diff options
Diffstat (limited to 'dts/Bindings/sound')
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 0000000000..547a49b56a --- /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 0000000000..5875ca459e --- /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 0000000000..46f3449653 --- /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 0000000000..b902ee39cf --- /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 0000000000..934fa02754 --- /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 0000000000..623d4e70ae --- /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 0000000000..dc3914fe6c --- /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 0000000000..ffd886d110 --- /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 0000000000..bf984d2386 --- /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 0000000000..9c5a9947b6 --- /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 0000000000..0720857089 --- /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 0000000000..8bbe50c884 --- /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 0000000000..65783de0ae --- /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 0000000000..6b222f9b8e --- /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 0000000000..e2cd1d7539 --- /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 0000000000..bc03c9312a --- /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 0000000000..80ae910dbf --- /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 0000000000..f631fbca62 --- /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 = <®_audio>; + VD-supply = <®_audio>; + VLS-supply = <®_audio>; + VLC-supply = <®_audio>; +}; diff --git a/dts/Bindings/sound/da9055.txt b/dts/Bindings/sound/da9055.txt new file mode 100644 index 0000000000..ed1b7cc6f2 --- /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 0000000000..963e100514 --- /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 0000000000..569b26c4a8 --- /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 0000000000..0d7985c864 --- /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 0000000000..aeb8c4a0b8 --- /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 0000000000..3e9e82c8ea --- /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 0000000000..3aa4a8f528 --- /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 0000000000..98611a6761 --- /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 0000000000..31af7bca30 --- /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 0000000000..e4acdd891e --- /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 0000000000..7d13479f9c --- /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 0000000000..f49450a878 --- /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 0000000000..f88a00e54c --- /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 0000000000..e4c8b36dcf --- /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 0000000000..74c9ba6c28 --- /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 0000000000..551fbb8348 --- /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 0000000000..cb8c07c81c --- /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 0000000000..601c518edd --- /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 0000000000..7ba07a118e --- /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 0000000000..57f40f9345 --- /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 0000000000..9c7c55c713 --- /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 0000000000..7788808dcd --- /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 0000000000..ef1fe73582 --- /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 0000000000..96f6a57dd6 --- /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 0000000000..b795d28281 --- /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 0000000000..436f6cd9d0 --- /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 0000000000..eaf00102d9 --- /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 0000000000..6de3a7ee4e --- /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 0000000000..dc30c6bfbe --- /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 0000000000..946e2ac460 --- /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 0000000000..0c113ffe38 --- /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 0000000000..fd40c852d7 --- /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 0000000000..fd8105f189 --- /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 0000000000..17cce44904 --- /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 0000000000..0741dff048 --- /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 0000000000..1ab6bc8404 --- /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 0000000000..970ba1ed57 --- /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 0000000000..faff75e645 --- /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 = <®_3v3_analog>; + DVDD-supply = <®_1v8>; + CPVDD-supply = <®_3v3>; + }; diff --git a/dts/Bindings/sound/renesas,fsi.txt b/dts/Bindings/sound/renesas,fsi.txt new file mode 100644 index 0000000000..c5be003f41 --- /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 0000000000..a44e9179fa --- /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 0000000000..068a1141b0 --- /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 0000000000..4686646fb1 --- /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 0000000000..7386d444ad --- /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 0000000000..955df60a11 --- /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 0000000000..131aa2ad7f --- /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 0000000000..062f5ec36f --- /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 0000000000..1f66de3c8f --- /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 0000000000..c88882ca37 --- /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 0000000000..80152a87f2 --- /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 0000000000..80f807bf8a --- /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 0000000000..55a85841dd --- /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 0000000000..59381a778c --- /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 0000000000..6a2c84247f --- /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 0000000000..4df17185ab --- /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 0000000000..d2866a0d6a --- /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 0000000000..74c66dee3e --- /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 = <®ulator>; + SPRVDD-supply = <®ulator>; + SPLVDD-supply = <®ulator>; + AVDD-supply = <®ulator>; + IOVDD-supply = <®ulator>; + DVDD-supply = <®ulator>; +}; diff --git a/dts/Bindings/sound/tlv320aic32x4.txt b/dts/Bindings/sound/tlv320aic32x4.txt new file mode 100644 index 0000000000..5e2741af27 --- /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 0000000000..5e6040c2c2 --- /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 = <®ulator>; + IOVDD-supply = <®ulator>; + DRVDD-supply = <®ulator>; + DVDD-supply = <®ulator>; +}; diff --git a/dts/Bindings/sound/tpa6130a2.txt b/dts/Bindings/sound/tpa6130a2.txt new file mode 100644 index 0000000000..6dfa740e4b --- /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 0000000000..48e071c96b --- /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 0000000000..99acd9c774 --- /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 0000000000..b6de5ba3b2 --- /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 0000000000..fa1a32b855 --- /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 0000000000..04746186b2 --- /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 0000000000..7d9821f348 --- /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 0000000000..8ed9998cd2 --- /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 0000000000..a8b5c3668e --- /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 0000000000..236690e99b --- /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 0000000000..4bc2cea3b1 --- /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 0000000000..74bda58c1b --- /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 0000000000..8db239fd5e --- /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 0000000000..8eee612821 --- /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 0000000000..866e00ca15 --- /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 0000000000..3b9ca49abc --- /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 0000000000..4d3a56f38a --- /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 0000000000..94ec32c194 --- /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 0000000000..7f82b59ec8 --- /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 0000000000..e045e90a09 --- /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 = <®ulator>; + CPVDD-supply = <®ulator>; + DBVDD1-supply = <®ulator>; + DBVDD2-supply = <®ulator>; + DBVDD3-supply = <®ulator>; + SPKVDD1-supply = <®ulator>; + SPKVDD2-supply = <®ulator>; +}; |