Merge tag 'sound-4.21-rc1' of git://git.kernel.org/pub/scm/linux/kern…

…el/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "There are no intensive changes in both ALSA and ASoC core parts while
  rather most of changes are a bunch of driver fixes and updates. A
  large diff pattern appears in ASoC TI part which now merges both OMAP
  and DaVinci stuff, but the rest spreads allover the places.

  Note that this pull request includes also some updates for LED trigger
  and platform drivers for mute LEDs, appearing in the diffstat as well.

  Some highlights:

  ASoC:
   - Preparatory work for merging the audio-graph and audio-graph-scu
     cards
   - A merge of TI OMAP and DaVinci directories, as both product lines
     get merged together. Also including a few architecture changes as
     well.
   - Major cleanups of the Maxim MAX9867 driver
   - Small fixes for tablets & co with Intel BYT/CHT chips
   - Lots of rsnd updates as usual
   - Support for Asahi Kaesi AKM4118, AMD ACP3x, Intel platforms with
     RT5660, Meson AXG S/PDIF inputs, several Qualcomm IPs and Xilinx
     I2S controllers

  HD-audio:
   - Introduce audio-mute LED trigger for replacing the former hackish
     dynamic binding
   - Huawei WMI hotkey and mute LED support
   - Refactoring of PM code and display power controls
   - Headset button support in the generic jack code
   - A few updates for Tegra
   - Fixups for HP EliteBook and ASUS UX391UA
   - Lots of updates for Intel ASoC HD-audio, including the improved DSP
     detection and the fallback binding from ASoC SST to legacy HD-audio
     controller drivers

  Others:
   - Updates for FireWire TASCAM and Fireface devices, some other fixes
   - A few potential Spectre v1 fixes that are all trivial"

* tag 'sound-4.21-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (280 commits)
  ALSA: HD-Audio: SKL+: force HDaudio legacy or SKL+ driver selection
  ALSA: HD-Audio: SKL+: abort probe if DSP is present and Skylake driver selected
  ALSA: HDA: export process_unsol_events()
  ALSA: hda/realtek: Enable audio jacks of ASUS UX391UA with ALC294
  ALSA: bebob: fix model-id of unit for Apogee Ensemble
  ALSA: emu10k1: Fix potential Spectre v1 vulnerabilities
  ALSA: rme9652: Fix potential Spectre v1 vulnerability
  ASoC: ti: Kconfig: Remove the deprecated options
  ARM: davinci_all_defconfig: Update the audio options
  ARM: omap1_defconfig: Do not select ASoC by default
  ARM: omap2plus_defconfig: Update the audio options
  ARM: davinci: dm365-evm: Update for the new ASoC Kcofnig options
  ARM: OMAP2: Update for new MCBSP Kconfig option
  ARM: OMAP1: Makefile: Update for new MCBSP Kconfig option
  MAINTAINERS: Add entry for sound/soc/ti and update the OMAP audio support
  ASoC: ti: Merge davinci and omap directories
  ALSA: hda: add mute LED support for HP EliteBook 840 G4
  ALSA: fireface: code refactoring to handle model-specific registers
  ALSA: fireface: add support for packet streaming on Fireface 800
  ALSA: fireface: allocate isochronous resources in mode-specific implementation
  ...

Documentation/devicetree/bindings/sound/ak4104.txt

@@ -12,8 +12,8 @@ Required properties:
 
 Optional properties:
 
-  - reset-gpio : a GPIO spec for the reset pin. If specified, it will be
-		 deasserted before communication to the device starts.
+  - reset-gpios : a GPIO spec for the reset pin. If specified, it will be
+		  deasserted before communication to the device starts.
 
 Example:
 

Documentation/devicetree/bindings/sound/ak4118.txt

@@ -0,0 +1,22 @@
+AK4118 S/PDIF transceiver
+
+This device supports I2C mode.
+
+Required properties:
+
+- compatible : "asahi-kasei,ak4118"
+- reg : The I2C address of the device for I2C
+- reset-gpios: A GPIO specifier for the reset pin
+- irq-gpios: A GPIO specifier for the IRQ pin
+
+Example:
+
+&i2c {
+	ak4118: ak4118@13 {
+		#sound-dai-cells = <0>;
+		compatible = "asahi-kasei,ak4118";
+		reg = <0x13>;
+		reset-gpios = <&gpio 0 GPIO_ACTIVE_LOW>
+		irq-gpios = <&gpio 1 GPIO_ACTIVE_HIGH>;
+	};
+};

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

@@ -0,0 +1,22 @@
+* Amlogic Audio SPDIF Input
+
+Required properties:
+- compatible: 'amlogic,axg-spdifin'
+- interrupts: interrupt specifier for the spdif input.
+- clocks: list of clock phandle, one for each entry clock-names.
+- clock-names: should contain the following:
+  * "pclk" : peripheral clock.
+  * "refclk" : spdif input reference clock
+- #sound-dai-cells: must be 0.
+
+Example on the A113 SoC:
+
+spdifin: audio-controller@400 {
+	compatible = "amlogic,axg-spdifin";
+	reg = <0x0 0x400 0x0 0x30>;
+	#sound-dai-cells = <0>;
+	interrupts = <GIC_SPI 87 IRQ_TYPE_EDGE_RISING>;
+	clocks = <&clkc_audio AUD_CLKID_SPDIFIN>,
+		 <&clkc_audio AUD_CLKID_SPDIFIN_CLK>;
+	clock-names = "pclk", "refclk";
+};

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

@@ -32,7 +32,9 @@ Required properties:
 Optional properties:
 - pa-gpios: GPIO used to control external amplifier.
 
+-----------------------
 Example: Single DAI case
+-----------------------
 
 	sound_card {
 		compatible = "audio-graph-card";
@@ -61,7 +63,9 @@ Example: Single DAI case
 		};
 	};
 
+-----------------------
 Example: Multi DAI case
+-----------------------
 
 	sound-card {
 		compatible = "audio-graph-card";
@@ -130,3 +134,204 @@ Example: Multi DAI case
 		};
 	};
 
+
+-----------------------
+Example: Sampling Rate Conversion
+-----------------------
+
+	sound_card {
+		compatible = "audio-graph-card";
+
+		label = "sound-card";
+		prefix = "codec";
+		routing = "codec Playback", "DAI0 Playback",
+			  "DAI0 Capture",   "codec Capture";
+		convert-rate = <48000>;
+
+		dais = <&cpu_port>;
+	};
+
+	audio-codec {
+		...
+		port {
+			codec_endpoint: endpoint {
+				remote-endpoint = <&cpu_endpoint>;
+			};
+		};
+	};
+
+	dai-controller {
+		...
+		cpu_port: port {
+			cpu_endpoint: endpoint {
+				remote-endpoint = <&codec_endpoint>;
+
+				dai-format = "left_j";
+				...
+			};
+		};
+	};
+
+-----------------------
+Example: 2 CPU 1 Codec (Mixing)
+-----------------------
+
+	sound_card {
+		compatible = "audio-graph-card";
+
+		label = "sound-card";
+		routing = "codec Playback", "DAI0 Playback",
+			  "codec Playback", "DAI1 Playback",
+			  "DAI0 Capture",   "codec Capture";
+
+		dais = <&cpu_port>;
+	};
+
+	audio-codec {
+		...
+
+		audio-graph-card,prefix = "codec";
+		audio-graph-card,convert-rate = <48000>;
+		port {
+			reg = <0>;
+			codec_endpoint0: endpoint@0 {
+				remote-endpoint = <&cpu_endpoint0>;
+			};
+			codec_endpoint1: endpoint@1 {
+				remote-endpoint = <&cpu_endpoint1>;
+			};
+		};
+	};
+
+	dai-controller {
+		...
+		cpu_port: port {
+			cpu_endpoint0: endpoint@0 {
+				remote-endpoint = <&codec_endpoint0>;
+
+				dai-format = "left_j";
+				...
+			};
+			cpu_endpoint1: endpoint@1 {
+				remote-endpoint = <&codec_endpoint1>;
+
+				dai-format = "left_j";
+				...
+			};
+		};
+	};
+
+-----------------------
+Example: Multi DAI with DPCM
+-----------------------
+
+	CPU0 ------ ak4613
+	CPU1 ------ HDMI
+	CPU2 ------ PCM3168A-p	/* DPCM 1ch/2ch */
+	CPU3 --/		/* DPCM 3ch/4ch */
+	CPU4 --/		/* DPCM 5ch/6ch */
+	CPU5 --/		/* DPCM 7ch/8ch */
+	CPU6 ------ PCM3168A-c
+
+	sound_card: sound {
+		compatible = "audio-graph-card";
+
+		label = "sound-card";
+
+		routing =	"pcm3168a Playback", "DAI2 Playback",
+				"pcm3168a Playback", "DAI3 Playback",
+				"pcm3168a Playback", "DAI4 Playback",
+				"pcm3168a Playback", "DAI5 Playback";
+
+		dais = <&snd_port0	/* ak4613 */
+			&snd_port1	/* HDMI0  */
+			&snd_port2	/* pcm3168a playback */
+			&snd_port3	/* pcm3168a capture  */
+			>;
+	};
+
+	ak4613: codec@10 {
+		...
+		port {
+			ak4613_endpoint: endpoint {
+				remote-endpoint = <&rsnd_endpoint0>;
+			};
+		};
+	};
+
+	pcm3168a: audio-codec@44 {
+		...
+		audio-graph-card,prefix = "pcm3168a";
+		audio-graph-card,convert-channels = <8>; /* TDM Split */
+		ports {
+			port@0 {
+				reg = <0>;
+				pcm3168a_endpoint_p1: endpoint@1 {
+					remote-endpoint = <&rsnd_endpoint2>;
+					...
+				};
+				pcm3168a_endpoint_p2: endpoint@2 {
+					remote-endpoint = <&rsnd_endpoint3>;
+					...
+				};
+				pcm3168a_endpoint_p3: endpoint@3 {
+					remote-endpoint = <&rsnd_endpoint4>;
+					...
+				};
+				pcm3168a_endpoint_p4: endpoint@4 {
+					remote-endpoint = <&rsnd_endpoint5>;
+					...
+				};
+			};
+			port@1 {
+				reg = <1>;
+				pcm3168a_endpoint_c: endpoint {
+					remote-endpoint = <&rsnd_endpoint6>;
+					...
+				};
+			};
+		};
+	};
+
+	&sound {
+		ports {
+			snd_port0: port@0 {
+				rsnd_endpoint0: endpoint {
+					remote-endpoint = <&ak4613_endpoint>;
+					...
+				};
+			};
+			snd_port1: port@1 {
+				rsnd_endpoint1: endpoint {
+					remote-endpoint = <&dw_hdmi0_snd_in>;
+					...
+				};
+			};
+			snd_port2: port@2 {
+				#address-cells = <1>;
+				#size-cells = <0>;
+				rsnd_endpoint2: endpoint@2 {
+					remote-endpoint = <&pcm3168a_endpoint_p1>;
+					...
+				};
+				rsnd_endpoint3: endpoint@3 {
+					remote-endpoint = <&pcm3168a_endpoint_p2>;
+					...
+				};
+				rsnd_endpoint4: endpoint@4 {
+					remote-endpoint = <&pcm3168a_endpoint_p3>;
+					...
+				};
+				rsnd_endpoint5: endpoint@5 {
+					remote-endpoint = <&pcm3168a_endpoint_p4>;
+					...
+				};
+			};
+			snd_port3: port@6 {
+				rsnd_endpoint6: endpoint {
+					remote-endpoint = <&pcm3168a_endpoint_c>;
+					...
+				};
+			};
+		};
+	};

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

@@ -77,11 +77,9 @@ Example 2. 2 CPU 1 Codec (Mixing)
 		compatible = "audio-graph-scu-card";
 
 		label = "sound-card";
-		prefix = "codec";
 		routing = "codec Playback", "DAI0 Playback",
 			  "codec Playback", "DAI1 Playback",
 			  "DAI0 Capture",   "codec Capture";
-		convert-rate = <48000>;
 
 		dais = <&cpu_port0
 			&cpu_port1>;
@@ -90,6 +88,8 @@ Example 2. 2 CPU 1 Codec (Mixing)
 	audio-codec {
 		...
 
+		audio-graph-card,prefix = "codec";
+		audio-graph-card,convert-rate = <48000>;
 		port {
 			codec_endpoint0: endpoint {
 				remote-endpoint = <&cpu_endpoint0>;

Documentation/devicetree/bindings/sound/cs4270.txt

@@ -10,8 +10,8 @@ Required properties:
 
 Optional properties:
 
-  - reset-gpio : a GPIO spec for the reset pin. If specified, it will be
-		 deasserted before communication to the codec starts.
+  - reset-gpios : a GPIO spec for the reset pin. If specified, it will be
+		  deasserted before communication to the codec starts.
 
 Example:
 

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

@@ -30,6 +30,11 @@ Optional properties:
 - ti,hwmods : Must be "mcasp<n>", n is controller instance starting 0
 - tx-num-evt : FIFO levels.
 - rx-num-evt : FIFO levels.
+- dismod : Specify the drive on TX pin during inactive slots
+	0 : 3-state
+	2 : logic low
+	3 : logic high
+	Defaults to 'logic low' when the property is not present
 - sram-size-playback : size of sram to be allocated during playback
 - sram-size-capture  : size of sram to be allocated during capture
 - interrupts : Interrupt numbers for McASP

Documentation/devicetree/bindings/sound/dmic.txt

@@ -9,6 +9,7 @@ Optional properties:
 	- dmicen-gpios: GPIO specifier for dmic to control start and stop
 	- num-channels: Number of microphones on this DAI
 	- wakeup-delay-ms: Delay (in ms) after enabling the DMIC
+	- modeswitch-delay-ms: Delay (in ms) to complete DMIC mode switch
 
 Example node:
 
@@ -17,4 +18,5 @@ Example node:
 		dmicen-gpios = <&gpio4 3 GPIO_ACTIVE_HIGH>;
 		num-channels = <1>;
 		wakeup-delay-ms <50>;
+		modeswitch-delay-ms <35>;
 	};

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

@@ -35,13 +35,13 @@ Required properties:
 
   - fsl,sai-synchronous-rx: This is a boolean property. If present, indicating
 			  that SAI will work in the synchronous mode (sync Tx
-			  with Rx) which means both the transimitter and the
+			  with Rx) which means both the transmitter and the
 			  receiver will send and receive data by following
 			  receiver's bit clocks and frame sync clocks.
 
   - fsl,sai-asynchronous: This is a boolean property. If present, indicating
 			  that SAI will work in the asynchronous mode, which
-			  means both transimitter and receiver will send and
+			  means both transmitter and receiver will send and
 			  receive data by following their own bit clocks and
 			  frame sync clocks separately.
 
@@ -58,8 +58,8 @@ Optional properties (for mx6ul):
 Note:
 - If both fsl,sai-asynchronous and fsl,sai-synchronous-rx are absent, the
   default synchronous mode (sync Rx with Tx) will be used, which means both
-  transimitter and receiver will send and receive data by following clocks
-  of transimitter.
+  transmitter and receiver will send and receive data by following clocks
+  of transmitter.
 - fsl,sai-asynchronous and fsl,sai-synchronous-rx are exclusive.
 
 Example:

Documentation/devicetree/bindings/sound/omap-mcpdm.txt

@@ -7,6 +7,8 @@ Required properties:
        <L3 interconnect address, size>;
 - interrupts: Interrupt number for McPDM
 - ti,hwmods: Name of the hwmod associated to the McPDM
+- clocks:  phandle for the pdmclk provider, likely <&twl6040>
+- clock-names: Must be "pdmclk"
 
 Example:
 
@@ -18,3 +20,11 @@ mcpdm: mcpdm@40132000 {
 	interrupt-parent = <&gic>;
 	ti,hwmods = "mcpdm";
 };
+
+In board DTS file the pdmclk needs to be added:
+
+&mcpdm {
+	clocks = <&twl6040>;
+	clock-names = "pdmclk";
+	status = "okay";
+};