assumed to pulled
+   up and i2s format is used
+
+Example:
+
+linein: audio-codec@2 {
+	#sound-dai-cells = <0>;
+	compatible = "everest,es7241";
+	VDDA-supply = <&vcc_3v3>;
+	VDDP-supply = <&vcc_3v3>;
+	VDDD-supply = <&vcc_3v3>;
+	reset-gpios = <&gpio GPIOH_42>;
+};
@@ -0,0 +1,27 @@
+Marvell PXA2xx audio complex
+
+This descriptions matches the AC97 controller found in pxa2xx and pxa3xx series.
+
+Required properties:
+  - compatible: should be one of the following:
+    "marvell,pxa250-ac97"
+    "marvell,pxa270-ac97"
+    "marvell,pxa300-ac97"
+  - reg: device MMIO address space
+  - interrupts: single interrupt generated by AC97 IP
+  - clocks: input clock of the AC97 IP, refer to clock-bindings.txt
+
+Optional properties:
+  - pinctrl-names, pinctrl-0: refer to pinctrl-bindings.txt
+  - reset-gpios: gpio used for AC97 reset, refer to gpio.txt
+
+Example:
+	ac97: sound@40500000 {
+		compatible = "marvell,pxa250-ac97";
+		reg = < 0x40500000 0x1000 >;
+		interrupts = <14>;
+		reset-gpios = <&gpio 113 GPIO_ACTIVE_HIGH>;
+		#sound-dai-cells = <1>;
+		pinctrl-names = "default";
+		pinctrl-0 = < &pmux_ac97_default >;
+	};
@@ -5,6 +5,14 @@ Required properties:
 	compatible	Must be "mrvl,pxa-ssp-dai"
 	port		A phandle reference to a PXA ssp upstream device

+Optional properties:
+
+	clock-names
+	clocks		Through "clock-names" and "clocks", external clocks
+			can be configured. If a clock names "extclk" exists,
+			it will be set to the mclk rate of the audio stream
+			and be used as clock provider of the DAI.
+
 Example:

 	/* upstream device */
@@ -1,15 +0,0 @@
-DT bindings for ARM PXA2xx PCM platform driver
-
-This is just a dummy driver that registers the PXA ASoC platform driver.
-It does not have any resources assigned.
-
-Required properties:
-
-	- compatible		'mrvl,pxa-pcm-audio'
-
-Example:
-
-	pxa_pcm_audio: snd_soc_pxa_audio {
-		compatible = "mrvl,pxa-pcm-audio";
-	};
-
@@ -0,0 +1,24 @@
+Name prefix:
+
+Card implementing the routing property define the connection between
+audio components as list of string pair. Component using the same
+sink/source names may use the name prefix property to prepend the
+name of their sinks/sources with the provided string.
+
+Optional name prefix property:
+- sound-name-prefix : string using as prefix for the sink/source names of
+		      the component.
+
+Example: Two instances of the same component.
+
+amp0: analog-amplifier@0 {
+	compatible = "simple-audio-amplifier";
+	enable-gpios = <&gpio GPIOH_3 0>;
+	sound-name-prefix = "FRONT";
+};
+
+amp1: analog-amplifier@1 {
+	compatible = "simple-audio-amplifier";
+	enable-gpios = <&gpio GPIOH_4 0>;
+	sound-name-prefix = "BACK";
+};
@@ -7,7 +7,7 @@ This binding describes the APQ8096 sound card, which uses qdsp for audio.
 	Value type: <stringlist>
 	Definition: must be "qcom,apq8096-sndcard"

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

+Obsolete:
+	qcom,model: String for soundcard name (Use model instead)
+	qcom,audio-routing: A list of the connections between audio components.
+			    (Use audio-routing instead)
+
 Example:

 audio {
 	compatible = "qcom,apq8096-sndcard";
-	qcom,model = "DB820c";
+	model = "DB820c";

 	mm1-dai-link {
 		link-name = "MultiMedia1";
@@ -18,6 +18,11 @@ used by the apr service device.
 = ADM routing
 "routing" subnode of the ADM node represents adm routing specific configuration

+- compatible:
+	Usage: required
+	Value type: <stringlist>
+	Definition: must be "qcom,q6adm-routing".
+
 - #sound-dai-cells
 	Usage: required
 	Value type: <u32>
@@ -28,6 +33,7 @@ q6adm@8 {
 	compatible = "qcom,q6adm";
 	reg = <APR_SVC_ADM>;
 	q6routing: routing {
+		compatible = "qcom,q6adm-routing";
 		#sound-dai-cells = <0>;
 	};
 };
@@ -17,6 +17,11 @@ used by all apr services. Must contain the following properties.
 subnode of "dais" representing board specific dai setup.
 "dais" node should have following properties followed by dai children.

+- compatible:
+	Usage: required
+	Value type: <stringlist>
+	Definition: must be "qcom,q6afe-dais"
+
 - #sound-dai-cells
 	Usage: required
 	Value type: <u32>
@@ -100,6 +105,7 @@ q6afe@4 {
 	reg = <APR_SVC_AFE>;

 	dais {
+		compatible = "qcom,q6afe-dais";
 		#sound-dai-cells = <1>;
 		#address-cells = <1>;
 		#size-cells = <0>;
@@ -17,6 +17,11 @@ used by the apr service device.
 = ASM DAIs (Digial Audio Interface)
 "dais" subnode of the ASM node represents dai specific configuration

+- compatible:
+	Usage: required
+	Value type: <stringlist>
+	Definition: must be "qcom,q6asm-dais".
+
 - #sound-dai-cells
 	Usage: required
 	Value type: <u32>
@@ -28,6 +33,7 @@ q6asm@7 {
 	compatible = "qcom,q6asm";
 	reg = <APR_SVC_ASM>;
 	q6asmdai: dais {
+		compatible = "qcom,q6asm-dais";
 		#sound-dai-cells = <1>;
 	};
 };
@@ -0,0 +1,80 @@
+* Qualcomm Technologies Inc. SDM845 ASoC sound card driver
+
+This binding describes the SDM845 sound card, which uses qdsp for audio.
+
+- compatible:
+	Usage: required
+	Value type: <stringlist>
+	Definition: must be "qcom,sdm845-sndcard"
+
+- audio-routing:
+	Usage: Optional
+	Value type: <stringlist>
+	Definition:  A list of the connections between audio components.
+		  Each entry is a pair of strings, the first being the
+		  connection's sink, the second being the connection's
+		  source. Valid names could be power supplies, MicBias
+		  of codec and the jacks on the board.
+
+- model:
+	Usage: required
+	Value type: <stringlist>
+	Definition: The user-visible name of this sound card.
+
+= dailinks
+Each subnode of sndcard represents either a dailink, and subnodes of each
+dailinks would be cpu/codec/platform dais.
+
+- link-name:
+	Usage: required
+	Value type: <string>
+	Definition: User friendly name for dai link
+
+= CPU, PLATFORM, CODEC dais subnodes
+- cpu:
+	Usage: required
+	Value type: <subnode>
+	Definition: cpu dai sub-node
+
+- codec:
+	Usage: required
+	Value type: <subnode>
+	Definition: codec dai sub-node
+
+- platform:
+	Usage: Optional
+	Value type: <subnode>
+	Definition: platform dai sub-node
+
+- sound-dai:
+	Usage: required
+	Value type: <phandle>
+	Definition: dai phandle/s and port of CPU/CODEC/PLATFORM node.
+
+Example:
+
+audio {
+	compatible = "qcom,sdm845-sndcard";
+	model = "sdm845-snd-card";
+	pinctrl-names = "default", "sleep";
+	pinctrl-0 = <&pri_mi2s_active &pri_mi2s_ws_active>;
+	pinctrl-1 = <&pri_mi2s_sleep &pri_mi2s_ws_sleep>;
+
+	mm1-dai-link {
+		link-name = "MultiMedia1";
+		cpu {
+			sound-dai = <&q6asmdai MSM_FRONTEND_DAI_MULTIMEDIA1>;
+		};
+	};
+
+	pri-mi2s-dai-link {
+		link-name = "PRI MI2S Playback";
+		cpu {
+			sound-dai = <&q6afedai PRIMARY_MI2S_RX>;
+		};
+
+		platform {
+			sound-dai = <&q6routing>;
+		};
+	};
+};
--- a/Show More
+++ b/Show More