summaryrefslogtreecommitdiffstats
path: root/dts/Bindings/sound
diff options
context:
space:
mode:
authorSascha Hauer <s.hauer@pengutronix.de>2014-04-25 11:22:32 +0200
committerSascha Hauer <s.hauer@pengutronix.de>2014-04-28 09:08:46 +0200
commit00ce25c6dcdae5582ae4be37147ab33678adc995 (patch)
tree41c93102ae304a61738c31353e3cb5336ef0b297 /dts/Bindings/sound
parent0af79fbb6779921d3f1962773adb7fb57d3c89d4 (diff)
downloadbarebox-00ce25c6dcdae5582ae4be37147ab33678adc995.tar.gz
Add devicetree source files as of Linux-3.15-rc2
This adds the Linux dts files to barebox. The dts files are generated from Ian Campbells device-tree-rebasing.git: git://xenbits.xen.org/people/ianc/device-tree-rebasing.git The dts are found in dts/ in the barebox repository and will be updated from upstream regularly, probably for each upstream -rc. To keep the synchronization with upstream easy no changes to the original files are allowed under dts/. Instead changes to upstream dts files will be done using overlays in arch/$ARCH/dts/. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Diffstat (limited to 'dts/Bindings/sound')
-rw-r--r--dts/Bindings/sound/adi,adau1701.txt35
-rw-r--r--dts/Bindings/sound/adi,axi-i2s.txt31
-rw-r--r--dts/Bindings/sound/adi,axi-spdif-tx.txt30
-rw-r--r--dts/Bindings/sound/ak4104.txt22
-rw-r--r--dts/Bindings/sound/ak4554.c11
-rw-r--r--dts/Bindings/sound/ak4642.txt17
-rw-r--r--dts/Bindings/sound/ak5386.txt19
-rw-r--r--dts/Bindings/sound/alc5632.txt43
-rw-r--r--dts/Bindings/sound/armada-370db-audio.txt27
-rw-r--r--dts/Bindings/sound/atmel-at91sam9g20ek-wm8731-audio.txt26
-rw-r--r--dts/Bindings/sound/atmel-sam9x5-wm8731-audio.txt35
-rw-r--r--dts/Bindings/sound/atmel-wm8904.txt55
-rw-r--r--dts/Bindings/sound/bcm2835-i2s.txt25
-rw-r--r--dts/Bindings/sound/cs4270.txt21
-rw-r--r--dts/Bindings/sound/cs4271.txt50
-rw-r--r--dts/Bindings/sound/cs42l52.txt46
-rw-r--r--dts/Bindings/sound/cs42l73.txt22
-rw-r--r--dts/Bindings/sound/cs42xx8.txt28
-rw-r--r--dts/Bindings/sound/da9055.txt22
-rw-r--r--dts/Bindings/sound/davinci-evm-audio.txt49
-rw-r--r--dts/Bindings/sound/davinci-mcasp-audio.txt60
-rw-r--r--dts/Bindings/sound/eukrea-tlv320.txt21
-rw-r--r--dts/Bindings/sound/fsl,esai.txt55
-rw-r--r--dts/Bindings/sound/fsl,spdif.txt59
-rw-r--r--dts/Bindings/sound/fsl,ssi.txt93
-rw-r--r--dts/Bindings/sound/fsl-sai.txt40
-rw-r--r--dts/Bindings/sound/hdmi.txt17
-rw-r--r--dts/Bindings/sound/imx-audio-sgtl5000.txt49
-rw-r--r--dts/Bindings/sound/imx-audio-spdif.txt34
-rw-r--r--dts/Bindings/sound/imx-audio-wm8962.txt46
-rw-r--r--dts/Bindings/sound/imx-audmux.txt22
-rw-r--r--dts/Bindings/sound/max98090.txt43
-rw-r--r--dts/Bindings/sound/mrvl,pxa-ssp.txt28
-rw-r--r--dts/Bindings/sound/mrvl,pxa2xx-pcm.txt15
-rw-r--r--dts/Bindings/sound/mvebu-audio.txt34
-rw-r--r--dts/Bindings/sound/mxs-audio-sgtl5000.txt17
-rw-r--r--dts/Bindings/sound/mxs-saif.txt41
-rw-r--r--dts/Bindings/sound/nvidia,tegra-audio-alc5632.txt48
-rw-r--r--dts/Bindings/sound/nvidia,tegra-audio-max98090.txt51
-rw-r--r--dts/Bindings/sound/nvidia,tegra-audio-rt5640.txt52
-rw-r--r--dts/Bindings/sound/nvidia,tegra-audio-trimslice.txt21
-rw-r--r--dts/Bindings/sound/nvidia,tegra-audio-wm8753.txt40
-rw-r--r--dts/Bindings/sound/nvidia,tegra-audio-wm8903.txt60
-rw-r--r--dts/Bindings/sound/nvidia,tegra-audio-wm9712.txt60
-rw-r--r--dts/Bindings/sound/nvidia,tegra20-ac97.txt36
-rw-r--r--dts/Bindings/sound/nvidia,tegra20-das.txt12
-rw-r--r--dts/Bindings/sound/nvidia,tegra20-i2s.txt30
-rw-r--r--dts/Bindings/sound/nvidia,tegra30-ahub.txt85
-rw-r--r--dts/Bindings/sound/nvidia,tegra30-i2s.txt24
-rw-r--r--dts/Bindings/sound/omap-abe-twl6040.txt91
-rw-r--r--dts/Bindings/sound/omap-dmic.txt21
-rw-r--r--dts/Bindings/sound/omap-mcbsp.txt37
-rw-r--r--dts/Bindings/sound/omap-mcpdm.txt21
-rw-r--r--dts/Bindings/sound/omap-twl4030.txt63
-rw-r--r--dts/Bindings/sound/pcm1792a.txt18
-rw-r--r--dts/Bindings/sound/pcm512x.txt30
-rw-r--r--dts/Bindings/sound/renesas,fsi.txt26
-rw-r--r--dts/Bindings/sound/renesas,rsnd.txt105
-rw-r--r--dts/Bindings/sound/rt5640.txt50
-rw-r--r--dts/Bindings/sound/samsung,smdk-wm8994.txt14
-rw-r--r--dts/Bindings/sound/samsung-i2s.txt53
-rw-r--r--dts/Bindings/sound/sgtl5000.txt16
-rw-r--r--dts/Bindings/sound/simple-card.txt136
-rw-r--r--dts/Bindings/sound/sirf-audio-codec.txt17
-rw-r--r--dts/Bindings/sound/sirf-audio-port.txt20
-rw-r--r--dts/Bindings/sound/sirf-audio.txt41
-rw-r--r--dts/Bindings/sound/soc-ac97link.txt28
-rw-r--r--dts/Bindings/sound/spdif-receiver.txt10
-rw-r--r--dts/Bindings/sound/spdif-transmitter.txt10
-rw-r--r--dts/Bindings/sound/ssm2518.txt20
-rw-r--r--dts/Bindings/sound/tdm-slot.txt20
-rw-r--r--dts/Bindings/sound/ti,pcm1681.txt15
-rw-r--r--dts/Bindings/sound/ti,tas5086.txt43
-rw-r--r--dts/Bindings/sound/tlv320aic31xx.txt61
-rw-r--r--dts/Bindings/sound/tlv320aic32x4.txt30
-rw-r--r--dts/Bindings/sound/tlv320aic3x.txt59
-rw-r--r--dts/Bindings/sound/tpa6130a2.txt27
-rw-r--r--dts/Bindings/sound/ux500-mop500.txt39
-rw-r--r--dts/Bindings/sound/ux500-msp.txt43
-rw-r--r--dts/Bindings/sound/widgets.txt20
-rw-r--r--dts/Bindings/sound/wm8510.txt18
-rw-r--r--dts/Bindings/sound/wm8523.txt16
-rw-r--r--dts/Bindings/sound/wm8580.txt16
-rw-r--r--dts/Bindings/sound/wm8711.txt18
-rw-r--r--dts/Bindings/sound/wm8728.txt18
-rw-r--r--dts/Bindings/sound/wm8731.txt27
-rw-r--r--dts/Bindings/sound/wm8737.txt18
-rw-r--r--dts/Bindings/sound/wm8741.txt18
-rw-r--r--dts/Bindings/sound/wm8750.txt18
-rw-r--r--dts/Bindings/sound/wm8753.txt40
-rw-r--r--dts/Bindings/sound/wm8770.txt16
-rw-r--r--dts/Bindings/sound/wm8776.txt18
-rw-r--r--dts/Bindings/sound/wm8804.txt18
-rw-r--r--dts/Bindings/sound/wm8903.txt69
-rw-r--r--dts/Bindings/sound/wm8962.txt39
-rw-r--r--dts/Bindings/sound/wm8994.txt78
96 files changed, 3436 insertions, 0 deletions
diff --git a/dts/Bindings/sound/adi,adau1701.txt b/dts/Bindings/sound/adi,adau1701.txt
new file mode 100644
index 0000000..547a49b
--- /dev/null
+++ b/dts/Bindings/sound/adi,adau1701.txt
@@ -0,0 +1,35 @@
+Analog Devices ADAU1701
+
+Required properties:
+
+ - compatible: Should contain "adi,adau1701"
+ - reg: The i2c address. Value depends on the state of ADDR0
+ and ADDR1, as wired in hardware.
+
+Optional properties:
+
+ - reset-gpio: A GPIO spec to define which pin is connected to the
+ chip's !RESET pin. If specified, the driver will
+ assert a hardware reset at probe time.
+ - adi,pll-mode-gpios: An array of two GPIO specs to describe the GPIOs
+ the ADAU's PLL config pins are connected to.
+ The state of the pins are set according to the
+ configured clock divider on ASoC side before the
+ firmware is loaded.
+ - adi,pin-config: An array of 12 numerical values selecting one of the
+ pin configurations as described in the datasheet,
+ table 53. Note that the value of this property has
+ to be prefixed with '/bits/ 8'.
+
+Examples:
+
+ i2c_bus {
+ adau1701@34 {
+ compatible = "adi,adau1701";
+ reg = <0x34>;
+ reset-gpio = <&gpio 23 0>;
+ adi,pll-mode-gpios = <&gpio 24 0 &gpio 25 0>;
+ adi,pin-config = /bits/ 8 <0x4 0x7 0x5 0x5 0x4 0x4
+ 0x4 0x4 0x4 0x4 0x4 0x4>;
+ };
+ };
diff --git a/dts/Bindings/sound/adi,axi-i2s.txt b/dts/Bindings/sound/adi,axi-i2s.txt
new file mode 100644
index 0000000..5875ca4
--- /dev/null
+++ b/dts/Bindings/sound/adi,axi-i2s.txt
@@ -0,0 +1,31 @@
+ADI AXI-I2S controller
+
+Required properties:
+ - compatible : Must be "adi,axi-i2s-1.00.a"
+ - reg : Must contain I2S core's registers location and length
+ - clocks : Pairs of phandle and specifier referencing the controller's clocks.
+ The controller expects two clocks, the clock used for the AXI interface and
+ the clock used as the sampling rate reference clock sample.
+ - clock-names : "axi" for the clock to the AXI interface, "ref" for the sample
+ rate reference clock.
+ - dmas: Pairs of phandle and specifier for the DMA channels that are used by
+ the core. The core expects two dma channels, one for transmit and one for
+ receive.
+ - dma-names : "tx" for the transmit channel, "rx" for the receive channel.
+
+For more details on the 'dma', 'dma-names', 'clock' and 'clock-names' properties
+please check:
+ * resource-names.txt
+ * clock/clock-bindings.txt
+ * dma/dma.txt
+
+Example:
+
+ i2s: i2s@0x77600000 {
+ compatible = "adi,axi-i2s-1.00.a";
+ reg = <0x77600000 0x1000>;
+ clocks = <&clk 15>, <&audio_clock>;
+ clock-names = "axi", "ref";
+ dmas = <&ps7_dma 0>, <&ps7_dma 1>;
+ dma-names = "tx", "rx";
+ };
diff --git a/dts/Bindings/sound/adi,axi-spdif-tx.txt b/dts/Bindings/sound/adi,axi-spdif-tx.txt
new file mode 100644
index 0000000..46f3449
--- /dev/null
+++ b/dts/Bindings/sound/adi,axi-spdif-tx.txt
@@ -0,0 +1,30 @@
+ADI AXI-SPDIF controller
+
+Required properties:
+ - compatible : Must be "adi,axi-spdif-1.00.a"
+ - reg : Must contain SPDIF core's registers location and length
+ - clocks : Pairs of phandle and specifier referencing the controller's clocks.
+ The controller expects two clocks, the clock used for the AXI interface and
+ the clock used as the sampling rate reference clock sample.
+ - clock-names: "axi" for the clock to the AXI interface, "ref" for the sample
+ rate reference clock.
+ - dmas: Pairs of phandle and specifier for the DMA channel that is used by
+ the core. The core expects one dma channel for transmit.
+ - dma-names : Must be "tx"
+
+For more details on the 'dma', 'dma-names', 'clock' and 'clock-names' properties
+please check:
+ * resource-names.txt
+ * clock/clock-bindings.txt
+ * dma/dma.txt
+
+Example:
+
+ spdif: spdif@0x77400000 {
+ compatible = "adi,axi-spdif-tx-1.00.a";
+ reg = <0x77600000 0x1000>;
+ clocks = <&clk 15>, <&audio_clock>;
+ clock-names = "axi", "ref";
+ dmas = <&ps7_dma 0>;
+ dma-names = "tx";
+ };
diff --git a/dts/Bindings/sound/ak4104.txt b/dts/Bindings/sound/ak4104.txt
new file mode 100644
index 0000000..b902ee3
--- /dev/null
+++ b/dts/Bindings/sound/ak4104.txt
@@ -0,0 +1,22 @@
+AK4104 S/PDIF transmitter
+
+This device supports SPI mode only.
+
+Required properties:
+
+ - compatible : "asahi-kasei,ak4104"
+
+ - reg : The chip select number on the SPI bus
+
+Optional properties:
+
+ - reset-gpio : a GPIO spec for the reset pin. If specified, it will be
+ deasserted before communication to the device starts.
+
+Example:
+
+spdif: ak4104@0 {
+ compatible = "asahi-kasei,ak4104";
+ reg = <0>;
+ spi-max-frequency = <5000000>;
+};
diff --git a/dts/Bindings/sound/ak4554.c b/dts/Bindings/sound/ak4554.c
new file mode 100644
index 0000000..934fa02
--- /dev/null
+++ b/dts/Bindings/sound/ak4554.c
@@ -0,0 +1,11 @@
+AK4554 ADC/DAC
+
+Required properties:
+
+ - compatible : "asahi-kasei,ak4554"
+
+Example:
+
+ak4554-adc-dac {
+ compatible = "asahi-kasei,ak4554";
+};
diff --git a/dts/Bindings/sound/ak4642.txt b/dts/Bindings/sound/ak4642.txt
new file mode 100644
index 0000000..623d4e7
--- /dev/null
+++ b/dts/Bindings/sound/ak4642.txt
@@ -0,0 +1,17 @@
+AK4642 I2C transmitter
+
+This device supports I2C mode only.
+
+Required properties:
+
+ - compatible : "asahi-kasei,ak4642" or "asahi-kasei,ak4643" or "asahi-kasei,ak4648"
+ - reg : The chip select number on the I2C bus
+
+Example:
+
+&i2c {
+ ak4648: ak4648@0x12 {
+ compatible = "asahi-kasei,ak4642";
+ reg = <0x12>;
+ };
+};
diff --git a/dts/Bindings/sound/ak5386.txt b/dts/Bindings/sound/ak5386.txt
new file mode 100644
index 0000000..dc3914f
--- /dev/null
+++ b/dts/Bindings/sound/ak5386.txt
@@ -0,0 +1,19 @@
+AK5386 Single-ended 24-Bit 192kHz delta-sigma ADC
+
+This device has no control interface.
+
+Required properties:
+
+ - compatible : "asahi-kasei,ak5386"
+
+Optional properties:
+
+ - reset-gpio : a GPIO spec for the reset/power down pin.
+ If specified, it will be deasserted at probe time.
+
+Example:
+
+spdif: ak5386@0 {
+ compatible = "asahi-kasei,ak5386";
+ reset-gpio = <&gpio0 23>;
+};
diff --git a/dts/Bindings/sound/alc5632.txt b/dts/Bindings/sound/alc5632.txt
new file mode 100644
index 0000000..ffd886d
--- /dev/null
+++ b/dts/Bindings/sound/alc5632.txt
@@ -0,0 +1,43 @@
+ALC5632 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+ - compatible : "realtek,alc5632"
+
+ - reg : the I2C address of the device.
+
+ - gpio-controller : Indicates this device is a GPIO controller.
+
+ - #gpio-cells : Should be two. The first cell is the pin number and the
+ second cell is used to specify optional parameters (currently unused).
+
+Pins on the device (for linking into audio routes):
+
+ * SPK_OUTP
+ * SPK_OUTN
+ * HP_OUT_L
+ * HP_OUT_R
+ * AUX_OUT_P
+ * AUX_OUT_N
+ * LINE_IN_L
+ * LINE_IN_R
+ * PHONE_P
+ * PHONE_N
+ * MIC1_P
+ * MIC1_N
+ * MIC2_P
+ * MIC2_N
+ * MICBIAS1
+ * DMICDAT
+
+Example:
+
+alc5632: alc5632@1e {
+ compatible = "realtek,alc5632";
+ reg = <0x1a>;
+
+ gpio-controller;
+ #gpio-cells = <2>;
+};
diff --git a/dts/Bindings/sound/armada-370db-audio.txt b/dts/Bindings/sound/armada-370db-audio.txt
new file mode 100644
index 0000000..bf984d2
--- /dev/null
+++ b/dts/Bindings/sound/armada-370db-audio.txt
@@ -0,0 +1,27 @@
+Device Tree bindings for the Armada 370 DB audio
+================================================
+
+These Device Tree bindings are used to describe the audio complex
+found on the Armada 370 DB platform.
+
+Mandatory properties:
+
+ * compatible: must be "marvell,a370db-audio"
+
+ * marvell,audio-controller: a phandle that points to the audio
+ controller of the Armada 370 SoC.
+
+ * marvell,audio-codec: a set of three phandles that points to:
+
+ 1/ the analog audio codec connected to the Armada 370 SoC
+ 2/ the S/PDIF transceiver
+ 3/ the S/PDIF receiver
+
+Example:
+
+ sound {
+ compatible = "marvell,a370db-audio";
+ marvell,audio-controller = <&audio_controller>;
+ marvell,audio-codec = <&audio_codec &spdif_out &spdif_in>;
+ status = "okay";
+ };
diff --git a/dts/Bindings/sound/atmel-at91sam9g20ek-wm8731-audio.txt b/dts/Bindings/sound/atmel-at91sam9g20ek-wm8731-audio.txt
new file mode 100644
index 0000000..9c5a994
--- /dev/null
+++ b/dts/Bindings/sound/atmel-at91sam9g20ek-wm8731-audio.txt
@@ -0,0 +1,26 @@
+* Atmel at91sam9g20ek wm8731 audio complex
+
+Required properties:
+ - compatible: "atmel,at91sam9g20ek-wm8731-audio"
+ - atmel,model: The user-visible name of this sound complex.
+ - atmel,audio-routing: A list of the connections between audio components.
+ - atmel,ssc-controller: The phandle of the SSC controller
+ - atmel,audio-codec: The phandle of the WM8731 audio codec
+Optional properties:
+ - pinctrl-names, pinctrl-0: Please refer to pinctrl-bindings.txt
+
+Example:
+sound {
+ compatible = "atmel,at91sam9g20ek-wm8731-audio";
+ pinctrl-names = "default";
+ pinctrl-0 = <&pinctrl_pck0_as_mck>;
+
+ atmel,model = "wm8731 @ AT91SAMG20EK";
+
+ atmel,audio-routing =
+ "Ext Spk", "LHPOUT",
+ "Int MIC", "MICIN";
+
+ atmel,ssc-controller = <&ssc0>;
+ atmel,audio-codec = <&wm8731>;
+};
diff --git a/dts/Bindings/sound/atmel-sam9x5-wm8731-audio.txt b/dts/Bindings/sound/atmel-sam9x5-wm8731-audio.txt
new file mode 100644
index 0000000..0720857
--- /dev/null
+++ b/dts/Bindings/sound/atmel-sam9x5-wm8731-audio.txt
@@ -0,0 +1,35 @@
+* Atmel at91sam9x5ek wm8731 audio complex
+
+Required properties:
+ - compatible: "atmel,sam9x5-wm8731-audio"
+ - atmel,model: The user-visible name of this sound complex.
+ - atmel,ssc-controller: The phandle of the SSC controller
+ - atmel,audio-codec: The phandle of the WM8731 audio codec
+ - atmel,audio-routing: 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.
+
+Available audio endpoints for the audio-routing table:
+
+Board connectors:
+ * Headphone Jack
+ * Line In Jack
+
+wm8731 pins:
+cf Documentation/devicetree/bindings/sound/wm8731.txt
+
+Example:
+sound {
+ compatible = "atmel,sam9x5-wm8731-audio";
+
+ atmel,model = "wm8731 @ AT91SAM9X5EK";
+
+ atmel,audio-routing =
+ "Headphone Jack", "RHPOUT",
+ "Headphone Jack", "LHPOUT",
+ "LLINEIN", "Line In Jack",
+ "RLINEIN", "Line In Jack";
+
+ atmel,ssc-controller = <&ssc0>;
+ atmel,audio-codec = <&wm8731>;
+};
diff --git a/dts/Bindings/sound/atmel-wm8904.txt b/dts/Bindings/sound/atmel-wm8904.txt
new file mode 100644
index 0000000..8bbe50c
--- /dev/null
+++ b/dts/Bindings/sound/atmel-wm8904.txt
@@ -0,0 +1,55 @@
+Atmel ASoC driver with wm8904 audio codec complex
+
+Required properties:
+ - compatible: "atmel,asoc-wm8904"
+ - atmel,model: The user-visible name of this sound complex.
+ - atmel,audio-routing: 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 for sources and
+ sinks are the WM8904's pins, and the jacks on the board:
+
+ WM8904 pins:
+
+ * IN1L
+ * IN1R
+ * IN2L
+ * IN2R
+ * IN3L
+ * IN3R
+ * HPOUTL
+ * HPOUTR
+ * LINEOUTL
+ * LINEOUTR
+ * MICBIAS
+
+ Board connectors:
+
+ * Headphone Jack
+ * Line In Jack
+ * Mic
+
+ - atmel,ssc-controller: The phandle of the SSC controller
+ - atmel,audio-codec: The phandle of the WM8904 audio codec
+
+Optional properties:
+ - pinctrl-names, pinctrl-0: Please refer to pinctrl-bindings.txt
+
+Example:
+sound {
+ compatible = "atmel,asoc-wm8904";
+ pinctrl-names = "default";
+ pinctrl-0 = <&pinctrl_pck0_as_mck>;
+
+ atmel,model = "wm8904 @ AT91SAM9N12EK";
+
+ atmel,audio-routing =
+ "Headphone Jack", "HPOUTL",
+ "Headphone Jack", "HPOUTR",
+ "IN2L", "Line In Jack",
+ "IN2R", "Line In Jack",
+ "Mic", "MICBIAS",
+ "IN1L", "Mic";
+
+ atmel,ssc-controller = <&ssc0>;
+ atmel,audio-codec = <&wm8904>;
+};
diff --git a/dts/Bindings/sound/bcm2835-i2s.txt b/dts/Bindings/sound/bcm2835-i2s.txt
new file mode 100644
index 0000000..65783de
--- /dev/null
+++ b/dts/Bindings/sound/bcm2835-i2s.txt
@@ -0,0 +1,25 @@
+* Broadcom BCM2835 SoC I2S/PCM module
+
+Required properties:
+- compatible: "brcm,bcm2835-i2s"
+- reg: A list of base address and size entries:
+ * The first entry should cover the PCM registers
+ * The second entry should cover the PCM clock registers
+- dmas: List of DMA controller phandle and DMA request line ordered pairs.
+- dma-names: Identifier string for each DMA request line in the dmas property.
+ These strings correspond 1:1 with the ordered pairs in dmas.
+
+ One of the DMA channels will be responsible for transmission (should be
+ named "tx") and one for reception (should be named "rx").
+
+Example:
+
+bcm2835_i2s: i2s@7e203000 {
+ compatible = "brcm,bcm2835-i2s";
+ reg = <0x7e203000 0x20>,
+ <0x7e101098 0x02>;
+
+ dmas = <&dma 2>,
+ <&dma 3>;
+ dma-names = "tx", "rx";
+};
diff --git a/dts/Bindings/sound/cs4270.txt b/dts/Bindings/sound/cs4270.txt
new file mode 100644
index 0000000..6b222f9
--- /dev/null
+++ b/dts/Bindings/sound/cs4270.txt
@@ -0,0 +1,21 @@
+CS4270 audio CODEC
+
+The driver for this device currently only supports I2C.
+
+Required properties:
+
+ - compatible : "cirrus,cs4270"
+
+ - reg : the I2C address of the device for I2C
+
+Optional properties:
+
+ - reset-gpio : a GPIO spec for the reset pin. If specified, it will be
+ deasserted before communication to the codec starts.
+
+Example:
+
+codec: cs4270@48 {
+ compatible = "cirrus,cs4270";
+ reg = <0x48>;
+};
diff --git a/dts/Bindings/sound/cs4271.txt b/dts/Bindings/sound/cs4271.txt
new file mode 100644
index 0000000..e2cd1d7
--- /dev/null
+++ b/dts/Bindings/sound/cs4271.txt
@@ -0,0 +1,50 @@
+Cirrus Logic CS4271 DT bindings
+
+This driver supports both the I2C and the SPI bus.
+
+Required properties:
+
+ - compatible: "cirrus,cs4271"
+
+For required properties on SPI, please consult
+Documentation/devicetree/bindings/spi/spi-bus.txt
+
+Required properties on I2C:
+
+ - reg: the i2c address
+
+
+Optional properties:
+
+ - reset-gpio: a GPIO spec to define which pin is connected to the chip's
+ !RESET pin
+ - cirrus,amuteb-eq-bmutec: When given, the Codec's AMUTEB=BMUTEC flag
+ is enabled.
+ - cirrus,enable-soft-reset:
+ The CS4271 requires its LRCLK and MCLK to be stable before its RESET
+ line is de-asserted. That also means that clocks cannot be changed
+ without putting the chip back into hardware reset, which also requires
+ a complete re-initialization of all registers.
+
+ One (undocumented) workaround is to assert and de-assert the PDN bit
+ in the MODE2 register. This workaround can be enabled with this DT
+ property.
+
+ Note that this is not needed in case the clocks are stable
+ throughout the entire runtime of the codec.
+
+Examples:
+
+ codec_i2c: cs4271@10 {
+ compatible = "cirrus,cs4271";
+ reg = <0x10>;
+ reset-gpio = <&gpio 23 0>;
+ };
+
+ codec_spi: cs4271@0 {
+ compatible = "cirrus,cs4271";
+ reg = <0x0>;
+ reset-gpio = <&gpio 23 0>;
+ spi-max-frequency = <6000000>;
+ };
+
diff --git a/dts/Bindings/sound/cs42l52.txt b/dts/Bindings/sound/cs42l52.txt
new file mode 100644
index 0000000..bc03c93
--- /dev/null
+++ b/dts/Bindings/sound/cs42l52.txt
@@ -0,0 +1,46 @@
+CS42L52 audio CODEC
+
+Required properties:
+
+ - compatible : "cirrus,cs42l52"
+
+ - reg : the I2C address of the device for I2C
+
+Optional properties:
+
+ - cirrus,reset-gpio : GPIO controller's phandle and the number
+ of the GPIO used to reset the codec.
+
+ - cirrus,chgfreq-divisor : Values used to set the Charge Pump Frequency.
+ Allowable values of 0x00 through 0x0F. These are raw values written to the
+ register, not the actual frequency. The frequency is determined by the following.
+ Frequency = (64xFs)/(N+2)
+ N = chgfreq_val
+ Fs = Sample Rate (variable)
+
+ - cirrus,mica-differential-cfg : boolean, If present, then the MICA input is configured
+ as a differential input. If not present then the MICA input is configured as
+ Single-ended input. Single-ended mode allows for MIC1 or MIC2 muxing for input.
+
+ - cirrus,micb-differential-cfg : boolean, If present, then the MICB input is configured
+ as a differential input. If not present then the MICB input is configured as
+ Single-ended input. Single-ended mode allows for MIC1 or MIC2 muxing for input.
+
+ - cirrus,micbias-lvl: Set the output voltage level on the MICBIAS Pin
+ 0 = 0.5 x VA
+ 1 = 0.6 x VA
+ 2 = 0.7 x VA
+ 3 = 0.8 x VA
+ 4 = 0.83 x VA
+ 5 = 0.91 x VA
+
+Example:
+
+codec: codec@4a {
+ compatible = "cirrus,cs42l52";
+ reg = <0x4a>;
+ reset-gpio = <&gpio 10 0>;
+ cirrus,chgfreq-divisor = <0x05>;
+ cirrus.mica-differential-cfg;
+ cirrus,micbias-lvl = <5>;
+};
diff --git a/dts/Bindings/sound/cs42l73.txt b/dts/Bindings/sound/cs42l73.txt
new file mode 100644
index 0000000..80ae910
--- /dev/null
+++ b/dts/Bindings/sound/cs42l73.txt
@@ -0,0 +1,22 @@
+CS42L73 audio CODEC
+
+Required properties:
+
+ - compatible : "cirrus,cs42l73"
+
+ - reg : the I2C address of the device for I2C
+
+Optional properties:
+
+ - reset_gpio : a GPIO spec for the reset pin.
+ - chgfreq : Charge Pump Frequency values 0x00-0x0F
+
+
+Example:
+
+codec: cs42l73@4a {
+ compatible = "cirrus,cs42l73";
+ reg = <0x4a>;
+ reset_gpio = <&gpio 10 0>;
+ chgfreq = <0x05>;
+}; \ No newline at end of file
diff --git a/dts/Bindings/sound/cs42xx8.txt b/dts/Bindings/sound/cs42xx8.txt
new file mode 100644
index 0000000..f631fbc
--- /dev/null
+++ b/dts/Bindings/sound/cs42xx8.txt
@@ -0,0 +1,28 @@
+CS42448/CS42888 audio CODEC
+
+Required properties:
+
+ - compatible : must contain one of "cirrus,cs42448" and "cirrus,cs42888"
+
+ - reg : the I2C address of the device for I2C
+
+ - clocks : a list of phandles + clock-specifiers, one for each entry in
+ clock-names
+
+ - clock-names : must contain "mclk"
+
+ - VA-supply, VD-supply, VLS-supply, VLC-supply: power supplies for the device,
+ as covered in Documentation/devicetree/bindings/regulator/regulator.txt
+
+Example:
+
+codec: cs42888@48 {
+ compatible = "cirrus,cs42888";
+ reg = <0x48>;
+ clocks = <&codec_mclk 0>;
+ clock-names = "mclk";
+ VA-supply = <&reg_audio>;
+ VD-supply = <&reg_audio>;
+ VLS-supply = <&reg_audio>;
+ VLC-supply = <&reg_audio>;
+};
diff --git a/dts/Bindings/sound/da9055.txt b/dts/Bindings/sound/da9055.txt
new file mode 100644
index 0000000..ed1b7cc
--- /dev/null
+++ b/dts/Bindings/sound/da9055.txt
@@ -0,0 +1,22 @@
+* Dialog DA9055 Audio CODEC
+
+DA9055 provides Audio CODEC support (I2C only).
+
+The Audio CODEC device in DA9055 has it's own I2C address which is configurable,
+so the device is instantiated separately from the PMIC (MFD) device.
+
+For details on accompanying PMIC I2C device, see the following:
+Documentation/devicetree/bindings/mfd/da9055.txt
+
+Required properties:
+
+ - compatible: "dlg,da9055-codec"
+ - reg: Specifies the I2C slave address
+
+
+Example:
+
+ codec: da9055-codec@1a {
+ compatible = "dlg,da9055-codec";
+ reg = <0x1a>;
+ };
diff --git a/dts/Bindings/sound/davinci-evm-audio.txt b/dts/Bindings/sound/davinci-evm-audio.txt
new file mode 100644
index 0000000..963e100
--- /dev/null
+++ b/dts/Bindings/sound/davinci-evm-audio.txt
@@ -0,0 +1,49 @@
+* Texas Instruments SoC audio setups with TLV320AIC3X Codec
+
+Required properties:
+- compatible : "ti,da830-evm-audio" : forDM365/DA8xx/OMAPL1x/AM33xx
+- ti,model : The user-visible name of this sound complex.
+- ti,audio-codec : The phandle of the TLV320AIC3x audio codec
+- ti,mcasp-controller : The phandle of the McASP controller
+- ti,audio-routing : 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 for sources and
+ sinks are the codec's pins, and the jacks on the board:
+
+Optional properties:
+- ti,codec-clock-rate : The Codec Clock rate (in Hz) applied to the Codec.
+- clocks : Reference to the master clock
+- clock-names : The clock should be named "mclk"
+- Either codec-clock-rate or the codec-clock reference has to be defined. If
+ the both are defined the driver attempts to set referenced clock to the
+ defined rate and takes the rate from the clock reference.
+
+ Board connectors:
+
+ * Headphone Jack
+ * Line Out
+ * Mic Jack
+ * Line In
+
+
+Example:
+
+sound {
+ compatible = "ti,da830-evm-audio";
+ ti,model = "DA830 EVM";
+ ti,audio-codec = <&tlv320aic3x>;
+ ti,mcasp-controller = <&mcasp1>;
+ ti,codec-clock-rate = <12000000>;
+ ti,audio-routing =
+ "Headphone Jack", "HPLOUT",
+ "Headphone Jack", "HPROUT",
+ "Line Out", "LLOUT",
+ "Line Out", "RLOUT",
+ "MIC3L", "Mic Bias 2V",
+ "MIC3R", "Mic Bias 2V",
+ "Mic Bias 2V", "Mic Jack",
+ "LINE1L", "Line In",
+ "LINE2L", "Line In",
+ "LINE1R", "Line In",
+ "LINE2R", "Line In";
+};
diff --git a/dts/Bindings/sound/davinci-mcasp-audio.txt b/dts/Bindings/sound/davinci-mcasp-audio.txt
new file mode 100644
index 0000000..569b26c
--- /dev/null
+++ b/dts/Bindings/sound/davinci-mcasp-audio.txt
@@ -0,0 +1,60 @@
+Texas Instruments McASP controller
+
+Required properties:
+- compatible :
+ "ti,dm646x-mcasp-audio" : for DM646x platforms
+ "ti,da830-mcasp-audio" : for both DA830 & DA850 platforms
+ "ti,am33xx-mcasp-audio" : for AM33xx platforms (AM33xx, AM43xx, TI81xx)
+ "ti,dra7-mcasp-audio" : for DRA7xx platforms
+
+- reg : Should contain reg specifiers for the entries in the reg-names property.
+- reg-names : Should contain:
+ * "mpu" for the main registers (required). For compatibility with
+ existing software, it is recommended this is the first entry.
+ * "dat" for separate data port register access (optional).
+- op-mode : I2S/DIT ops mode. 0 for I2S mode. 1 for DIT mode used for S/PDIF,
+ IEC60958-1, and AES-3 formats.
+- tdm-slots : Slots for TDM operation. Indicates number of channels transmitted
+ or received over one serializer.
+- serial-dir : A list of serializer configuration. Each entry is a number
+ indication for serializer pin direction.
+ (0 - INACTIVE, 1 - TX, 2 - RX)
+- dmas: two element list of DMA controller phandles and DMA request line
+ ordered pairs.
+- dma-names: identifier string for each DMA request line in the dmas property.
+ These strings correspond 1:1 with the ordered pairs in dmas. The dma
+ identifiers must be "rx" and "tx".
+
+Optional properties:
+
+- ti,hwmods : Must be "mcasp<n>", n is controller instance starting 0
+- tx-num-evt : FIFO levels.
+- rx-num-evt : FIFO levels.
+- 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, currently not used by the driver
+- interrupt-names : Known interrupt names are "tx" and "rx"
+- pinctrl-0: Should specify pin control group used for this controller.
+- pinctrl-names: Should contain only one value - "default", for more details
+ please refer to pinctrl-bindings.txt
+- fck_parent : Should contain a valid clock name which will be used as parent
+ for the McASP fck
+
+Example:
+
+mcasp0: mcasp0@1d00000 {
+ compatible = "ti,da830-mcasp-audio";
+ reg = <0x100000 0x3000>;
+ reg-names "mpu";
+ interrupts = <82>, <83>;
+ interrupts-names = "tx", "rx";
+ op-mode = <0>; /* MCASP_IIS_MODE */
+ tdm-slots = <2>;
+ serial-dir = <
+ 0 0 0 0 /* 0: INACTIVE, 1: TX, 2: RX */
+ 0 0 0 0
+ 0 0 0 1
+ 2 0 0 0 >;
+ tx-num-evt = <1>;
+ rx-num-evt = <1>;
+};
diff --git a/dts/Bindings/sound/eukrea-tlv320.txt b/dts/Bindings/sound/eukrea-tlv320.txt
new file mode 100644
index 0000000..0d7985c
--- /dev/null
+++ b/dts/Bindings/sound/eukrea-tlv320.txt
@@ -0,0 +1,21 @@
+Audio complex for Eukrea boards with tlv320aic23 codec.
+
+Required properties:
+- compatible : "eukrea,asoc-tlv320"
+- eukrea,model : The user-visible name of this sound complex.
+- ssi-controller : The phandle of the SSI controller.
+- fsl,mux-int-port : The internal port of the i.MX audio muxer (AUDMUX).
+- fsl,mux-ext-port : The external port of the i.MX audio muxer.
+
+Note: The AUDMUX port numbering should start at 1, which is consistent with
+hardware manual.
+
+Example:
+
+ sound {
+ compatible = "eukrea,asoc-tlv320";
+ eukrea,model = "imx51-eukrea-tlv320aic23";
+ ssi-controller = <&ssi2>;
+ fsl,mux-int-port = <2>;
+ fsl,mux-ext-port = <3>;
+ };
diff --git a/dts/Bindings/sound/fsl,esai.txt b/dts/Bindings/sound/fsl,esai.txt
new file mode 100644
index 0000000..aeb8c4a
--- /dev/null
+++ b/dts/Bindings/sound/fsl,esai.txt
@@ -0,0 +1,55 @@
+Freescale Enhanced Serial Audio Interface (ESAI) Controller
+
+The Enhanced Serial Audio Interface (ESAI) provides a full-duplex serial port
+for serial communication with a variety of serial devices, including industry
+standard codecs, Sony/Phillips Digital Interface (S/PDIF) transceivers, and
+other DSPs. It has up to six transmitters and four receivers.
+
+Required properties:
+
+ - compatible : Compatible list, must contain "fsl,imx35-esai".
+
+ - reg : Offset and length of the register set for the device.
+
+ - interrupts : Contains the spdif interrupt.
+
+ - dmas : Generic dma devicetree binding as described in
+ Documentation/devicetree/bindings/dma/dma.txt.
+
+ - dma-names : Two dmas have to be defined, "tx" and "rx".
+
+ - clocks: Contains an entry for each entry in clock-names.
+
+ - clock-names : Includes the following entries:
+ "core" The core clock used to access registers
+ "extal" The esai baud clock for esai controller used to derive
+ HCK, SCK and FS.
+ "fsys" The system clock derived from ahb clock used to derive
+ HCK, SCK and FS.
+
+ - fsl,fifo-depth: The number of elements in the transmit and receive FIFOs.
+ This number is the maximum allowed value for TFCR[TFWM] or RFCR[RFWM].
+
+ - fsl,esai-synchronous: This is a boolean property. If present, indicating
+ that ESAI would work in the synchronous mode, which means all the settings
+ for Receiving would be duplicated from Transmition related registers.
+
+ - 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.
+
+Example:
+
+esai: esai@02024000 {
+ compatible = "fsl,imx35-esai";
+ reg = <0x02024000 0x4000>;
+ interrupts = <0 51 0x04>;
+ clocks = <&clks 208>, <&clks 118>, <&clks 208>;
+ clock-names = "core", "extal", "fsys";
+ dmas = <&sdma 23 21 0>, <&sdma 24 21 0>;
+ dma-names = "rx", "tx";
+ fsl,fifo-depth = <128>;
+ fsl,esai-synchronous;
+ big-endian;
+ status = "disabled";
+};
diff --git a/dts/Bindings/sound/fsl,spdif.txt b/dts/Bindings/sound/fsl,spdif.txt
new file mode 100644
index 0000000..3e9e82c
--- /dev/null
+++ b/dts/Bindings/sound/fsl,spdif.txt
@@ -0,0 +1,59 @@
+Freescale Sony/Philips Digital Interface Format (S/PDIF) Controller
+
+The Freescale S/PDIF audio block is a stereo transceiver that allows the
+processor to receive and transmit digital audio via an coaxial cable or
+a fibre cable.
+
+Required properties:
+
+ - compatible : Compatible list, must contain "fsl,imx35-spdif".
+
+ - reg : Offset and length of the register set for the device.
+
+ - interrupts : Contains the spdif interrupt.
+
+ - dmas : Generic dma devicetree binding as described in
+ Documentation/devicetree/bindings/dma/dma.txt.
+
+ - dma-names : Two dmas have to be defined, "tx" and "rx".
+
+ - clocks : Contains an entry for each entry in clock-names.
+
+ - clock-names : Includes the following entries:
+ "core" The core clock of spdif controller
+ "rxtx<0-7>" Clock source list for tx and rx clock.
+ This clock list should be identical to
+ the source list connecting to the spdif
+ clock mux in "SPDIF Transceiver Clock
+ Diagram" of SoC reference manual. It
+ can also be referred to TxClk_Source
+ bit of register SPDIF_STC.
+
+ - 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.
+
+Example:
+
+spdif: spdif@02004000 {
+ compatible = "fsl,imx35-spdif";
+ reg = <0x02004000 0x4000>;
+ interrupts = <0 52 0x04>;
+ dmas = <&sdma 14 18 0>,
+ <&sdma 15 18 0>;
+ dma-names = "rx", "tx";
+
+ clocks = <&clks 197>, <&clks 3>,
+ <&clks 197>, <&clks 107>,
+ <&clks 0>, <&clks 118>,
+ <&clks 62>, <&clks 139>,
+ <&clks 0>;
+ clock-names = "core", "rxtx0",
+ "rxtx1", "rxtx2",
+ "rxtx3", "rxtx4",
+ "rxtx5", "rxtx6",
+ "rxtx7";
+
+ big-endian;
+ status = "okay";
+};
diff --git a/dts/Bindings/sound/fsl,ssi.txt b/dts/Bindings/sound/fsl,ssi.txt
new file mode 100644
index 0000000..3aa4a8f
--- /dev/null
+++ b/dts/Bindings/sound/fsl,ssi.txt
@@ -0,0 +1,93 @@
+Freescale Synchronous Serial Interface
+
+The SSI is a serial device that communicates with audio codecs. It can
+be programmed in AC97, I2S, left-justified, or right-justified modes.
+
+Required properties:
+- compatible: Compatible list, should contain one of the following
+ compatibles:
+ fsl,mpc8610-ssi
+ fsl,imx51-ssi
+ fsl,imx35-ssi
+ fsl,imx21-ssi
+- cell-index: The SSI, <0> = SSI1, <1> = SSI2, and so on.
+- reg: Offset and length of the register set for the device.
+- interrupts: <a b> where a is the interrupt number and b is a
+ field that represents an encoding of the sense and
+ level information for the interrupt. This should be
+ encoded based on the information in section 2)
+ depending on the type of interrupt controller you
+ have.
+- interrupt-parent: The phandle for the interrupt controller that
+ services interrupts for this device.
+- fsl,playback-dma: Phandle to a node for the DMA channel to use for
+ playback of audio. This is typically dictated by SOC
+ design. See the notes below.
+- fsl,capture-dma: Phandle to a node for the DMA channel to use for
+ capture (recording) of audio. This is typically dictated
+ by SOC design. See the notes below.
+- fsl,fifo-depth: The number of elements in the transmit and receive FIFOs.
+ This number is the maximum allowed value for SFCSR[TFWM0].
+- fsl,ssi-asynchronous:
+ If specified, the SSI is to be programmed in asynchronous
+ mode. In this mode, pins SRCK, STCK, SRFS, and STFS must
+ all be connected to valid signals. In synchronous mode,
+ SRCK and SRFS are ignored. Asynchronous mode allows
+ playback and capture to use different sample sizes and
+ sample rates. Some drivers may require that SRCK and STCK
+ be connected together, and SRFS and STFS be connected
+ together. This would still allow different sample sizes,
+ but not different sample rates.
+ - clocks: "ipg" - Required clock for the SSI unit
+ "baud" - Required clock for SSI master mode. Otherwise this
+ clock is not used
+
+Required are also ac97 link bindings if ac97 is used. See
+Documentation/devicetree/bindings/sound/soc-ac97link.txt for the necessary
+bindings.
+
+Optional properties:
+- codec-handle: Phandle to a 'codec' node that defines an audio
+ codec connected to this SSI. This node is typically
+ a child of an I2C or other control node.
+- fsl,fiq-stream-filter: Bool property. Disabled DMA and use FIQ instead to
+ filter the codec stream. This is necessary for some boards
+ where an incompatible codec is connected to this SSI, e.g.
+ on pca100 and pcm043.
+- dmas: Generic dma devicetree binding as described in
+ Documentation/devicetree/bindings/dma/dma.txt.
+- dma-names: Two dmas have to be defined, "tx" and "rx", if fsl,imx-fiq
+ is not defined.
+- fsl,mode: The operating mode for the SSI interface.
+ "i2s-slave" - I2S mode, SSI is clock slave
+ "i2s-master" - I2S mode, SSI is clock master
+ "lj-slave" - left-justified mode, SSI is clock slave
+ "lj-master" - l.j. mode, SSI is clock master
+ "rj-slave" - right-justified mode, SSI is clock slave
+ "rj-master" - r.j., SSI is clock master
+ "ac97-slave" - AC97 mode, SSI is clock slave
+ "ac97-master" - AC97 mode, SSI is clock master
+
+Child 'codec' node required properties:
+- compatible: Compatible list, contains the name of the codec
+
+Child 'codec' node optional properties:
+- clock-frequency: The frequency of the input clock, which typically comes
+ from an on-board dedicated oscillator.
+
+Notes on fsl,playback-dma and fsl,capture-dma:
+
+On SOCs that have an SSI, specific DMA channels are hard-wired for playback
+and capture. On the MPC8610, for example, SSI1 must use DMA channel 0 for
+playback and DMA channel 1 for capture. SSI2 must use DMA channel 2 for
+playback and DMA channel 3 for capture. The developer can choose which
+DMA controller to use, but the channels themselves are hard-wired. The
+purpose of these two properties is to represent this hardware design.
+
+The device tree nodes for the DMA channels that are referenced by
+"fsl,playback-dma" and "fsl,capture-dma" must be marked as compatible with
+"fsl,ssi-dma-channel". The SOC-specific compatible string (e.g.
+"fsl,mpc8610-dma-channel") can remain. If these nodes are left as
+"fsl,elo-dma-channel" or "fsl,eloplus-dma-channel", then the generic Elo DMA
+drivers (fsldma) will attempt to use them, and it will conflict with the
+sound drivers.
diff --git a/dts/Bindings/sound/fsl-sai.txt b/dts/Bindings/sound/fsl-sai.txt
new file mode 100644
index 0000000..98611a6
--- /dev/null
+++ b/dts/Bindings/sound/fsl-sai.txt
@@ -0,0 +1,40 @@
+Freescale Synchronous Audio Interface (SAI).
+
+The SAI is based on I2S module that used communicating with audio codecs,
+which provides a synchronous audio interface that supports fullduplex
+serial interfaces with frame synchronization such as I2S, AC97, TDM, and
+codec/DSP interfaces.
+
+
+Required properties:
+- compatible: Compatible list, contains "fsl,vf610-sai".
+- reg: Offset and length of the register set for the device.
+- clocks: Must contain an entry for each entry in clock-names.
+- clock-names : Must include the "sai" entry.
+- dmas : Generic dma devicetree binding as described in
+ Documentation/devicetree/bindings/dma/dma.txt.
+- dma-names : Two dmas have to be defined, "tx" and "rx".
+- pinctrl-names: Must contain a "default" entry.
+- pinctrl-NNN: One property must exist for each entry in pinctrl-names.
+ See ../pinctrl/pinctrl-bindings.txt for details of the property values.
+- big-endian-regs: If this property is absent, the little endian mode will
+ be in use as default, or the big endian mode will be in use for all the
+ device registers.
+- big-endian-data: If this property is absent, the little endian mode will
+ be in use as default, or the big endian mode will be in use for all the
+ fifo data.
+
+Example:
+sai2: sai@40031000 {
+ compatible = "fsl,vf610-sai";
+ reg = <0x40031000 0x1000>;
+ pinctrl-names = "default";
+ pinctrl-0 = <&pinctrl_sai2_1>;
+ clocks = <&clks VF610_CLK_SAI2>;
+ clock-names = "sai";
+ dma-names = "tx", "rx";
+ dmas = <&edma0 0 VF610_EDMA_MUXID0_SAI2_TX>,
+ <&edma0 0 VF610_EDMA_MUXID0_SAI2_RX>;
+ big-endian-regs;
+ big-endian-data;
+};
diff --git a/dts/Bindings/sound/hdmi.txt b/dts/Bindings/sound/hdmi.txt
new file mode 100644
index 0000000..31af7bc
--- /dev/null
+++ b/dts/Bindings/sound/hdmi.txt
@@ -0,0 +1,17 @@
+Device-Tree bindings for dummy HDMI codec
+
+Required properties:
+ - compatible: should be "linux,hdmi-audio".
+
+CODEC output pins:
+ * TX
+
+CODEC input pins:
+ * RX
+
+Example node:
+
+ hdmi_audio: hdmi_audio@0 {
+ compatible = "linux,hdmi-audio";
+ status = "okay";
+ };
diff --git a/dts/Bindings/sound/imx-audio-sgtl5000.txt b/dts/Bindings/sound/imx-audio-sgtl5000.txt
new file mode 100644
index 0000000..e4acdd8
--- /dev/null
+++ b/dts/Bindings/sound/imx-audio-sgtl5000.txt
@@ -0,0 +1,49 @@
+Freescale i.MX audio complex with SGTL5000 codec
+
+Required properties:
+- compatible : "fsl,imx-audio-sgtl5000"
+- model : The user-visible name of this sound complex
+- ssi-controller : The phandle of the i.MX SSI controller
+- audio-codec : The phandle of the SGTL5000 audio codec
+- audio-routing : 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, SGTL5000 pins, and the jacks on the board:
+
+ Power supplies:
+ * Mic Bias
+
+ SGTL5000 pins:
+ * MIC_IN
+ * LINE_IN
+ * HP_OUT
+ * LINE_OUT
+
+ Board connectors:
+ * Mic Jack
+ * Line In Jack
+ * Headphone Jack
+ * Line Out Jack
+ * Ext Spk
+
+- mux-int-port : The internal port of the i.MX audio muxer (AUDMUX)
+- mux-ext-port : The external port of the i.MX audio muxer
+
+Note: The AUDMUX port numbering should start at 1, which is consistent with
+hardware manual.
+
+Example:
+
+sound {
+ compatible = "fsl,imx51-babbage-sgtl5000",
+ "fsl,imx-audio-sgtl5000";
+ model = "imx51-babbage-sgtl5000";
+ ssi-controller = <&ssi1>;
+ audio-codec = <&sgtl5000>;
+ audio-routing =
+ "MIC_IN", "Mic Jack",
+ "Mic Jack", "Mic Bias",
+ "Headphone Jack", "HP_OUT";
+ mux-int-port = <1>;
+ mux-ext-port = <3>;
+};
diff --git a/dts/Bindings/sound/imx-audio-spdif.txt b/dts/Bindings/sound/imx-audio-spdif.txt
new file mode 100644
index 0000000..7d13479
--- /dev/null
+++ b/dts/Bindings/sound/imx-audio-spdif.txt
@@ -0,0 +1,34 @@
+Freescale i.MX audio complex with S/PDIF transceiver
+
+Required properties:
+
+ - compatible : "fsl,imx-audio-spdif"
+
+ - model : The user-visible name of this sound complex
+
+ - spdif-controller : The phandle of the i.MX S/PDIF controller
+
+
+Optional properties:
+
+ - spdif-out : This is a boolean property. If present, the transmitting
+ function of S/PDIF will be enabled, indicating there's a physical
+ S/PDIF out connector/jack on the board or it's connecting to some
+ other IP block, such as an HDMI encoder/display-controller.
+
+ - spdif-in : This is a boolean property. If present, the receiving
+ function of S/PDIF will be enabled, indicating there's a physical
+ S/PDIF in connector/jack on the board.
+
+* Note: At least one of these two properties should be set in the DT binding.
+
+
+Example:
+
+sound-spdif {
+ compatible = "fsl,imx-audio-spdif";
+ model = "imx-spdif";
+ spdif-controller = <&spdif>;
+ spdif-out;
+ spdif-in;
+};
diff --git a/dts/Bindings/sound/imx-audio-wm8962.txt b/dts/Bindings/sound/imx-audio-wm8962.txt
new file mode 100644
index 0000000..f49450a
--- /dev/null
+++ b/dts/Bindings/sound/imx-audio-wm8962.txt
@@ -0,0 +1,46 @@
+Freescale i.MX audio complex with WM8962 codec
+
+Required properties:
+- compatible : "fsl,imx-audio-wm8962"
+- model : The user-visible name of this sound complex
+- ssi-controller : The phandle of the i.MX SSI controller
+- audio-codec : The phandle of the WM8962 audio codec
+- audio-routing : 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, WM8962 pins, and the jacks on the board:
+
+ Power supplies:
+ * Mic Bias
+
+ Board connectors:
+ * Mic Jack
+ * Headphone Jack
+ * Ext Spk
+
+- mux-int-port : The internal port of the i.MX audio muxer (AUDMUX)
+- mux-ext-port : The external port of the i.MX audio muxer
+
+Note: The AUDMUX port numbering should start at 1, which is consistent with
+hardware manual.
+
+Example:
+
+sound {
+ compatible = "fsl,imx6q-sabresd-wm8962",
+ "fsl,imx-audio-wm8962";
+ model = "wm8962-audio";
+ ssi-controller = <&ssi2>;
+ audio-codec = <&codec>;
+ audio-routing =
+ "Headphone Jack", "HPOUTL",
+ "Headphone Jack", "HPOUTR",
+ "Ext Spk", "SPKOUTL",
+ "Ext Spk", "SPKOUTR",
+ "MICBIAS", "AMIC",
+ "IN3R", "MICBIAS",
+ "DMIC", "MICBIAS",
+ "DMICDAT", "DMIC";
+ mux-int-port = <2>;
+ mux-ext-port = <3>;
+};
diff --git a/dts/Bindings/sound/imx-audmux.txt b/dts/Bindings/sound/imx-audmux.txt
new file mode 100644
index 0000000..f88a00e
--- /dev/null
+++ b/dts/Bindings/sound/imx-audmux.txt
@@ -0,0 +1,22 @@
+Freescale Digital Audio Mux (AUDMUX) device
+
+Required properties:
+- compatible : "fsl,imx21-audmux" for AUDMUX version firstly used on i.MX21,
+ or "fsl,imx31-audmux" for the version firstly used on i.MX31.
+- reg : Should contain AUDMUX registers location and length
+
+An initial configuration can be setup using child nodes.
+
+Required properties of optional child nodes:
+- fsl,audmux-port : Integer of the audmux port that is configured by this
+ child node.
+- fsl,port-config : List of configuration options for the specific port. For
+ imx31-audmux and above, it is a list of tuples <ptcr pdcr>. For
+ imx21-audmux it is a list of pcr values.
+
+Example:
+
+audmux@021d8000 {
+ compatible = "fsl,imx6q-audmux", "fsl,imx31-audmux";
+ reg = <0x021d8000 0x4000>;
+};
diff --git a/dts/Bindings/sound/max98090.txt b/dts/Bindings/sound/max98090.txt
new file mode 100644
index 0000000..e4c8b36
--- /dev/null
+++ b/dts/Bindings/sound/max98090.txt
@@ -0,0 +1,43 @@
+MAX98090 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+- compatible : "maxim,max98090".
+
+- reg : The I2C address of the device.
+
+- interrupts : The CODEC's interrupt output.
+
+Pins on the device (for linking into audio routes):
+
+ * MIC1
+ * MIC2
+ * DMICL
+ * DMICR
+ * IN1
+ * IN2
+ * IN3
+ * IN4
+ * IN5
+ * IN6
+ * IN12
+ * IN34
+ * IN56
+ * HPL
+ * HPR
+ * SPKL
+ * SPKR
+ * RCVL
+ * RCVR
+ * MICBIAS
+
+Example:
+
+audio-codec@10 {
+ compatible = "maxim,max98090";
+ reg = <0x10>;
+ interrupt-parent = <&gpio>;
+ interrupts = <TEGRA_GPIO(H, 4) GPIO_ACTIVE_HIGH>;
+};
diff --git a/dts/Bindings/sound/mrvl,pxa-ssp.txt b/dts/Bindings/sound/mrvl,pxa-ssp.txt
new file mode 100644
index 0000000..74c9ba6
--- /dev/null
+++ b/dts/Bindings/sound/mrvl,pxa-ssp.txt
@@ -0,0 +1,28 @@
+Marvell PXA SSP CPU DAI bindings
+
+Required properties:
+
+ compatible Must be "mrvl,pxa-ssp-dai"
+ port A phandle reference to a PXA ssp upstream device
+
+Example:
+
+ /* upstream device */
+
+ ssp0: ssp@41000000 {
+ compatible = "mrvl,pxa3xx-ssp";
+ reg = <0x41000000 0x40>;
+ interrupts = <24>;
+ clock-names = "pxa27x-ssp.0";
+ dmas = <&dma 13
+ &dma 14>;
+ dma-names = "rx", "tx";
+ };
+
+ /* DAI as user */
+
+ ssp_dai0: ssp_dai@0 {
+ compatible = "mrvl,pxa-ssp-dai";
+ port = <&ssp0>;
+ };
+
diff --git a/dts/Bindings/sound/mrvl,pxa2xx-pcm.txt b/dts/Bindings/sound/mrvl,pxa2xx-pcm.txt
new file mode 100644
index 0000000..551fbb8
--- /dev/null
+++ b/dts/Bindings/sound/mrvl,pxa2xx-pcm.txt
@@ -0,0 +1,15 @@
+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";
+ };
+
diff --git a/dts/Bindings/sound/mvebu-audio.txt b/dts/Bindings/sound/mvebu-audio.txt
new file mode 100644
index 0000000..cb8c07c
--- /dev/null
+++ b/dts/Bindings/sound/mvebu-audio.txt
@@ -0,0 +1,34 @@
+* mvebu (Kirkwood, Dove, Armada 370) audio controller
+
+Required properties:
+
+- compatible:
+ "marvell,kirkwood-audio" for Kirkwood platforms
+ "marvell,dove-audio" for Dove platforms
+ "marvell,armada370-audio" for Armada 370 platforms
+
+- reg: physical base address of the controller and length of memory mapped
+ region.
+
+- interrupts:
+ with "marvell,kirkwood-audio", the audio interrupt
+ with "marvell,dove-audio", a list of two interrupts, the first for
+ the data flow, and the second for errors.
+
+- clocks: one or two phandles.
+ The first one is mandatory and defines the internal clock.
+ The second one is optional and defines an external clock.
+
+- clock-names: names associated to the clocks:
+ "internal" for the internal clock
+ "extclk" for the external clock
+
+Example:
+
+i2s1: audio-controller@b4000 {
+ compatible = "marvell,dove-audio";
+ reg = <0xb4000 0x2210>;
+ interrupts = <21>, <22>;
+ clocks = <&gate_clk 13>;
+ clock-names = "internal";
+};
diff --git a/dts/Bindings/sound/mxs-audio-sgtl5000.txt b/dts/Bindings/sound/mxs-audio-sgtl5000.txt
new file mode 100644
index 0000000..601c518
--- /dev/null
+++ b/dts/Bindings/sound/mxs-audio-sgtl5000.txt
@@ -0,0 +1,17 @@
+* Freescale MXS audio complex with SGTL5000 codec
+
+Required properties:
+- compatible: "fsl,mxs-audio-sgtl5000"
+- model: The user-visible name of this sound complex
+- saif-controllers: The phandle list of the MXS SAIF controller
+- audio-codec: The phandle of the SGTL5000 audio codec
+
+Example:
+
+sound {
+ compatible = "fsl,imx28-evk-sgtl5000",
+ "fsl,mxs-audio-sgtl5000";
+ model = "imx28-evk-sgtl5000";
+ saif-controllers = <&saif0 &saif1>;
+ audio-codec = <&sgtl5000>;
+};
diff --git a/dts/Bindings/sound/mxs-saif.txt b/dts/Bindings/sound/mxs-saif.txt
new file mode 100644
index 0000000..7ba07a1
--- /dev/null
+++ b/dts/Bindings/sound/mxs-saif.txt
@@ -0,0 +1,41 @@
+* Freescale MXS Serial Audio Interface (SAIF)
+
+Required properties:
+- compatible: Should be "fsl,<chip>-saif"
+- reg: Should contain registers location and length
+- interrupts: Should contain ERROR interrupt number
+- dmas: DMA specifier, consisting of a phandle to DMA controller node
+ and SAIF DMA channel ID.
+ Refer to dma.txt and fsl-mxs-dma.txt for details.
+- dma-names: Must be "rx-tx".
+
+Optional properties:
+- fsl,saif-master: phandle to the master SAIF. It's only required for
+ the slave SAIF.
+
+Note: Each SAIF controller should have an alias correctly numbered
+in "aliases" node.
+
+Example:
+
+aliases {
+ saif0 = &saif0;
+ saif1 = &saif1;
+};
+
+saif0: saif@80042000 {
+ compatible = "fsl,imx28-saif";
+ reg = <0x80042000 2000>;
+ interrupts = <59>;
+ dmas = <&dma_apbx 4>;
+ dma-names = "rx-tx";
+};
+
+saif1: saif@80046000 {
+ compatible = "fsl,imx28-saif";
+ reg = <0x80046000 2000>;
+ interrupts = <58>;
+ dmas = <&dma_apbx 5>;
+ dma-names = "rx-tx";
+ fsl,saif-master = <&saif0>;
+};
diff --git a/dts/Bindings/sound/nvidia,tegra-audio-alc5632.txt b/dts/Bindings/sound/nvidia,tegra-audio-alc5632.txt
new file mode 100644
index 0000000..57f40f9
--- /dev/null
+++ b/dts/Bindings/sound/nvidia,tegra-audio-alc5632.txt
@@ -0,0 +1,48 @@
+NVIDIA Tegra audio complex
+
+Required properties:
+- compatible : "nvidia,tegra-audio-alc5632"
+- clocks : Must contain an entry for each entry in clock-names.
+ See ../clocks/clock-bindings.txt for details.
+- clock-names : Must include the following entries:
+ - pll_a
+ - pll_a_out0
+ - mclk (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk)
+- nvidia,model : The user-visible name of this sound complex.
+- nvidia,audio-routing : 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 for sources and
+ sinks are the ALC5632's pins as documented in the binding for the device
+ and:
+
+ * Headset Stereophone
+ * Int Spk
+ * Headset Mic
+ * Digital Mic
+
+- nvidia,i2s-controller : The phandle of the Tegra I2S controller
+- nvidia,audio-codec : The phandle of the ALC5632 audio codec
+
+Example:
+
+sound {
+ compatible = "nvidia,tegra-audio-alc5632-paz00",
+ "nvidia,tegra-audio-alc5632";
+
+ nvidia,model = "Compal PAZ00";
+
+ nvidia,audio-routing =
+ "Int Spk", "SPK_OUTP",
+ "Int Spk", "SPK_OUTN",
+ "Headset Mic","MICBIAS1",
+ "MIC1_N", "Headset Mic",
+ "MIC1_P", "Headset Mic",
+ "Headset Stereophone", "HP_OUT_R",
+ "Headset Stereophone", "HP_OUT_L";
+
+ nvidia,i2s-controller = <&tegra_i2s1>;
+ nvidia,audio-codec = <&alc5632>;
+
+ clocks = <&tegra_car 112>, <&tegra_car 113>, <&tegra_car 93>;
+ clock-names = "pll_a", "pll_a_out0", "mclk";
+};
diff --git a/dts/Bindings/sound/nvidia,tegra-audio-max98090.txt b/dts/Bindings/sound/nvidia,tegra-audio-max98090.txt
new file mode 100644
index 0000000..9c7c55c
--- /dev/null
+++ b/dts/Bindings/sound/nvidia,tegra-audio-max98090.txt
@@ -0,0 +1,51 @@
+NVIDIA Tegra audio complex, with MAX98090 CODEC
+
+Required properties:
+- compatible : "nvidia,tegra-audio-max98090"
+- clocks : Must contain an entry for each entry in clock-names.
+ See ../clocks/clock-bindings.txt for details.
+- clock-names : Must include the following entries:
+ - pll_a
+ - pll_a_out0
+ - mclk (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk)
+- nvidia,model : The user-visible name of this sound complex.
+- nvidia,audio-routing : 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 for sources and
+ sinks are the MAX98090's pins (as documented in its binding), and the jacks
+ on the board:
+
+ * Headphones
+ * Speakers
+ * Mic Jack
+
+- nvidia,i2s-controller : The phandle of the Tegra I2S controller that's
+ connected to the CODEC.
+- nvidia,audio-codec : The phandle of the MAX98090 audio codec.
+
+Optional properties:
+- nvidia,hp-det-gpios : The GPIO that detect headphones are plugged in
+
+Example:
+
+sound {
+ compatible = "nvidia,tegra-audio-max98090-venice2",
+ "nvidia,tegra-audio-max98090";
+ nvidia,model = "NVIDIA Tegra Venice2";
+
+ nvidia,audio-routing =
+ "Headphones", "HPR",
+ "Headphones", "HPL",
+ "Speakers", "SPKR",
+ "Speakers", "SPKL",
+ "Mic Jack", "MICBIAS",
+ "IN34", "Mic Jack";
+
+ nvidia,i2s-controller = <&tegra_i2s1>;
+ nvidia,audio-codec = <&acodec>;
+
+ clocks = <&tegra_car TEGRA124_CLK_PLL_A>,
+ <&tegra_car TEGRA124_CLK_PLL_A_OUT0>,
+ <&tegra_car TEGRA124_CLK_EXTERN1>;
+ clock-names = "pll_a", "pll_a_out0", "mclk";
+};
diff --git a/dts/Bindings/sound/nvidia,tegra-audio-rt5640.txt b/dts/Bindings/sound/nvidia,tegra-audio-rt5640.txt
new file mode 100644
index 0000000..7788808
--- /dev/null
+++ b/dts/Bindings/sound/nvidia,tegra-audio-rt5640.txt
@@ -0,0 +1,52 @@
+NVIDIA Tegra audio complex, with RT5640 CODEC
+
+Required properties:
+- compatible : "nvidia,tegra-audio-rt5640"
+- clocks : Must contain an entry for each entry in clock-names.
+ See ../clocks/clock-bindings.txt for details.
+- clock-names : Must include the following entries:
+ - pll_a
+ - pll_a_out0
+ - mclk (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk)
+- nvidia,model : The user-visible name of this sound complex.
+- nvidia,audio-routing : 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 for sources and
+ sinks are the RT5640's pins (as documented in its binding), and the jacks
+ on the board:
+
+ * Headphones
+ * Speakers
+ * Mic Jack
+
+- nvidia,i2s-controller : The phandle of the Tegra I2S controller that's
+ connected to the CODEC.
+- nvidia,audio-codec : The phandle of the RT5640 audio codec. This binding
+ assumes that AIF1 on the CODEC is connected to Tegra.
+
+Optional properties:
+- nvidia,hp-det-gpios : The GPIO that detects headphones are plugged in
+
+Example:
+
+sound {
+ compatible = "nvidia,tegra-audio-rt5640-dalmore",
+ "nvidia,tegra-audio-rt5640";
+ nvidia,model = "NVIDIA Tegra Dalmore";
+
+ nvidia,audio-routing =
+ "Headphones", "HPOR",
+ "Headphones", "HPOL",
+ "Speakers", "SPORP",
+ "Speakers", "SPORN",
+ "Speakers", "SPOLP",
+ "Speakers", "SPOLN";
+
+ nvidia,i2s-controller = <&tegra_i2s1>;
+ nvidia,audio-codec = <&rt5640>;
+
+ nvidia,hp-det-gpios = <&gpio 143 0>; /* GPIO PR7 */
+
+ clocks = <&tegra_car 216>, <&tegra_car 217>, <&tegra_car 120>;
+ clock-names = "pll_a", "pll_a_out0", "mclk";
+};
diff --git a/dts/Bindings/sound/nvidia,tegra-audio-trimslice.txt b/dts/Bindings/sound/nvidia,tegra-audio-trimslice.txt
new file mode 100644
index 0000000..ef1fe73
--- /dev/null
+++ b/dts/Bindings/sound/nvidia,tegra-audio-trimslice.txt
@@ -0,0 +1,21 @@
+NVIDIA Tegra audio complex for TrimSlice
+
+Required properties:
+- compatible : "nvidia,tegra-audio-trimslice"
+- clocks : Must contain an entry for each entry in clock-names.
+- clock-names : Must include the following entries:
+ "pll_a" (The Tegra clock of that name),
+ "pll_a_out0" (The Tegra clock of that name),
+ "mclk" (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk)
+- nvidia,i2s-controller : The phandle of the Tegra I2S1 controller
+- nvidia,audio-codec : The phandle of the WM8903 audio codec
+
+Example:
+
+sound {
+ compatible = "nvidia,tegra-audio-trimslice";
+ nvidia,i2s-controller = <&tegra_i2s1>;
+ nvidia,audio-codec = <&codec>;
+ clocks = <&tegra_car 112>, <&tegra_car 113>, <&tegra_car 93>;
+ clock-names = "pll_a", "pll_a_out0", "mclk";
+};
diff --git a/dts/Bindings/sound/nvidia,tegra-audio-wm8753.txt b/dts/Bindings/sound/nvidia,tegra-audio-wm8753.txt
new file mode 100644
index 0000000..96f6a57
--- /dev/null
+++ b/dts/Bindings/sound/nvidia,tegra-audio-wm8753.txt
@@ -0,0 +1,40 @@
+NVIDIA Tegra audio complex
+
+Required properties:
+- compatible : "nvidia,tegra-audio-wm8753"
+- clocks : Must contain an entry for each entry in clock-names.
+ See ../clocks/clock-bindings.txt for details.
+- clock-names : Must include the following entries:
+ - pll_a
+ - pll_a_out0
+ - mclk (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk)
+- nvidia,model : The user-visible name of this sound complex.
+- nvidia,audio-routing : 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 for sources and
+ sinks are the WM8753's pins as documented in the binding for the WM8753,
+ and the jacks on the board:
+
+ * Headphone Jack
+ * Mic Jack
+
+- nvidia,i2s-controller : The phandle of the Tegra I2S1 controller
+- nvidia,audio-codec : The phandle of the WM8753 audio codec
+Example:
+
+sound {
+ compatible = "nvidia,tegra-audio-wm8753-whistler",
+ "nvidia,tegra-audio-wm8753"
+ nvidia,model = "tegra-wm8753-harmony";
+
+ nvidia,audio-routing =
+ "Headphone Jack", "LOUT1",
+ "Headphone Jack", "ROUT1";
+
+ nvidia,i2s-controller = <&i2s1>;
+ nvidia,audio-codec = <&wm8753>;
+
+ clocks = <&tegra_car 112>, <&tegra_car 113>, <&tegra_car 93>;
+ clock-names = "pll_a", "pll_a_out0", "mclk";
+};
+
diff --git a/dts/Bindings/sound/nvidia,tegra-audio-wm8903.txt b/dts/Bindings/sound/nvidia,tegra-audio-wm8903.txt
new file mode 100644
index 0000000..b795d28
--- /dev/null
+++ b/dts/Bindings/sound/nvidia,tegra-audio-wm8903.txt
@@ -0,0 +1,60 @@
+NVIDIA Tegra audio complex
+
+Required properties:
+- compatible : "nvidia,tegra-audio-wm8903"
+- clocks : Must contain an entry for each entry in clock-names.
+ See ../clocks/clock-bindings.txt for details.
+- clock-names : Must include the following entries:
+ - pll_a
+ - pll_a_out0
+ - mclk (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk)
+- nvidia,model : The user-visible name of this sound complex.
+- nvidia,audio-routing : 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 for sources and
+ sinks are the WM8903's pins (documented in the WM8903 binding document),
+ and the jacks on the board:
+
+ * Headphone Jack
+ * Int Spk
+ * Mic Jack
+
+- nvidia,i2s-controller : The phandle of the Tegra I2S1 controller
+- nvidia,audio-codec : The phandle of the WM8903 audio codec
+
+Optional properties:
+- nvidia,spkr-en-gpios : The GPIO that enables the speakers
+- nvidia,hp-mute-gpios : The GPIO that mutes the headphones
+- nvidia,hp-det-gpios : The GPIO that detect headphones are plugged in
+- nvidia,int-mic-en-gpios : The GPIO that enables the internal microphone
+- nvidia,ext-mic-en-gpios : The GPIO that enables the external microphone
+
+Example:
+
+sound {
+ compatible = "nvidia,tegra-audio-wm8903-harmony",
+ "nvidia,tegra-audio-wm8903"
+ nvidia,model = "tegra-wm8903-harmony";
+
+ nvidia,audio-routing =
+ "Headphone Jack", "HPOUTR",
+ "Headphone Jack", "HPOUTL",
+ "Int Spk", "ROP",
+ "Int Spk", "RON",
+ "Int Spk", "LOP",
+ "Int Spk", "LON",
+ "Mic Jack", "MICBIAS",
+ "IN1L", "Mic Jack";
+
+ nvidia,i2s-controller = <&i2s1>;
+ nvidia,audio-codec = <&wm8903>;
+
+ nvidia,spkr-en-gpios = <&codec 2 0>;
+ nvidia,hp-det-gpios = <&gpio 178 0>; /* gpio PW2 */
+ nvidia,int-mic-en-gpios = <&gpio 184 0>; /*gpio PX0 */
+ nvidia,ext-mic-en-gpios = <&gpio 185 0>; /* gpio PX1 */
+
+ clocks = <&tegra_car 112>, <&tegra_car 113>, <&tegra_car 93>;
+ clock-names = "pll_a", "pll_a_out0", "mclk";
+};
+
diff --git a/dts/Bindings/sound/nvidia,tegra-audio-wm9712.txt b/dts/Bindings/sound/nvidia,tegra-audio-wm9712.txt
new file mode 100644
index 0000000..436f6cd
--- /dev/null
+++ b/dts/Bindings/sound/nvidia,tegra-audio-wm9712.txt
@@ -0,0 +1,60 @@
+NVIDIA Tegra audio complex
+
+Required properties:
+- compatible : "nvidia,tegra-audio-wm9712"
+- clocks : Must contain an entry for each entry in clock-names.
+ See ../clocks/clock-bindings.txt for details.
+- clock-names : Must include the following entries:
+ - pll_a
+ - pll_a_out0
+ - mclk (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk)
+- nvidia,model : The user-visible name of this sound complex.
+- nvidia,audio-routing : 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 for sources and
+ sinks are the WM9712's pins, and the jacks on the board:
+
+ WM9712 pins:
+
+ * MONOOUT
+ * HPOUTL
+ * HPOUTR
+ * LOUT2
+ * ROUT2
+ * OUT3
+ * LINEINL
+ * LINEINR
+ * PHONE
+ * PCBEEP
+ * MIC1
+ * MIC2
+ * Mic Bias
+
+ Board connectors:
+
+ * Headphone
+ * LineIn
+ * Mic
+
+- nvidia,ac97-controller : The phandle of the Tegra AC97 controller
+
+
+Example:
+
+sound {
+ compatible = "nvidia,tegra-audio-wm9712-colibri_t20",
+ "nvidia,tegra-audio-wm9712";
+ nvidia,model = "Toradex Colibri T20";
+
+ nvidia,audio-routing =
+ "Headphone", "HPOUTL",
+ "Headphone", "HPOUTR",
+ "LineIn", "LINEINL",
+ "LineIn", "LINEINR",
+ "Mic", "MIC1";
+
+ nvidia,ac97-controller = <&ac97>;
+
+ clocks = <&tegra_car 112>, <&tegra_car 113>, <&tegra_car 93>;
+ clock-names = "pll_a", "pll_a_out0", "mclk";
+};
diff --git a/dts/Bindings/sound/nvidia,tegra20-ac97.txt b/dts/Bindings/sound/nvidia,tegra20-ac97.txt
new file mode 100644
index 0000000..eaf0010
--- /dev/null
+++ b/dts/Bindings/sound/nvidia,tegra20-ac97.txt
@@ -0,0 +1,36 @@
+NVIDIA Tegra 20 AC97 controller
+
+Required properties:
+- compatible : "nvidia,tegra20-ac97"
+- reg : Should contain AC97 controller registers location and length
+- interrupts : Should contain AC97 interrupt
+- resets : Must contain an entry for each entry in reset-names.
+ See ../reset/reset.txt for details.
+- reset-names : Must include the following entries:
+ - ac97
+- dmas : Must contain an entry for each entry in clock-names.
+ See ../dma/dma.txt for details.
+- dma-names : Must include the following entries:
+ - rx
+ - tx
+- clocks : Must contain one entry, for the module clock.
+ See ../clocks/clock-bindings.txt for details.
+- nvidia,codec-reset-gpio : The Tegra GPIO controller's phandle and the number
+ of the GPIO used to reset the external AC97 codec
+- nvidia,codec-sync-gpio : The Tegra GPIO controller's phandle and the number
+ of the GPIO corresponding with the AC97 DAP _FS line
+
+Example:
+
+ac97@70002000 {
+ compatible = "nvidia,tegra20-ac97";
+ reg = <0x70002000 0x200>;
+ interrupts = <0 81 0x04>;
+ nvidia,codec-reset-gpio = <&gpio 170 0>;
+ nvidia,codec-sync-gpio = <&gpio 120 0>;
+ clocks = <&tegra_car 3>;
+ resets = <&tegra_car 3>;
+ reset-names = "ac97";
+ dmas = <&apbdma 12>, <&apbdma 12>;
+ dma-names = "rx", "tx";
+};
diff --git a/dts/Bindings/sound/nvidia,tegra20-das.txt b/dts/Bindings/sound/nvidia,tegra20-das.txt
new file mode 100644
index 0000000..6de3a7e
--- /dev/null
+++ b/dts/Bindings/sound/nvidia,tegra20-das.txt
@@ -0,0 +1,12 @@
+NVIDIA Tegra 20 DAS (Digital Audio Switch) controller
+
+Required properties:
+- compatible : "nvidia,tegra20-das"
+- reg : Should contain DAS registers location and length
+
+Example:
+
+das@70000c00 {
+ compatible = "nvidia,tegra20-das";
+ reg = <0x70000c00 0x80>;
+};
diff --git a/dts/Bindings/sound/nvidia,tegra20-i2s.txt b/dts/Bindings/sound/nvidia,tegra20-i2s.txt
new file mode 100644
index 0000000..dc30c6b
--- /dev/null
+++ b/dts/Bindings/sound/nvidia,tegra20-i2s.txt
@@ -0,0 +1,30 @@
+NVIDIA Tegra 20 I2S controller
+
+Required properties:
+- compatible : "nvidia,tegra20-i2s"
+- reg : Should contain I2S registers location and length
+- interrupts : Should contain I2S interrupt
+- resets : Must contain an entry for each entry in reset-names.
+ See ../reset/reset.txt for details.
+- reset-names : Must include the following entries:
+ - i2s
+- dmas : Must contain an entry for each entry in clock-names.
+ See ../dma/dma.txt for details.
+- dma-names : Must include the following entries:
+ - rx
+ - tx
+- clocks : Must contain one entry, for the module clock.
+ See ../clocks/clock-bindings.txt for details.
+
+Example:
+
+i2s@70002800 {
+ compatible = "nvidia,tegra20-i2s";
+ reg = <0x70002800 0x200>;
+ interrupts = < 45 >;
+ clocks = <&tegra_car 11>;
+ resets = <&tegra_car 11>;
+ reset-names = "i2s";
+ dmas = <&apbdma 21>, <&apbdma 21>;
+ dma-names = "rx", "tx";
+};
diff --git a/dts/Bindings/sound/nvidia,tegra30-ahub.txt b/dts/Bindings/sound/nvidia,tegra30-ahub.txt
new file mode 100644
index 0000000..946e2ac
--- /dev/null
+++ b/dts/Bindings/sound/nvidia,tegra30-ahub.txt
@@ -0,0 +1,85 @@
+NVIDIA Tegra30 AHUB (Audio Hub)
+
+Required properties:
+- compatible : "nvidia,tegra30-ahub", "nvidia,tegra114-ahub", etc.
+- reg : Should contain the register physical address and length for each of
+ the AHUB's register blocks.
+ - Tegra30 requires 2 entries, for the APBIF and AHUB/AUDIO register blocks.
+ - Tegra114 requires an additional entry, for the APBIF2 register block.
+- interrupts : Should contain AHUB interrupt
+- clocks : Must contain an entry for each entry in clock-names.
+ See ../clocks/clock-bindings.txt for details.
+- clock-names : Must include the following entries:
+ - d_audio
+ - apbif
+- resets : Must contain an entry for each entry in reset-names.
+ See ../reset/reset.txt for details.
+- reset-names : Must include the following entries:
+ Tegra30 and later:
+ - d_audio
+ - apbif
+ - i2s0
+ - i2s1
+ - i2s2
+ - i2s3
+ - i2s4
+ - dam0
+ - dam1
+ - dam2
+ - spdif
+ Tegra114 and later additionally require:
+ - amx
+ - adx
+ Tegra124 and later additionally require:
+ - amx1
+ - adx1
+ - afc0
+ - afc1
+ - afc2
+ - afc3
+ - afc4
+ - afc5
+- ranges : The bus address mapping for the configlink register bus.
+ Can be empty since the mapping is 1:1.
+- dmas : Must contain an entry for each entry in clock-names.
+ See ../dma/dma.txt for details.
+- dma-names : Must include the following entries:
+ - rx0 .. rx<n>
+ - tx0 .. tx<n>
+ ... where n is:
+ Tegra30: 3
+ Tegra114, Tegra124: 9
+- #address-cells : For the configlink bus. Should be <1>;
+- #size-cells : For the configlink bus. Should be <1>.
+
+AHUB client modules need to specify the IDs of their CIFs (Client InterFaces).
+For RX CIFs, the numbers indicate the register number within AHUB routing
+register space (APBIF 0..3 RX, I2S 0..5 RX, DAM 0..2 RX 0..1, SPDIF RX 0..1).
+For TX CIFs, the numbers indicate the bit position within the AHUB routing
+registers (APBIF 0..3 TX, I2S 0..5 TX, DAM 0..2 TX, SPDIF TX 0..1).
+
+Example:
+
+ahub@70080000 {
+ compatible = "nvidia,tegra30-ahub";
+ reg = <0x70080000 0x200 0x70080200 0x100>;
+ interrupts = < 0 103 0x04 >;
+ nvidia,dma-request-selector = <&apbdma 1>;
+ clocks = <&tegra_car 106>, <&tegra_car 107>;
+ clock-names = "d_audio", "apbif";
+ resets = <&tegra_car 106>, <&tegra_car 107>, <&tegra_car 30>,
+ <&tegra_car 11>, <&tegra_car 18>, <&tegra_car 101>,
+ <&tegra_car 102>, <&tegra_car 108>, <&tegra_car 109>,
+ <&tegra_car 110>, <&tegra_car 10>;
+ reset-names = "d_audio", "apbif", "i2s0", "i2s1", "i2s2",
+ "i2s3", "i2s4", "dam0", "dam1", "dam2",
+ "spdif";
+ dmas = <&apbdma 1>, <&apbdma 1>;
+ <&apbdma 2>, <&apbdma 2>;
+ <&apbdma 3>, <&apbdma 3>;
+ <&apbdma 4>, <&apbdma 4>;
+ dma-names = "rx0", "tx0", "rx1", "tx1", "rx2", "tx2", "rx3", "tx3";
+ ranges;
+ #address-cells = <1>;
+ #size-cells = <1>;
+};
diff --git a/dts/Bindings/sound/nvidia,tegra30-i2s.txt b/dts/Bindings/sound/nvidia,tegra30-i2s.txt
new file mode 100644
index 0000000..0c113ff
--- /dev/null
+++ b/dts/Bindings/sound/nvidia,tegra30-i2s.txt
@@ -0,0 +1,24 @@
+NVIDIA Tegra30 I2S controller
+
+Required properties:
+- compatible : "nvidia,tegra30-i2s"
+- reg : Should contain I2S registers location and length
+- clocks : Must contain one entry, for the module clock.
+ See ../clocks/clock-bindings.txt for details.
+- resets : Must contain an entry for each entry in reset-names.
+ See ../reset/reset.txt for details.
+- reset-names : Must include the following entries:
+ - i2s
+- nvidia,ahub-cif-ids : The list of AHUB CIF IDs for this port, rx (playback)
+ first, tx (capture) second. See nvidia,tegra30-ahub.txt for values.
+
+Example:
+
+i2s@70080300 {
+ compatible = "nvidia,tegra30-i2s";
+ reg = <0x70080300 0x100>;
+ nvidia,ahub-cif-ids = <4 4>;
+ clocks = <&tegra_car 11>;
+ resets = <&tegra_car 11>;
+ reset-names = "i2s";
+};
diff --git a/dts/Bindings/sound/omap-abe-twl6040.txt b/dts/Bindings/sound/omap-abe-twl6040.txt
new file mode 100644
index 0000000..fd40c85
--- /dev/null
+++ b/dts/Bindings/sound/omap-abe-twl6040.txt
@@ -0,0 +1,91 @@
+* Texas Instruments OMAP4+ and twl6040 based audio setups
+
+Required properties:
+- compatible: "ti,abe-twl6040"
+- ti,model: Name of the sound card ( for example "SDP4430")
+- ti,mclk-freq: MCLK frequency for HPPLL operation
+- ti,mcpdm: phandle for the McPDM node
+- ti,twl6040: phandle for the twl6040 core node
+- ti,audio-routing: List of 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.
+
+Optional properties:
+- ti,dmic: phandle for the OMAP dmic node if the machine have it connected
+- ti,jack_detection: Need to be present if the board capable to detect jack
+ insertion, removal.
+
+Available audio endpoints for the audio-routing table:
+
+Board connectors:
+ * Headset Stereophone
+ * Earphone Spk
+ * Ext Spk
+ * Line Out
+ * Vibrator
+ * Headset Mic
+ * Main Handset Mic
+ * Sub Handset Mic
+ * Line In
+ * Digital Mic
+
+twl6040 pins:
+ * HSOL
+ * HSOR
+ * EP
+ * HFL
+ * HFR
+ * AUXL
+ * AUXR
+ * VIBRAL
+ * VIBRAR
+ * HSMIC
+ * MAINMIC
+ * SUBMIC
+ * AFML
+ * AFMR
+
+ * Headset Mic Bias
+ * Main Mic Bias
+ * Digital Mic1 Bias
+ * Digital Mic2 Bias
+
+Digital mic pins:
+ * DMic
+
+Example:
+
+sound {
+ compatible = "ti,abe-twl6040";
+ ti,model = "SDP4430";
+
+ ti,jack-detection;
+ ti,mclk-freq = <38400000>;
+
+ ti,mcpdm = <&mcpdm>;
+ ti,dmic = <&dmic>;
+
+ ti,twl6040 = <&twl6040>;
+
+ /* Audio routing */
+ ti,audio-routing =
+ "Headset Stereophone", "HSOL",
+ "Headset Stereophone", "HSOR",
+ "Earphone Spk", "EP",
+ "Ext Spk", "HFL",
+ "Ext Spk", "HFR",
+ "Line Out", "AUXL",
+ "Line Out", "AUXR",
+ "Vibrator", "VIBRAL",
+ "Vibrator", "VIBRAR",
+ "HSMIC", "Headset Mic",
+ "Headset Mic", "Headset Mic Bias",
+ "MAINMIC", "Main Handset Mic",
+ "Main Handset Mic", "Main Mic Bias",
+ "SUBMIC", "Sub Handset Mic",
+ "Sub Handset Mic", "Main Mic Bias",
+ "AFML", "Line In",
+ "AFMR", "Line In",
+ "DMic", "Digital Mic",
+ "Digital Mic", "Digital Mic1 Bias";
+};
diff --git a/dts/Bindings/sound/omap-dmic.txt b/dts/Bindings/sound/omap-dmic.txt
new file mode 100644
index 0000000..fd8105f
--- /dev/null
+++ b/dts/Bindings/sound/omap-dmic.txt
@@ -0,0 +1,21 @@
+* Texas Instruments OMAP4+ Digital Microphone Module
+
+Required properties:
+- compatible: "ti,omap4-dmic"
+- reg: Register location and size as an array:
+ <MPU access base address, size>,
+ <L3 interconnect address, size>;
+- interrupts: Interrupt number for DMIC
+- interrupt-parent: The parent interrupt controller
+- ti,hwmods: Name of the hwmod associated with OMAP dmic IP
+
+Example:
+
+dmic: dmic@4012e000 {
+ compatible = "ti,omap4-dmic";
+ reg = <0x4012e000 0x7f>, /* MPU private access */
+ <0x4902e000 0x7f>; /* L3 Interconnect */
+ interrupts = <0 114 0x4>;
+ interrupt-parent = <&gic>;
+ ti,hwmods = "dmic";
+};
diff --git a/dts/Bindings/sound/omap-mcbsp.txt b/dts/Bindings/sound/omap-mcbsp.txt
new file mode 100644
index 0000000..17cce44
--- /dev/null
+++ b/dts/Bindings/sound/omap-mcbsp.txt
@@ -0,0 +1,37 @@
+* Texas Instruments OMAP2+ McBSP module
+
+Required properties:
+- compatible: "ti,omap2420-mcbsp" for McBSP on OMAP2420
+ "ti,omap2430-mcbsp" for McBSP on OMAP2430
+ "ti,omap3-mcbsp" for McBSP on OMAP3
+ "ti,omap4-mcbsp" for McBSP on OMAP4 and newer SoC
+- reg: Register location and size, for OMAP4+ as an array:
+ <MPU access base address, size>,
+ <L3 interconnect address, size>;
+- reg-names: Array of strings associated with the address space
+- interrupts: Interrupt numbers for the McBSP port, as an array in case the
+ McBSP IP have more interrupt lines:
+ <OCP compliant irq>,
+ <TX irq>,
+ <RX irq>;
+- interrupt-names: Array of strings associated with the interrupt numbers
+- interrupt-parent: The parent interrupt controller
+- ti,buffer-size: Size of the FIFO on the port (OMAP2430 and newer SoC)
+- ti,hwmods: Name of the hwmod associated to the McBSP port
+
+Example:
+
+mcbsp2: mcbsp@49022000 {
+ compatible = "ti,omap3-mcbsp";
+ reg = <0x49022000 0xff>,
+ <0x49028000 0xff>;
+ reg-names = "mpu", "sidetone";
+ interrupts = <0 17 0x4>, /* OCP compliant interrupt */
+ <0 62 0x4>, /* TX interrupt */
+ <0 63 0x4>, /* RX interrupt */
+ <0 4 0x4>; /* Sidetone */
+ interrupt-names = "common", "tx", "rx", "sidetone";
+ interrupt-parent = <&intc>;
+ ti,buffer-size = <1280>;
+ ti,hwmods = "mcbsp2";
+};
diff --git a/dts/Bindings/sound/omap-mcpdm.txt b/dts/Bindings/sound/omap-mcpdm.txt
new file mode 100644
index 0000000..0741dff
--- /dev/null
+++ b/dts/Bindings/sound/omap-mcpdm.txt
@@ -0,0 +1,21 @@
+* Texas Instruments OMAP4+ McPDM
+
+Required properties:
+- compatible: "ti,omap4-mcpdm"
+- reg: Register location and size as an array:
+ <MPU access base address, size>,
+ <L3 interconnect address, size>;
+- interrupts: Interrupt number for McPDM
+- interrupt-parent: The parent interrupt controller
+- ti,hwmods: Name of the hwmod associated to the McPDM
+
+Example:
+
+mcpdm: mcpdm@40132000 {
+ compatible = "ti,omap4-mcpdm";
+ reg = <0x40132000 0x7f>, /* MPU private access */
+ <0x49032000 0x7f>; /* L3 Interconnect */
+ interrupts = <0 112 0x4>;
+ interrupt-parent = <&gic>;
+ ti,hwmods = "mcpdm";
+};
diff --git a/dts/Bindings/sound/omap-twl4030.txt b/dts/Bindings/sound/omap-twl4030.txt
new file mode 100644
index 0000000..1ab6bc8
--- /dev/null
+++ b/dts/Bindings/sound/omap-twl4030.txt
@@ -0,0 +1,63 @@
+* Texas Instruments SoC with twl4030 based audio setups
+
+Required properties:
+- compatible: "ti,omap-twl4030"
+- ti,model: Name of the sound card (for example "omap3beagle")
+- ti,mcbsp: phandle for the McBSP node
+- ti,codec: phandle for the twl4030 audio node
+
+Optional properties:
+- ti,mcbsp-voice: phandle for the McBSP node connected to the voice port of twl
+- ti, jack-det-gpio: Jack detect GPIO
+- ti,audio-routing: List of 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.
+ If the routing is not provided all possible connection will be available
+
+Available audio endpoints for the audio-routing table:
+
+Board connectors:
+ * Headset Stereophone
+ * Earpiece Spk
+ * Handsfree Spk
+ * Ext Spk
+ * Main Mic
+ * Sub Mic
+ * Headset Mic
+ * Carkit Mic
+ * Digital0 Mic
+ * Digital1 Mic
+ * Line In
+
+twl4030 pins:
+ * HSOL
+ * HSOR
+ * EARPIECE
+ * HFL
+ * HFR
+ * PREDRIVEL
+ * PREDRIVER
+ * CARKITL
+ * CARKITR
+ * MAINMIC
+ * SUBMIC
+ * HSMIC
+ * DIGIMIC0
+ * DIGIMIC1
+ * CARKITMIC
+ * AUXL
+ * AUXR
+
+ * Headset Mic Bias
+ * Mic Bias 1 /* Used for Main Mic or Digimic0 */
+ * Mic Bias 2 /* Used for Sub Mic or Digimic1 */
+
+Example:
+
+sound {
+ compatible = "ti,omap-twl4030";
+ ti,model = "omap3beagle";
+
+ ti,mcbsp = <&mcbsp2>;
+ ti,codec = <&twl_audio>;
+};
diff --git a/dts/Bindings/sound/pcm1792a.txt b/dts/Bindings/sound/pcm1792a.txt
new file mode 100644
index 0000000..970ba1e
--- /dev/null
+++ b/dts/Bindings/sound/pcm1792a.txt
@@ -0,0 +1,18 @@
+Texas Instruments pcm1792a DT bindings
+
+This driver supports the SPI bus.
+
+Required properties:
+
+ - compatible: "ti,pcm1792a"
+
+For required properties on SPI, please consult
+Documentation/devicetree/bindings/spi/spi-bus.txt
+
+Examples:
+
+ codec_spi: 1792a@0 {
+ compatible = "ti,pcm1792a";
+ spi-max-frequency = <600000>;
+ };
+
diff --git a/dts/Bindings/sound/pcm512x.txt b/dts/Bindings/sound/pcm512x.txt
new file mode 100644
index 0000000..faff75e
--- /dev/null
+++ b/dts/Bindings/sound/pcm512x.txt
@@ -0,0 +1,30 @@
+PCM512x audio CODECs
+
+These devices support both I2C and SPI (configured with pin strapping
+on the board).
+
+Required properties:
+
+ - compatible : One of "ti,pcm5121" or "ti,pcm5122"
+
+ - reg : the I2C address of the device for I2C, the chip select
+ number for SPI.
+
+ - AVDD-supply, DVDD-supply, and CPVDD-supply : power supplies for the
+ device, as covered in bindings/regulator/regulator.txt
+
+Optional properties:
+
+ - clocks : A clock specifier for the clock connected as SCLK. If this
+ is absent the device will be configured to clock from BCLK.
+
+Example:
+
+ pcm5122: pcm5122@4c {
+ compatible = "ti,pcm5122";
+ reg = <0x4c>;
+
+ AVDD-supply = <&reg_3v3_analog>;
+ DVDD-supply = <&reg_1v8>;
+ CPVDD-supply = <&reg_3v3>;
+ };
diff --git a/dts/Bindings/sound/renesas,fsi.txt b/dts/Bindings/sound/renesas,fsi.txt
new file mode 100644
index 0000000..c5be003
--- /dev/null
+++ b/dts/Bindings/sound/renesas,fsi.txt
@@ -0,0 +1,26 @@
+Renesas FSI
+
+Required properties:
+- compatible : "renesas,sh_fsi2" or "renesas,sh_fsi"
+- reg : Should contain the register physical address and length
+- interrupts : Should contain FSI interrupt
+
+- fsia,spdif-connection : FSI is connected by S/PDFI
+- fsia,stream-mode-support : FSI supports 16bit stream mode.
+- fsia,use-internal-clock : FSI uses internal clock when master mode.
+
+- fsib,spdif-connection : same as fsia
+- fsib,stream-mode-support : same as fsia
+- fsib,use-internal-clock : same as fsia
+
+Example:
+
+sh_fsi2: sh_fsi2@0xec230000 {
+ compatible = "renesas,sh_fsi2";
+ reg = <0xec230000 0x400>;
+ interrupts = <0 146 0x4>;
+
+ fsia,spdif-connection;
+ fsia,stream-mode-support;
+ fsia,use-internal-clock;
+};
diff --git a/dts/Bindings/sound/renesas,rsnd.txt b/dts/Bindings/sound/renesas,rsnd.txt
new file mode 100644
index 0000000..a44e917
--- /dev/null
+++ b/dts/Bindings/sound/renesas,rsnd.txt
@@ -0,0 +1,105 @@
+Renesas R-Car sound
+
+Required properties:
+- compatible : "renesas,rcar_sound-gen1" if generation1
+ "renesas,rcar_sound-gen2" if generation2
+- reg : Should contain the register physical address.
+ required register is
+ SRU/ADG/SSI if generation1
+ SRU/ADG/SSIU/SSI if generation2
+- rcar_sound,ssi : Should contain SSI feature.
+ The number of SSI subnode should be same as HW.
+ see below for detail.
+- rcar_sound,src : Should contain SRC feature.
+ The number of SRC subnode should be same as HW.
+ see below for detail.
+- rcar_sound,dai : DAI contents.
+ The number of DAI subnode should be same as HW.
+ see below for detail.
+
+SSI subnode properties:
+- interrupts : Should contain SSI interrupt for PIO transfer
+- shared-pin : if shared clock pin
+
+SRC subnode properties:
+no properties at this point
+
+DAI subnode properties:
+- playback : list of playback modules
+- capture : list of capture modules
+
+Example:
+
+rcar_sound: rcar_sound@0xffd90000 {
+ #sound-dai-cells = <1>;
+ compatible = "renesas,rcar_sound-gen2";
+ reg = <0 0xec500000 0 0x1000>, /* SCU */
+ <0 0xec5a0000 0 0x100>, /* ADG */
+ <0 0xec540000 0 0x1000>, /* SSIU */
+ <0 0xec541000 0 0x1280>; /* SSI */
+
+ rcar_sound,src {
+ src0: src@0 { };
+ src1: src@1 { };
+ src2: src@2 { };
+ src3: src@3 { };
+ src4: src@4 { };
+ src5: src@5 { };
+ src6: src@6 { };
+ src7: src@7 { };
+ src8: src@8 { };
+ src9: src@9 { };
+ };
+
+ rcar_sound,ssi {
+ ssi0: ssi@0 {
+ interrupts = <0 370 IRQ_TYPE_LEVEL_HIGH>;
+ };
+ ssi1: ssi@1 {
+ interrupts = <0 371 IRQ_TYPE_LEVEL_HIGH>;
+ };
+ ssi2: ssi@2 {
+ interrupts = <0 372 IRQ_TYPE_LEVEL_HIGH>;
+ };
+ ssi3: ssi@3 {
+ interrupts = <0 373 IRQ_TYPE_LEVEL_HIGH>;
+ };
+ ssi4: ssi@4 {
+ interrupts = <0 374 IRQ_TYPE_LEVEL_HIGH>;
+ };
+ ssi5: ssi@5 {
+ interrupts = <0 375 IRQ_TYPE_LEVEL_HIGH>;
+ };
+ ssi6: ssi@6 {
+ interrupts = <0 376 IRQ_TYPE_LEVEL_HIGH>;
+ };
+ ssi7: ssi@7 {
+ interrupts = <0 377 IRQ_TYPE_LEVEL_HIGH>;
+ };
+ ssi8: ssi@8 {
+ interrupts = <0 378 IRQ_TYPE_LEVEL_HIGH>;
+ };
+ ssi9: ssi@9 {
+ interrupts = <0 379 IRQ_TYPE_LEVEL_HIGH>;
+ };
+ };
+
+ rcar_sound,dai {
+ dai0 {
+ playback = <&ssi5 &src5>;
+ capture = <&ssi6>;
+ };
+ dai1 {
+ playback = <&ssi3>;
+ };
+ dai2 {
+ capture = <&ssi4>;
+ };
+ dai3 {
+ playback = <&ssi7>;
+ };
+ dai4 {
+ capture = <&ssi8>;
+ };
+ };
+};
diff --git a/dts/Bindings/sound/rt5640.txt b/dts/Bindings/sound/rt5640.txt
new file mode 100644
index 0000000..068a114
--- /dev/null
+++ b/dts/Bindings/sound/rt5640.txt
@@ -0,0 +1,50 @@
+RT5640 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+- compatible : "realtek,rt5640".
+
+- reg : The I2C address of the device.
+
+- interrupts : The CODEC's interrupt output.
+
+Optional properties:
+
+- realtek,in1-differential
+- realtek,in2-differential
+ Boolean. Indicate MIC1/2 input are differential, rather than single-ended.
+
+- realtek,ldo1-en-gpios : The GPIO that controls the CODEC's LDO1_EN pin.
+
+Pins on the device (for linking into audio routes):
+
+ * DMIC1
+ * DMIC2
+ * MICBIAS1
+ * IN1P
+ * IN1R
+ * IN2P
+ * IN2R
+ * HPOL
+ * HPOR
+ * LOUTL
+ * LOUTR
+ * MONOP
+ * MONON
+ * SPOLP
+ * SPOLN
+ * SPORP
+ * SPORN
+
+Example:
+
+rt5640 {
+ compatible = "realtek,rt5640";
+ reg = <0x1c>;
+ interrupt-parent = <&gpio>;
+ interrupts = <TEGRA_GPIO(W, 3) GPIO_ACTIVE_HIGH>;
+ realtek,ldo1-en-gpios =
+ <&gpio TEGRA_GPIO(V, 3) GPIO_ACTIVE_HIGH>;
+};
diff --git a/dts/Bindings/sound/samsung,smdk-wm8994.txt b/dts/Bindings/sound/samsung,smdk-wm8994.txt
new file mode 100644
index 0000000..4686646
--- /dev/null
+++ b/dts/Bindings/sound/samsung,smdk-wm8994.txt
@@ -0,0 +1,14 @@
+Samsung SMDK audio complex
+
+Required properties:
+- compatible : "samsung,smdk-wm8994"
+- samsung,i2s-controller: The phandle of the Samsung I2S0 controller
+- samsung,audio-codec: The phandle of the WM8994 audio codec
+Example:
+
+sound {
+ compatible = "samsung,smdk-wm8994";
+
+ samsung,i2s-controller = <&i2s0>;
+ samsung,audio-codec = <&wm8994>;
+};
diff --git a/dts/Bindings/sound/samsung-i2s.txt b/dts/Bindings/sound/samsung-i2s.txt
new file mode 100644
index 0000000..7386d44
--- /dev/null
+++ b/dts/Bindings/sound/samsung-i2s.txt
@@ -0,0 +1,53 @@
+* Samsung I2S controller
+
+Required SoC Specific Properties:
+
+- compatible : should be one of the following.
+ - samsung,s3c6410-i2s: for 8/16/24bit stereo I2S.
+ - samsung,s5pv210-i2s: for 8/16/24bit multichannel(5.1) I2S with
+ secondary fifo, s/w reset control and internal mux for root clk src.
+ - samsung,exynos5420-i2s: for 8/16/24bit multichannel(7.1) I2S with
+ secondary fifo, s/w reset control, internal mux for root clk src and
+ TDM support. TDM (Time division multiplexing) is to allow transfer of
+ multiple channel audio data on single data line.
+
+- reg: physical base address of the controller and length of memory mapped
+ region.
+- dmas: list of DMA controller phandle and DMA request line ordered pairs.
+- dma-names: identifier string for each DMA request line in the dmas property.
+ These strings correspond 1:1 with the ordered pairs in dmas.
+- clocks: Handle to iis clock and RCLK source clk.
+- clock-names:
+ i2s0 uses some base clks from CMU and some are from audio subsystem internal
+ clock controller. The clock names for i2s0 should be "iis", "i2s_opclk0" and
+ "i2s_opclk1" as shown in the example below.
+ i2s1 and i2s2 uses clocks from CMU. The clock names for i2s1 and i2s2 should
+ be "iis" and "i2s_opclk0".
+ "iis" is the i2s bus clock and i2s_opclk0, i2s_opclk1 are sources of the root
+ clk. i2s0 has internal mux to select the source of root clk and i2s1 and i2s2
+ doesn't have any such mux.
+
+Optional SoC Specific Properties:
+
+- samsung,idma-addr: Internal DMA register base address of the audio
+ sub system(used in secondary sound source).
+- pinctrl-0: Should specify pin control groups used for this controller.
+- pinctrl-names: Should contain only one value - "default".
+
+Example:
+
+i2s0: i2s@03830000 {
+ compatible = "samsung,s5pv210-i2s";
+ reg = <0x03830000 0x100>;
+ dmas = <&pdma0 10
+ &pdma0 9
+ &pdma0 8>;
+ dma-names = "tx", "rx", "tx-sec";
+ clocks = <&clock_audss EXYNOS_I2S_BUS>,
+ <&clock_audss EXYNOS_I2S_BUS>,
+ <&clock_audss EXYNOS_SCLK_I2S>;
+ clock-names = "iis", "i2s_opclk0", "i2s_opclk1";
+ samsung,idma-addr = <0x03000000>;
+ pinctrl-names = "default";
+ pinctrl-0 = <&i2s0_bus>;
+};
diff --git a/dts/Bindings/sound/sgtl5000.txt b/dts/Bindings/sound/sgtl5000.txt
new file mode 100644
index 0000000..955df60
--- /dev/null
+++ b/dts/Bindings/sound/sgtl5000.txt
@@ -0,0 +1,16 @@
+* Freescale SGTL5000 Stereo Codec
+
+Required properties:
+- compatible : "fsl,sgtl5000".
+
+- reg : the I2C address of the device
+
+- clocks : the clock provider of SYS_MCLK
+
+Example:
+
+codec: sgtl5000@0a {
+ compatible = "fsl,sgtl5000";
+ reg = <0x0a>;
+ clocks = <&clks 150>;
+};
diff --git a/dts/Bindings/sound/simple-card.txt b/dts/Bindings/sound/simple-card.txt
new file mode 100644
index 0000000..131aa2a
--- /dev/null
+++ b/dts/Bindings/sound/simple-card.txt
@@ -0,0 +1,136 @@
+Simple-Card:
+
+Simple-Card specifies audio DAI connection of SoC <-> codec.
+
+Required properties:
+
+- compatible : "simple-audio-card"
+
+Optional properties:
+
+- simple-audio-card,name : User specified audio sound card name, one string
+ property.
+- simple-audio-card,format : CPU/CODEC common audio format.
+ "i2s", "right_j", "left_j" , "dsp_a"
+ "dsp_b", "ac97", "pdm", "msb", "lsb"
+- simple-audio-card,widgets : Please refer to widgets.txt.
+- simple-audio-card,routing : 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.
+- dai-tdm-slot-num : Please refer to tdm-slot.txt.
+- dai-tdm-slot-width : Please refer to tdm-slot.txt.
+
+Required subnodes:
+
+- simple-audio-card,dai-link : container for the CPU and CODEC sub-nodes
+ This container may be omitted when the
+ card has only one DAI link.
+ See the examples.
+
+- simple-audio-card,cpu : CPU sub-node
+- simple-audio-card,codec : CODEC sub-node
+
+Required CPU/CODEC subnodes properties:
+
+- sound-dai : phandle and port of CPU/CODEC
+
+Optional CPU/CODEC subnodes properties:
+
+- format : CPU/CODEC specific audio format if needed.
+ see simple-audio-card,format
+- frame-master : bool property. add this if subnode is frame master
+- bitclock-master : bool property. add this if subnode is bitclock master
+- bitclock-inversion : bool property. add this if subnode has clock inversion
+- frame-inversion : bool property. add this if subnode has frame inversion
+- clocks / system-clock-frequency : specify subnode's clock if needed.
+ it can be specified via "clocks" if system has
+ clock node (= common clock), or "system-clock-frequency"
+ (if system doens't support common clock)
+
+Note:
+ * For 'format', 'frame-master', 'bitclock-master', 'bitclock-inversion' and
+ 'frame-inversion', the simple card will use the settings of CODEC for both
+ CPU and CODEC sides as we need to keep the settings identical for both ends
+ of the link.
+
+Example 1 - single DAI link:
+
+sound {
+ compatible = "simple-audio-card";
+ simple-audio-card,name = "VF610-Tower-Sound-Card";
+ simple-audio-card,format = "left_j";
+ simple-audio-card,widgets =
+ "Microphone", "Microphone Jack",
+ "Headphone", "Headphone Jack",
+ "Speaker", "External Speaker";
+ simple-audio-card,routing =
+ "MIC_IN", "Microphone Jack",
+ "Headphone Jack", "HP_OUT",
+ "External Speaker", "LINE_OUT";
+
+ dai-tdm-slot-num = <2>;
+ dai-tdm-slot-width = <8>;
+
+ simple-audio-card,cpu {
+ sound-dai = <&sh_fsi2 0>;
+ };
+
+ simple-audio-card,codec {
+ sound-dai = <&ak4648>;
+ bitclock-master;
+ frame-master;
+ clocks = <&osc>;
+ };
+};
+
+&i2c0 {
+ ak4648: ak4648@12 {
+ #sound-dai-cells = <0>;
+ compatible = "asahi-kasei,ak4648";
+ reg = <0x12>;
+ };
+};
+
+sh_fsi2: sh_fsi2@ec230000 {
+ #sound-dai-cells = <1>;
+ compatible = "renesas,sh_fsi2";
+ reg = <0xec230000 0x400>;
+ interrupt-parent = <&gic>;
+ interrupts = <0 146 0x4>;
+};
+
+Example 2 - many DAI links:
+
+sound {
+ compatible = "simple-audio-card";
+ simple-audio-card,name = "Cubox Audio";
+ simple-audio-card,format = "i2s";
+
+ simple-audio-card,dai-link@0 { /* I2S - HDMI */
+ simple-audio-card,cpu {
+ sound-dai = <&audio1 0>;
+ };
+ simple-audio-card,codec {
+ sound-dai = <&tda998x 0>;
+ };
+ };
+
+ simple-audio-card,dai-link@1 { /* S/PDIF - HDMI */
+ simple-audio-card,cpu {
+ sound-dai = <&audio1 1>;
+ };
+ simple-audio-card,codec {
+ sound-dai = <&tda998x 1>;
+ };
+ };
+
+ simple-audio-card,dai-link@2 { /* S/PDIF - S/PDIF */
+ simple-audio-card,cpu {
+ sound-dai = <&audio1 1>;
+ };
+ simple-audio-card,codec {
+ sound-dai = <&spdif_codec>;
+ };
+ };
+};
diff --git a/dts/Bindings/sound/sirf-audio-codec.txt b/dts/Bindings/sound/sirf-audio-codec.txt
new file mode 100644
index 0000000..062f5ec
--- /dev/null
+++ b/dts/Bindings/sound/sirf-audio-codec.txt
@@ -0,0 +1,17 @@
+SiRF internal audio CODEC
+
+Required properties:
+
+ - compatible : "sirf,atlas6-audio-codec" or "sirf,prima2-audio-codec"
+
+ - reg : the register address of the device.
+
+ - clocks: the clock of SiRF internal audio codec
+
+Example:
+
+audiocodec: audiocodec@b0040000 {
+ compatible = "sirf,atlas6-audio-codec";
+ reg = <0xb0040000 0x10000>;
+ clocks = <&clks 27>;
+};
diff --git a/dts/Bindings/sound/sirf-audio-port.txt b/dts/Bindings/sound/sirf-audio-port.txt
new file mode 100644
index 0000000..1f66de3
--- /dev/null
+++ b/dts/Bindings/sound/sirf-audio-port.txt
@@ -0,0 +1,20 @@
+* SiRF SoC audio port
+
+Required properties:
+- compatible: "sirf,audio-port"
+- reg: Base address and size entries:
+- dmas: List of DMA controller phandle and DMA request line ordered pairs.
+- dma-names: Identifier string for each DMA request line in the dmas property.
+ These strings correspond 1:1 with the ordered pairs in dmas.
+
+ One of the DMA channels will be responsible for transmission (should be
+ named "tx") and one for reception (should be named "rx").
+
+Example:
+
+audioport: audioport@b0040000 {
+ compatible = "sirf,audio-port";
+ reg = <0xb0040000 0x10000>;
+ dmas = <&dmac1 3>, <&dmac1 8>;
+ dma-names = "rx", "tx";
+};
diff --git a/dts/Bindings/sound/sirf-audio.txt b/dts/Bindings/sound/sirf-audio.txt
new file mode 100644
index 0000000..c88882c
--- /dev/null
+++ b/dts/Bindings/sound/sirf-audio.txt
@@ -0,0 +1,41 @@
+* SiRF atlas6 and prima2 internal audio codec and port based audio setups
+
+Required properties:
+- compatible: "sirf,sirf-audio-card"
+- sirf,audio-platform: phandle for the platform node
+- sirf,audio-codec: phandle for the SiRF internal codec node
+
+Optional properties:
+- hp-pa-gpios: Need to be present if the board need control external
+ headphone amplifier.
+- spk-pa-gpios: Need to be present if the board need control external
+ speaker amplifier.
+- hp-switch-gpios: Need to be present if the board capable to detect jack
+ insertion, removal.
+
+Available audio endpoints for the audio-routing table:
+
+Board connectors:
+ * Headset Stereophone
+ * Ext Spk
+ * Line In
+ * Mic
+
+SiRF internal audio codec pins:
+ * HPOUTL
+ * HPOUTR
+ * SPKOUT
+ * Ext Mic
+ * Mic Bias
+
+Example:
+
+sound {
+ compatible = "sirf,sirf-audio-card";
+ sirf,audio-codec = <&audiocodec>;
+ sirf,audio-platform = <&audioport>;
+ hp-pa-gpios = <&gpio 44 0>;
+ spk-pa-gpios = <&gpio 46 0>;
+ hp-switch-gpios = <&gpio 45 0>;
+};
+
diff --git a/dts/Bindings/sound/soc-ac97link.txt b/dts/Bindings/sound/soc-ac97link.txt
new file mode 100644
index 0000000..80152a8
--- /dev/null
+++ b/dts/Bindings/sound/soc-ac97link.txt
@@ -0,0 +1,28 @@
+AC97 link bindings
+
+These bindings can be included within any other device node.
+
+Required properties:
+ - pinctrl-names: Has to contain following states to setup the correct
+ pinmuxing for the used gpios:
+ "ac97-running": AC97-link is active
+ "ac97-reset": AC97-link reset state
+ "ac97-warm-reset": AC97-link warm reset state
+ - ac97-gpios: List of gpio phandles with args in the order ac97-sync,
+ ac97-sdata, ac97-reset
+
+
+Example:
+
+ssi {
+ ...
+
+ pinctrl-names = "default", "ac97-running", "ac97-reset", "ac97-warm-reset";
+ pinctrl-0 = <&ac97link_running>;
+ pinctrl-1 = <&ac97link_running>;
+ pinctrl-2 = <&ac97link_reset>;
+ pinctrl-3 = <&ac97link_warm_reset>;
+ ac97-gpios = <&gpio3 20 0 &gpio3 22 0 &gpio3 28 0>;
+
+ ...
+};
diff --git a/dts/Bindings/sound/spdif-receiver.txt b/dts/Bindings/sound/spdif-receiver.txt
new file mode 100644
index 0000000..80f807b
--- /dev/null
+++ b/dts/Bindings/sound/spdif-receiver.txt
@@ -0,0 +1,10 @@
+Device-Tree bindings for dummy spdif receiver
+
+Required properties:
+ - compatible: should be "linux,spdif-dir".
+
+Example node:
+
+ codec: spdif-receiver {
+ compatible = "linux,spdif-dir";
+ };
diff --git a/dts/Bindings/sound/spdif-transmitter.txt b/dts/Bindings/sound/spdif-transmitter.txt
new file mode 100644
index 0000000..55a8584
--- /dev/null
+++ b/dts/Bindings/sound/spdif-transmitter.txt
@@ -0,0 +1,10 @@
+Device-Tree bindings for dummy spdif transmitter
+
+Required properties:
+ - compatible: should be "linux,spdif-dit".
+
+Example node:
+
+ codec: spdif-transmitter {
+ compatible = "linux,spdif-dit";
+ };
diff --git a/dts/Bindings/sound/ssm2518.txt b/dts/Bindings/sound/ssm2518.txt
new file mode 100644
index 0000000..59381a7
--- /dev/null
+++ b/dts/Bindings/sound/ssm2518.txt
@@ -0,0 +1,20 @@
+SSM2518 audio amplifier
+
+This device supports I2C only.
+
+Required properties:
+ - compatible : Must be "adi,ssm2518"
+ - reg : the I2C address of the device. This will either be 0x34 (ADDR pin low)
+ or 0x35 (ADDR pin high)
+
+Optional properties:
+ - gpios : GPIO connected to the nSD pin. If the property is not present it is
+ assumed that the nSD pin is hardwired to always on.
+
+Example:
+
+ ssm2518: ssm2518@34 {
+ compatible = "adi,ssm2518";
+ reg = <0x34>;
+ gpios = <&gpio 5 0>;
+ };
diff --git a/dts/Bindings/sound/tdm-slot.txt b/dts/Bindings/sound/tdm-slot.txt
new file mode 100644
index 0000000..6a2c842
--- /dev/null
+++ b/dts/Bindings/sound/tdm-slot.txt
@@ -0,0 +1,20 @@
+TDM slot:
+
+This specifies audio DAI's TDM slot.
+
+TDM slot properties:
+dai-tdm-slot-num : Number of slots in use.
+dai-tdm-slot-width : Width in bits for each slot.
+
+For instance:
+ dai-tdm-slot-num = <2>;
+ dai-tdm-slot-width = <8>;
+
+And for each spcified driver, there could be one .of_xlate_tdm_slot_mask()
+to specify a explicit mapping of the channels and the slots. If it's absent
+the default snd_soc_of_xlate_tdm_slot_mask() will be used to generating the
+tx and rx masks.
+
+For snd_soc_of_xlate_tdm_slot_mask(), the tx and rx masks will use a 1 bit
+for an active slot as default, and the default active bits are at the LSB of
+the masks.
diff --git a/dts/Bindings/sound/ti,pcm1681.txt b/dts/Bindings/sound/ti,pcm1681.txt
new file mode 100644
index 0000000..4df1718
--- /dev/null
+++ b/dts/Bindings/sound/ti,pcm1681.txt
@@ -0,0 +1,15 @@
+Texas Instruments PCM1681 8-channel PWM Processor
+
+Required properties:
+
+ - compatible: Should contain "ti,pcm1681".
+ - reg: The i2c address. Should contain <0x4c>.
+
+Examples:
+
+ i2c_bus {
+ pcm1681@4c {
+ compatible = "ti,pcm1681";
+ reg = <0x4c>;
+ };
+ };
diff --git a/dts/Bindings/sound/ti,tas5086.txt b/dts/Bindings/sound/ti,tas5086.txt
new file mode 100644
index 0000000..d2866a0
--- /dev/null
+++ b/dts/Bindings/sound/ti,tas5086.txt
@@ -0,0 +1,43 @@
+Texas Instruments TAS5086 6-channel PWM Processor
+
+Required properties:
+
+ - compatible: Should contain "ti,tas5086".
+ - reg: The i2c address. Should contain <0x1b>.
+
+Optional properties:
+
+ - reset-gpio: A GPIO spec to define which pin is connected to the
+ chip's !RESET pin. If specified, the driver will
+ assert a hardware reset at probe time.
+
+ - ti,charge-period: This property should contain the time in microseconds
+ that closely matches the external single-ended
+ split-capacitor charge period. The hardware chip
+ waits for this period of time before starting the
+ PWM signals. This helps reduce pops and clicks.
+
+ When not specified, the hardware default of 1300ms
+ is retained.
+
+ - ti,mid-z-channel-X: Boolean properties, X being a number from 1 to 6.
+ If given, channel X will start with the Mid-Z start
+ sequence, otherwise the default Low-Z scheme is used.
+
+ The correct configuration depends on how the power
+ stages connected to the PWM output pins work. Not all
+ power stages are compatible to Mid-Z - please refer
+ to the datasheets for more details.
+
+ Most systems should not set any of these properties.
+
+Examples:
+
+ i2c_bus {
+ tas5086@1b {
+ compatible = "ti,tas5086";
+ reg = <0x1b>;
+ reset-gpio = <&gpio 23 0>;
+ ti,charge-period = <156000>;
+ };
+ };
diff --git a/dts/Bindings/sound/tlv320aic31xx.txt b/dts/Bindings/sound/tlv320aic31xx.txt
new file mode 100644
index 0000000..74c66de
--- /dev/null
+++ b/dts/Bindings/sound/tlv320aic31xx.txt
@@ -0,0 +1,61 @@
+Texas Instruments - tlv320aic31xx Codec module
+
+The tlv320aic31xx serial control bus communicates through I2C protocols
+
+Required properties:
+
+- compatible - "string" - One of:
+ "ti,tlv320aic310x" - Generic TLV320AIC31xx with mono speaker amp
+ "ti,tlv320aic311x" - Generic TLV320AIC31xx with stereo speaker amp
+ "ti,tlv320aic3100" - TLV320AIC3100 (mono speaker amp, no MiniDSP)
+ "ti,tlv320aic3110" - TLV320AIC3110 (stereo speaker amp, no MiniDSP)
+ "ti,tlv320aic3120" - TLV320AIC3120 (mono speaker amp, MiniDSP)
+ "ti,tlv320aic3111" - TLV320AIC3111 (stereo speaker amp, MiniDSP)
+
+- reg - <int> - I2C slave address
+
+
+Optional properties:
+
+- gpio-reset - gpio pin number used for codec reset
+- ai31xx-micbias-vg - MicBias Voltage setting
+ 1 or MICBIAS_2_0V - MICBIAS output is powered to 2.0V
+ 2 or MICBIAS_2_5V - MICBIAS output is powered to 2.5V
+ 3 or MICBIAS_AVDD - MICBIAS output is connected to AVDD
+ If this node is not mentioned or if the value is unknown, then
+ micbias is set to 2.0V.
+- HPVDD-supply, SPRVDD-supply, SPLVDD-supply, AVDD-supply, IOVDD-supply,
+ DVDD-supply : power supplies for the device as covered in
+ Documentation/devicetree/bindings/regulator/regulator.txt
+
+CODEC output pins:
+ * HPL
+ * HPR
+ * SPL, devices with stereo speaker amp
+ * SPR, devices with stereo speaker amp
+ * SPK, devices with mono speaker amp
+ * MICBIAS
+
+CODEC input pins:
+ * MIC1LP
+ * MIC1RP
+ * MIC1LM
+
+The pins can be used in referring sound node's audio-routing property.
+
+Example:
+#include <dt-bindings/sound/tlv320aic31xx-micbias.h>
+
+tlv320aic31xx: tlv320aic31xx@18 {
+ compatible = "ti,tlv320aic311x";
+ reg = <0x18>;
+
+ ai31xx-micbias-vg = <MICBIAS_OFF>;
+
+ HPVDD-supply = <&regulator>;
+ SPRVDD-supply = <&regulator>;
+ SPLVDD-supply = <&regulator>;
+ AVDD-supply = <&regulator>;
+ IOVDD-supply = <&regulator>;
+ DVDD-supply = <&regulator>;
+};
diff --git a/dts/Bindings/sound/tlv320aic32x4.txt b/dts/Bindings/sound/tlv320aic32x4.txt
new file mode 100644
index 0000000..5e2741a
--- /dev/null
+++ b/dts/Bindings/sound/tlv320aic32x4.txt
@@ -0,0 +1,30 @@
+Texas Instruments - tlv320aic32x4 Codec module
+
+The tlv320aic32x4 serial control bus communicates through I2C protocols
+
+Required properties:
+ - compatible: Should be "ti,tlv320aic32x4"
+ - reg: I2C slave address
+ - supply-*: Required supply regulators are:
+ "iov" - digital IO power supply
+ "ldoin" - LDO power supply
+ "dv" - Digital core power supply
+ "av" - Analog core power supply
+ If you supply ldoin, dv and av are optional. Otherwise they are required
+ See regulator/regulator.txt for more information about the detailed binding
+ format.
+
+Optional properties:
+ - reset-gpios: Reset-GPIO phandle with args as described in gpio/gpio.txt
+ - clocks/clock-names: Clock named 'mclk' for the master clock of the codec.
+ See clock/clock-bindings.txt for information about the detailed format.
+
+
+Example:
+
+codec: tlv320aic32x4@18 {
+ compatible = "ti,tlv320aic32x4";
+ reg = <0x18>;
+ clocks = <&clks 201>;
+ clock-names = "mclk";
+};
diff --git a/dts/Bindings/sound/tlv320aic3x.txt b/dts/Bindings/sound/tlv320aic3x.txt
new file mode 100644
index 0000000..5e6040c
--- /dev/null
+++ b/dts/Bindings/sound/tlv320aic3x.txt
@@ -0,0 +1,59 @@
+Texas Instruments - tlv320aic3x Codec module
+
+The tlv320aic3x serial control bus communicates through I2C protocols
+
+Required properties:
+
+- compatible - "string" - One of:
+ "ti,tlv320aic3x" - Generic TLV320AIC3x device
+ "ti,tlv320aic33" - TLV320AIC33
+ "ti,tlv320aic3007" - TLV320AIC3007
+ "ti,tlv320aic3106" - TLV320AIC3106
+
+
+- reg - <int> - I2C slave address
+
+
+Optional properties:
+
+- gpio-reset - gpio pin number used for codec reset
+- ai3x-gpio-func - <array of 2 int> - AIC3X_GPIO1 & AIC3X_GPIO2 Functionality
+- ai3x-micbias-vg - MicBias Voltage required.
+ 1 - MICBIAS output is powered to 2.0V,
+ 2 - MICBIAS output is powered to 2.5V,
+ 3 - MICBIAS output is connected to AVDD,
+ If this node is not mentioned or if the value is incorrect, then MicBias
+ is powered down.
+- AVDD-supply, IOVDD-supply, DRVDD-supply, DVDD-supply : power supplies for the
+ device as covered in Documentation/devicetree/bindings/regulator/regulator.txt
+
+CODEC output pins:
+ * LLOUT
+ * RLOUT
+ * MONO_LOUT
+ * HPLOUT
+ * HPROUT
+ * HPLCOM
+ * HPRCOM
+
+CODEC input pins:
+ * MIC3L
+ * MIC3R
+ * LINE1L
+ * LINE2L
+ * LINE1R
+ * LINE2R
+
+The pins can be used in referring sound node's audio-routing property.
+
+Example:
+
+tlv320aic3x: tlv320aic3x@1b {
+ compatible = "ti,tlv320aic3x";
+ reg = <0x1b>;
+
+ AVDD-supply = <&regulator>;
+ IOVDD-supply = <&regulator>;
+ DRVDD-supply = <&regulator>;
+ DVDD-supply = <&regulator>;
+};
diff --git a/dts/Bindings/sound/tpa6130a2.txt b/dts/Bindings/sound/tpa6130a2.txt
new file mode 100644
index 0000000..6dfa740
--- /dev/null
+++ b/dts/Bindings/sound/tpa6130a2.txt
@@ -0,0 +1,27 @@
+Texas Instruments - tpa6130a2 Codec module
+
+The tpa6130a2 serial control bus communicates through I2C protocols
+
+Required properties:
+
+- compatible - "string" - One of:
+ "ti,tpa6130a2" - TPA6130A2
+ "ti,tpa6140a2" - TPA6140A2
+
+
+- reg - <int> - I2C slave address
+
+- Vdd-supply - <phandle> - power supply regulator
+
+Optional properties:
+
+- power-gpio - gpio pin to power the device
+
+Example:
+
+tpa6130a2: tpa6130a2@60 {
+ compatible = "ti,tpa6130a2";
+ reg = <0x60>;
+ Vdd-supply = <&vmmc2>;
+ power-gpio = <&gpio4 2 GPIO_ACTIVE_HIGH>;
+};
diff --git a/dts/Bindings/sound/ux500-mop500.txt b/dts/Bindings/sound/ux500-mop500.txt
new file mode 100644
index 0000000..48e071c
--- /dev/null
+++ b/dts/Bindings/sound/ux500-mop500.txt
@@ -0,0 +1,39 @@
+* MOP500 Audio Machine Driver
+
+This node is responsible for linking together all ux500 Audio Driver components.
+
+Required properties:
+ - compatible : "stericsson,snd-soc-mop500"
+
+Non-standard properties:
+ - stericsson,cpu-dai : Phandle to the CPU-side DAI
+ - stericsson,audio-codec : Phandle to the Audio CODEC
+ - stericsson,card-name : Over-ride default card name
+
+Example:
+
+ sound {
+ compatible = "stericsson,snd-soc-mop500";
+
+ stericsson,cpu-dai = <&msp1 &msp3>;
+ stericsson,audio-codec = <&codec>;
+ };
+
+ msp1: msp@80124000 {
+ compatible = "stericsson,ux500-msp-i2s";
+ reg = <0x80124000 0x1000>;
+ interrupts = <0 62 0x4>;
+ v-ape-supply = <&db8500_vape_reg>;
+ };
+
+ msp3: msp@80125000 {
+ compatible = "stericsson,ux500-msp-i2s";
+ reg = <0x80125000 0x1000>;
+ interrupts = <0 62 0x4>;
+ v-ape-supply = <&db8500_vape_reg>;
+ };
+
+ codec: ab8500-codec {
+ compatible = "stericsson,ab8500-codec";
+ stericsson,earpeice-cmv = <950>; /* Units in mV. */
+ };
diff --git a/dts/Bindings/sound/ux500-msp.txt b/dts/Bindings/sound/ux500-msp.txt
new file mode 100644
index 0000000..99acd9c
--- /dev/null
+++ b/dts/Bindings/sound/ux500-msp.txt
@@ -0,0 +1,43 @@
+* ux500 MSP (CPU-side Digital Audio Interface)
+
+Required properties:
+ - compatible :"stericsson,ux500-msp-i2s"
+ - reg : Physical base address and length of the device's registers.
+
+Optional properties:
+ - interrupts : The interrupt output from the device.
+ - interrupt-parent : The parent interrupt controller.
+ - <name>-supply : Phandle to the regulator <name> supply
+
+Example:
+
+ sound {
+ compatible = "stericsson,snd-soc-mop500";
+
+ stericsson,platform-pcm-dma = <&pcm>;
+ stericsson,cpu-dai = <&msp1 &msp3>;
+ stericsson,audio-codec = <&codec>;
+ };
+
+ pcm: ux500-pcm {
+ compatible = "stericsson,ux500-pcm";
+ };
+
+ msp1: msp@80124000 {
+ compatible = "stericsson,ux500-msp-i2s";
+ reg = <0x80124000 0x1000>;
+ interrupts = <0 62 0x4>;
+ v-ape-supply = <&db8500_vape_reg>;
+ };
+
+ msp3: msp@80125000 {
+ compatible = "stericsson,ux500-msp-i2s";
+ reg = <0x80125000 0x1000>;
+ interrupts = <0 62 0x4>;
+ v-ape-supply = <&db8500_vape_reg>;
+ };
+
+ codec: ab8500-codec {
+ compatible = "stericsson,ab8500-codec";
+ stericsson,earpeice-cmv = <950>; /* Units in mV. */
+ };
diff --git a/dts/Bindings/sound/widgets.txt b/dts/Bindings/sound/widgets.txt
new file mode 100644
index 0000000..b6de5ba
--- /dev/null
+++ b/dts/Bindings/sound/widgets.txt
@@ -0,0 +1,20 @@
+Widgets:
+
+This mainly specifies audio off-codec DAPM widgets.
+
+Each entry is a pair of strings in DT:
+
+ "template-wname", "user-supplied-wname"
+
+The "template-wname" being the template widget name and currently includes:
+"Microphone", "Line", "Headphone" and "Speaker".
+
+The "user-supplied-wname" being the user specified widget name.
+
+For instance:
+ simple-audio-widgets =
+ "Microphone", "Microphone Jack",
+ "Line", "Line In Jack",
+ "Line", "Line Out Jack",
+ "Headphone", "Headphone Jack",
+ "Speaker", "Speaker External";
diff --git a/dts/Bindings/sound/wm8510.txt b/dts/Bindings/sound/wm8510.txt
new file mode 100644
index 0000000..fa1a32b
--- /dev/null
+++ b/dts/Bindings/sound/wm8510.txt
@@ -0,0 +1,18 @@
+WM8510 audio CODEC
+
+This device supports both I2C and SPI (configured with pin strapping
+on the board).
+
+Required properties:
+
+ - compatible : "wlf,wm8510"
+
+ - reg : the I2C address of the device for I2C, the chip select
+ number for SPI.
+
+Example:
+
+codec: wm8510@1a {
+ compatible = "wlf,wm8510";
+ reg = <0x1a>;
+};
diff --git a/dts/Bindings/sound/wm8523.txt b/dts/Bindings/sound/wm8523.txt
new file mode 100644
index 0000000..0474618
--- /dev/null
+++ b/dts/Bindings/sound/wm8523.txt
@@ -0,0 +1,16 @@
+WM8523 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+ - compatible : "wlf,wm8523"
+
+ - reg : the I2C address of the device.
+
+Example:
+
+codec: wm8523@1a {
+ compatible = "wlf,wm8523";
+ reg = <0x1a>;
+};
diff --git a/dts/Bindings/sound/wm8580.txt b/dts/Bindings/sound/wm8580.txt
new file mode 100644
index 0000000..7d9821f
--- /dev/null
+++ b/dts/Bindings/sound/wm8580.txt
@@ -0,0 +1,16 @@
+WM8580 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+ - compatible : "wlf,wm8580"
+
+ - reg : the I2C address of the device.
+
+Example:
+
+codec: wm8580@1a {
+ compatible = "wlf,wm8580";
+ reg = <0x1a>;
+};
diff --git a/dts/Bindings/sound/wm8711.txt b/dts/Bindings/sound/wm8711.txt
new file mode 100644
index 0000000..8ed9998
--- /dev/null
+++ b/dts/Bindings/sound/wm8711.txt
@@ -0,0 +1,18 @@
+WM8711 audio CODEC
+
+This device supports both I2C and SPI (configured with pin strapping
+on the board).
+
+Required properties:
+
+ - compatible : "wlf,wm8711"
+
+ - reg : the I2C address of the device for I2C, the chip select
+ number for SPI.
+
+Example:
+
+codec: wm8711@1a {
+ compatible = "wlf,wm8711";
+ reg = <0x1a>;
+};
diff --git a/dts/Bindings/sound/wm8728.txt b/dts/Bindings/sound/wm8728.txt
new file mode 100644
index 0000000..a8b5c36
--- /dev/null
+++ b/dts/Bindings/sound/wm8728.txt
@@ -0,0 +1,18 @@
+WM8728 audio CODEC
+
+This device supports both I2C and SPI (configured with pin strapping
+on the board).
+
+Required properties:
+
+ - compatible : "wlf,wm8728"
+
+ - reg : the I2C address of the device for I2C, the chip select
+ number for SPI.
+
+Example:
+
+codec: wm8728@1a {
+ compatible = "wlf,wm8728";
+ reg = <0x1a>;
+};
diff --git a/dts/Bindings/sound/wm8731.txt b/dts/Bindings/sound/wm8731.txt
new file mode 100644
index 0000000..236690e
--- /dev/null
+++ b/dts/Bindings/sound/wm8731.txt
@@ -0,0 +1,27 @@
+WM8731 audio CODEC
+
+This device supports both I2C and SPI (configured with pin strapping
+on the board).
+
+Required properties:
+
+ - compatible : "wlf,wm8731"
+
+ - reg : the I2C address of the device for I2C, the chip select
+ number for SPI.
+
+Example:
+
+codec: wm8731@1a {
+ compatible = "wlf,wm8731";
+ reg = <0x1a>;
+};
+
+Available audio endpoints for an audio-routing table:
+ * LOUT: Left Channel Line Output
+ * ROUT: Right Channel Line Output
+ * LHPOUT: Left Channel Headphone Output
+ * RHPOUT: Right Channel Headphone Output
+ * LLINEIN: Left Channel Line Input
+ * RLINEIN: Right Channel Line Input
+ * MICIN: Microphone Input
diff --git a/dts/Bindings/sound/wm8737.txt b/dts/Bindings/sound/wm8737.txt
new file mode 100644
index 0000000..4bc2cea
--- /dev/null
+++ b/dts/Bindings/sound/wm8737.txt
@@ -0,0 +1,18 @@
+WM8737 audio CODEC
+
+This device supports both I2C and SPI (configured with pin strapping
+on the board).
+
+Required properties:
+
+ - compatible : "wlf,wm8737"
+
+ - reg : the I2C address of the device for I2C, the chip select
+ number for SPI.
+
+Example:
+
+codec: wm8737@1a {
+ compatible = "wlf,wm8737";
+ reg = <0x1a>;
+};
diff --git a/dts/Bindings/sound/wm8741.txt b/dts/Bindings/sound/wm8741.txt
new file mode 100644
index 0000000..74bda58
--- /dev/null
+++ b/dts/Bindings/sound/wm8741.txt
@@ -0,0 +1,18 @@
+WM8741 audio CODEC
+
+This device supports both I2C and SPI (configured with pin strapping
+on the board).
+
+Required properties:
+
+ - compatible : "wlf,wm8741"
+
+ - reg : the I2C address of the device for I2C, the chip select
+ number for SPI.
+
+Example:
+
+codec: wm8741@1a {
+ compatible = "wlf,wm8741";
+ reg = <0x1a>;
+};
diff --git a/dts/Bindings/sound/wm8750.txt b/dts/Bindings/sound/wm8750.txt
new file mode 100644
index 0000000..8db239f
--- /dev/null
+++ b/dts/Bindings/sound/wm8750.txt
@@ -0,0 +1,18 @@
+WM8750 and WM8987 audio CODECs
+
+These devices support both I2C and SPI (configured with pin strapping
+on the board).
+
+Required properties:
+
+ - compatible : "wlf,wm8750" or "wlf,wm8987"
+
+ - reg : the I2C address of the device for I2C, the chip select
+ number for SPI.
+
+Example:
+
+codec: wm8750@1a {
+ compatible = "wlf,wm8750";
+ reg = <0x1a>;
+};
diff --git a/dts/Bindings/sound/wm8753.txt b/dts/Bindings/sound/wm8753.txt
new file mode 100644
index 0000000..8eee612
--- /dev/null
+++ b/dts/Bindings/sound/wm8753.txt
@@ -0,0 +1,40 @@
+WM8753 audio CODEC
+
+This device supports both I2C and SPI (configured with pin strapping
+on the board).
+
+Required properties:
+
+ - compatible : "wlf,wm8753"
+
+ - reg : the I2C address of the device for I2C, the chip select
+ number for SPI.
+
+Pins on the device (for linking into audio routes):
+
+ * LOUT1
+ * LOUT2
+ * ROUT1
+ * ROUT2
+ * MONO1
+ * MONO2
+ * OUT3
+ * OUT4
+ * LINE1
+ * LINE2
+ * RXP
+ * RXN
+ * ACIN
+ * ACOP
+ * MIC1N
+ * MIC1
+ * MIC2N
+ * MIC2
+ * Mic Bias
+
+Example:
+
+codec: wm8753@1a {
+ compatible = "wlf,wm8753";
+ reg = <0x1a>;
+};
diff --git a/dts/Bindings/sound/wm8770.txt b/dts/Bindings/sound/wm8770.txt
new file mode 100644
index 0000000..866e00c
--- /dev/null
+++ b/dts/Bindings/sound/wm8770.txt
@@ -0,0 +1,16 @@
+WM8770 audio CODEC
+
+This device supports SPI.
+
+Required properties:
+
+ - compatible : "wlf,wm8770"
+
+ - reg : the chip select number.
+
+Example:
+
+codec: wm8770@1 {
+ compatible = "wlf,wm8770";
+ reg = <1>;
+};
diff --git a/dts/Bindings/sound/wm8776.txt b/dts/Bindings/sound/wm8776.txt
new file mode 100644
index 0000000..3b9ca49
--- /dev/null
+++ b/dts/Bindings/sound/wm8776.txt
@@ -0,0 +1,18 @@
+WM8776 audio CODEC
+
+This device supports both I2C and SPI (configured with pin strapping
+on the board).
+
+Required properties:
+
+ - compatible : "wlf,wm8776"
+
+ - reg : the I2C address of the device for I2C, the chip select
+ number for SPI.
+
+Example:
+
+codec: wm8776@1a {
+ compatible = "wlf,wm8776";
+ reg = <0x1a>;
+};
diff --git a/dts/Bindings/sound/wm8804.txt b/dts/Bindings/sound/wm8804.txt
new file mode 100644
index 0000000..4d3a56f
--- /dev/null
+++ b/dts/Bindings/sound/wm8804.txt
@@ -0,0 +1,18 @@
+WM8804 audio CODEC
+
+This device supports both I2C and SPI (configured with pin strapping
+on the board).
+
+Required properties:
+
+ - compatible : "wlf,wm8804"
+
+ - reg : the I2C address of the device for I2C, the chip select
+ number for SPI.
+
+Example:
+
+codec: wm8804@1a {
+ compatible = "wlf,wm8804";
+ reg = <0x1a>;
+};
diff --git a/dts/Bindings/sound/wm8903.txt b/dts/Bindings/sound/wm8903.txt
new file mode 100644
index 0000000..94ec32c
--- /dev/null
+++ b/dts/Bindings/sound/wm8903.txt
@@ -0,0 +1,69 @@
+WM8903 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+ - compatible : "wlf,wm8903"
+
+ - reg : the I2C address of the device.
+
+ - gpio-controller : Indicates this device is a GPIO controller.
+
+ - #gpio-cells : Should be two. The first cell is the pin number and the
+ second cell is used to specify optional parameters (currently unused).
+
+Optional properties:
+
+ - interrupts : The interrupt line the codec is connected to.
+
+ - micdet-cfg : Default register value for R6 (Mic Bias). If absent, the
+ default is 0.
+
+ - micdet-delay : The debounce delay for microphone detection in mS. If
+ absent, the default is 100.
+
+ - gpio-cfg : A list of GPIO configuration register values. The list must
+ be 5 entries long. If absent, no configuration of these registers is
+ performed. If any entry has the value 0xffffffff, that GPIO's
+ configuration will not be modified.
+
+Pins on the device (for linking into audio routes):
+
+ * IN1L
+ * IN1R
+ * IN2L
+ * IN2R
+ * IN3L
+ * IN3R
+ * DMICDAT
+ * HPOUTL
+ * HPOUTR
+ * LINEOUTL
+ * LINEOUTR
+ * LOP
+ * LON
+ * ROP
+ * RON
+ * MICBIAS
+
+Example:
+
+codec: wm8903@1a {
+ compatible = "wlf,wm8903";
+ reg = <0x1a>;
+ interrupts = < 347 >;
+
+ gpio-controller;
+ #gpio-cells = <2>;
+
+ micdet-cfg = <0>;
+ micdet-delay = <100>;
+ gpio-cfg = <
+ 0x0600 /* DMIC_LR, output */
+ 0x0680 /* DMIC_DAT, input */
+ 0x0000 /* GPIO, output, low */
+ 0x0200 /* Interrupt, output */
+ 0x01a0 /* BCLK, input, active high */
+ >;
+};
diff --git a/dts/Bindings/sound/wm8962.txt b/dts/Bindings/sound/wm8962.txt
new file mode 100644
index 0000000..7f82b59
--- /dev/null
+++ b/dts/Bindings/sound/wm8962.txt
@@ -0,0 +1,39 @@
+WM8962 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+ - compatible : "wlf,wm8962"
+
+ - reg : the I2C address of the device.
+
+Optional properties:
+ - spk-mono: This is a boolean property. If present, the SPK_MONO bit
+ of R51 (Class D Control 2) gets set, indicating that the speaker is
+ in mono mode.
+
+ - mic-cfg : Default register value for R48 (Additional Control 4).
+ If absent, the default should be the register default.
+
+ - gpio-cfg : A list of GPIO configuration register values. The list must
+ be 6 entries long. If absent, no configuration of these registers is
+ performed. And note that only the value within [0x0, 0xffff] is valid.
+ Any other value is regarded as setting the GPIO register by its reset
+ value 0x0.
+
+Example:
+
+codec: wm8962@1a {
+ compatible = "wlf,wm8962";
+ reg = <0x1a>;
+
+ gpio-cfg = <
+ 0x0000 /* 0:Default */
+ 0x0000 /* 1:Default */
+ 0x0013 /* 2:FN_DMICCLK */
+ 0x0000 /* 3:Default */
+ 0x8014 /* 4:FN_DMICCDAT */
+ 0x0000 /* 5:Default */
+ >;
+};
diff --git a/dts/Bindings/sound/wm8994.txt b/dts/Bindings/sound/wm8994.txt
new file mode 100644
index 0000000..e045e90
--- /dev/null
+++ b/dts/Bindings/sound/wm8994.txt
@@ -0,0 +1,78 @@
+WM1811/WM8994/WM8958 audio CODEC
+
+These devices support both I2C and SPI (configured with pin strapping
+on the board).
+
+Required properties:
+
+ - compatible : One of "wlf,wm1811", "wlf,wm8994" or "wlf,wm8958".
+
+ - reg : the I2C address of the device for I2C, the chip select
+ number for SPI.
+
+ - gpio-controller : Indicates this device is a GPIO controller.
+ - #gpio-cells : Must be 2. The first cell is the pin number and the
+ second cell is used to specify optional parameters (currently unused).
+
+ - AVDD2-supply, DBVDD1-supply, DBVDD2-supply, DBVDD3-supply, CPVDD-supply,
+ SPKVDD1-supply, SPKVDD2-supply : power supplies for the device, as covered
+ in Documentation/devicetree/bindings/regulator/regulator.txt
+
+Optional properties:
+
+ - interrupts : The interrupt line the IRQ signal for the device is
+ connected to. This is optional, if it is not connected then none
+ of the interrupt related properties should be specified.
+ - interrupt-controller : These devices contain interrupt controllers
+ and may provide interrupt services to other devices if they have an
+ interrupt line connected.
+ - interrupt-parent : The parent interrupt controller.
+ - #interrupt-cells: the number of cells to describe an IRQ, this should be 2.
+ The first cell is the IRQ number.
+ The second cell is the flags, encoded as the trigger masks from
+ Documentation/devicetree/bindings/interrupts.txt
+
+ - clocks : A list of up to two phandle and clock specifier pairs
+ - clock-names : A list of clock names sorted in the same order as clocks.
+ Valid clock names are "MCLK1" and "MCLK2".
+
+ - wlf,gpio-cfg : A list of GPIO configuration register values. If absent,
+ no configuration of these registers is performed. If any value is
+ over 0xffff then the register will be left as default. If present 11
+ values must be supplied.
+
+ - wlf,micbias-cfg : Two MICBIAS register values for WM1811 or
+ WM8958. If absent the register defaults will be used.
+
+ - wlf,ldo1ena : GPIO specifier for control of LDO1ENA input to device.
+ - wlf,ldo2ena : GPIO specifier for control of LDO2ENA input to device.
+
+ - wlf,lineout1-se : If present LINEOUT1 is in single ended mode.
+ - wlf,lineout2-se : If present LINEOUT2 is in single ended mode.
+
+ - wlf,lineout1-feedback : If present LINEOUT1 has common mode feedback
+ connected.
+ - wlf,lineout2-feedback : If present LINEOUT2 has common mode feedback
+ connected.
+
+ - wlf,ldoena-always-driven : If present LDOENA is always driven.
+
+Example:
+
+codec: wm8994@1a {
+ compatible = "wlf,wm8994";
+ reg = <0x1a>;
+
+ gpio-controller;
+ #gpio-cells = <2>;
+
+ lineout1-se;
+
+ AVDD2-supply = <&regulator>;
+ CPVDD-supply = <&regulator>;
+ DBVDD1-supply = <&regulator>;
+ DBVDD2-supply = <&regulator>;
+ DBVDD3-supply = <&regulator>;
+ SPKVDD1-supply = <&regulator>;
+ SPKVDD2-supply = <&regulator>;
+};