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| author | Venkata Narendra Kumar Gutta <vgutta@quicinc.com> | 2019-04-24 12:36:05 -0700 |
|---|---|---|
| committer | Venkata Narendra Kumar Gutta <vgutta@quicinc.com> | 2019-05-02 16:41:24 -0700 |
| commit | ed241f132720397b8abac31ca170a274ea6e54e5 (patch) | |
| tree | 892aeadc1cc2fc0a47d982a533f8112d96866d37 /bindings/sound | |
| parent | 1f62b25b429becfcc7d8608f389705b3e6ef0e20 (diff) | |
| download | devicetree-ed241f132720397b8abac31ca170a274ea6e54e5.tar.gz | |
dt-bindings: Add devicetree bindings to devicetree project
Add devicetree bindings snapshot to the devicetree project.
This snapshot is taken as of
'commit f3dd4aaeb34438c877ccd42f5a48ccd554dd765a (Merge
"platform: qpnp-revid: Add REVID support for PM7250B")' of
the kernel project.
Change-Id: I5e0ec0eae63ff9c071b2924bd84c5b20d3f6554d
Diffstat (limited to 'bindings/sound')
262 files changed, 14417 insertions, 0 deletions
diff --git a/bindings/sound/ac97-bus.txt b/bindings/sound/ac97-bus.txt new file mode 100644 index 00000000..103c428f --- /dev/null +++ b/bindings/sound/ac97-bus.txt @@ -0,0 +1,32 @@ +Generic AC97 Device Properties + +This documents describes the devicetree bindings for an ac97 controller child +node describing ac97 codecs. + +Required properties: +-compatible : Must be "ac97,vendor_id1,vendor_id2 + The ids shall be the 4 characters hexadecimal encoding, such as + given by "%04x" formatting of printf +-reg : Must be the ac97 codec number, between 0 and 3 + +Example: +ac97: sound@40500000 { + compatible = "marvell,pxa270-ac97"; + reg = < 0x40500000 0x1000 >; + interrupts = <14>; + reset-gpios = <&gpio 95 GPIO_ACTIVE_HIGH>; + #sound-dai-cells = <1>; + pinctrl-names = "default"; + pinctrl-0 = < &pinctrl_ac97_default >; + clocks = <&clks CLK_AC97>, <&clks CLK_AC97CONF>; + clock-names = "AC97CLK", "AC97CONFCLK"; + + #address-cells = <1>; + #size-cells = <0>; + audio-codec@0 { + reg = <0>; + compatible = "ac97,574d,4c13"; + clocks = <&fixed_wm9713_clock>; + clock-names = "ac97_clk"; + } +}; diff --git a/bindings/sound/adi,adau1701.txt b/bindings/sound/adi,adau1701.txt new file mode 100644 index 00000000..0d1128ce --- /dev/null +++ b/bindings/sound/adi,adau1701.txt @@ -0,0 +1,39 @@ +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'. + - avdd-supply: Power supply for AVDD, providing 3.3V + - dvdd-supply: Power supply for DVDD, providing 3.3V + +Examples: + + i2c_bus { + adau1701@34 { + compatible = "adi,adau1701"; + reg = <0x34>; + reset-gpio = <&gpio 23 0>; + avdd-supply = <&vdd_3v3_reg>; + dvdd-supply = <&vdd_3v3_reg>; + 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/bindings/sound/adi,adau17x1.txt b/bindings/sound/adi,adau17x1.txt new file mode 100644 index 00000000..1447dec2 --- /dev/null +++ b/bindings/sound/adi,adau17x1.txt @@ -0,0 +1,32 @@ +Analog Devices ADAU1361/ADAU1461/ADAU1761/ADAU1961/ADAU1381/ADAU1781 + +Required properties: + + - compatible: Should contain one of the following: + "adi,adau1361" + "adi,adau1461" + "adi,adau1761" + "adi,adau1961" + "adi,adau1381" + "adi,adau1781" + + - reg: The i2c address. Value depends on the state of ADDR0 + and ADDR1, as wired in hardware. + +Optional properties: + - clock-names: If provided must be "mclk". + - clocks: phandle + clock-specifiers for the clock that provides + the audio master clock for the device. + +Examples: +#include <dt-bindings/sound/adau17x1.h> + + i2c_bus { + adau1361@38 { + compatible = "adi,adau1761"; + reg = <0x38>; + + clock-names = "mclk"; + clocks = <&audio_clock>; + }; + }; diff --git a/bindings/sound/adi,adau7002.txt b/bindings/sound/adi,adau7002.txt new file mode 100644 index 00000000..f144ee1a --- /dev/null +++ b/bindings/sound/adi,adau7002.txt @@ -0,0 +1,19 @@ +Analog Devices ADAU7002 Stereo PDM-to-I2S/TDM Converter + +Required properties: + + - compatible: Must be "adi,adau7002" + +Optional properties: + + - IOVDD-supply: Phandle and specifier for the power supply providing the IOVDD + supply as covered in Documentation/devicetree/bindings/regulator/regulator.txt + + If this property is not present it is assumed that the supply pin is + hardwired to always on. + +Example: + adau7002: pdm-to-i2s { + compatible = "adi,adau7002"; + IOVDD-supply = <&supply>; + }; diff --git a/bindings/sound/adi,axi-i2s.txt b/bindings/sound/adi,axi-i2s.txt new file mode 100644 index 00000000..4248b662 --- /dev/null +++ b/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@77600000 { + 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/bindings/sound/adi,axi-spdif-tx.txt b/bindings/sound/adi,axi-spdif-tx.txt new file mode 100644 index 00000000..7b664e7c --- /dev/null +++ b/bindings/sound/adi,axi-spdif-tx.txt @@ -0,0 +1,30 @@ +ADI AXI-SPDIF controller + +Required properties: + - compatible : Must be "adi,axi-spdif-tx-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@77400000 { + 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/bindings/sound/adi,ssm2305.txt b/bindings/sound/adi,ssm2305.txt new file mode 100644 index 00000000..a9c9d83c --- /dev/null +++ b/bindings/sound/adi,ssm2305.txt @@ -0,0 +1,14 @@ +Analog Devices SSM2305 Speaker Amplifier +======================================== + +Required properties: + - compatible : "adi,ssm2305" + - shutdown-gpios : The gpio connected to the shutdown pin. + The gpio signal is ACTIVE_LOW. + +Example: + +ssm2305: analog-amplifier { + compatible = "adi,ssm2305"; + shutdown-gpios = <&gpio3 20 GPIO_ACTIVE_LOW>; +}; diff --git a/bindings/sound/adi,ssm2602.txt b/bindings/sound/adi,ssm2602.txt new file mode 100644 index 00000000..3b3302fe --- /dev/null +++ b/bindings/sound/adi,ssm2602.txt @@ -0,0 +1,19 @@ +Analog Devices SSM2602, SSM2603 and SSM2604 I2S audio CODEC devices + +SSM2602 support both I2C and SPI as the configuration interface, +the selection is made by the MODE strap-in pin. +SSM2603 and SSM2604 only support I2C as the configuration interface. + +Required properties: + + - compatible : One of "adi,ssm2602", "adi,ssm2603" or "adi,ssm2604" + + - reg : the I2C address of the device for I2C, the chip select + number for SPI. + + Example: + + ssm2602: ssm2602@1a { + compatible = "adi,ssm2602"; + reg = <0x1a>; + }; diff --git a/bindings/sound/ak4104.txt b/bindings/sound/ak4104.txt new file mode 100644 index 00000000..deca5e18 --- /dev/null +++ b/bindings/sound/ak4104.txt @@ -0,0 +1,25 @@ +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 + + - vdd-supply : A regulator node, providing 2.7V - 3.6V + +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>; + vdd-supply = <&vdd_3v3_reg>; +}; diff --git a/bindings/sound/ak4458.txt b/bindings/sound/ak4458.txt new file mode 100644 index 00000000..7839be78 --- /dev/null +++ b/bindings/sound/ak4458.txt @@ -0,0 +1,23 @@ +AK4458 audio DAC + +This device supports I2C mode. + +Required properties: + +- compatible : "asahi-kasei,ak4458" +- reg : The I2C address of the device for I2C + +Optional properties: +- reset-gpios: A GPIO specifier for the power down & reset pin +- mute-gpios: A GPIO specifier for the soft mute pin + +Example: + +&i2c { + ak4458: dac@10 { + compatible = "asahi-kasei,ak4458"; + reg = <0x10>; + reset-gpios = <&gpio1 10 GPIO_ACTIVE_LOW> + mute-gpios = <&gpio1 11 GPIO_ACTIVE_HIGH> + }; +}; diff --git a/bindings/sound/ak4554.txt b/bindings/sound/ak4554.txt new file mode 100644 index 00000000..934fa027 --- /dev/null +++ b/bindings/sound/ak4554.txt @@ -0,0 +1,11 @@ +AK4554 ADC/DAC + +Required properties: + + - compatible : "asahi-kasei,ak4554" + +Example: + +ak4554-adc-dac { + compatible = "asahi-kasei,ak4554"; +}; diff --git a/bindings/sound/ak4613.txt b/bindings/sound/ak4613.txt new file mode 100644 index 00000000..49a2e74f --- /dev/null +++ b/bindings/sound/ak4613.txt @@ -0,0 +1,27 @@ +AK4613 I2C transmitter + +This device supports I2C mode only. + +Required properties: + +- compatible : "asahi-kasei,ak4613" +- reg : The chip select number on the I2C bus + +Optional properties: +- asahi-kasei,in1-single-end : Boolean. Indicate input / output pins are single-ended. +- asahi-kasei,in2-single-end rather than differential. +- asahi-kasei,out1-single-end +- asahi-kasei,out2-single-end +- asahi-kasei,out3-single-end +- asahi-kasei,out4-single-end +- asahi-kasei,out5-single-end +- asahi-kasei,out6-single-end + +Example: + +&i2c { + ak4613: ak4613@10 { + compatible = "asahi-kasei,ak4613"; + reg = <0x10>; + }; +}; diff --git a/bindings/sound/ak4642.txt b/bindings/sound/ak4642.txt new file mode 100644 index 00000000..58e48ee9 --- /dev/null +++ b/bindings/sound/ak4642.txt @@ -0,0 +1,37 @@ +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 + +Optional properties: + + - #clock-cells : common clock binding; shall be set to 0 + - clocks : common clock binding; MCKI clock + - clock-frequency : common clock binding; frequency of MCKO + - clock-output-names : common clock binding; MCKO clock name + +Example 1: + +&i2c { + ak4648: ak4648@12 { + compatible = "asahi-kasei,ak4642"; + reg = <0x12>; + }; +}; + +Example 2: + +&i2c { + ak4643: codec@12 { + compatible = "asahi-kasei,ak4643"; + reg = <0x12>; + #clock-cells = <0>; + clocks = <&audio_clock>; + clock-frequency = <12288000>; + clock-output-names = "ak4643_mcko"; + }; +}; diff --git a/bindings/sound/ak5386.txt b/bindings/sound/ak5386.txt new file mode 100644 index 00000000..ec3df3ab --- /dev/null +++ b/bindings/sound/ak5386.txt @@ -0,0 +1,23 @@ +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. + - va-supply : a regulator spec, providing 5.0V + - vd-supply : a regulator spec, providing 3.3V + +Example: + +spdif: ak5386@0 { + compatible = "asahi-kasei,ak5386"; + reset-gpio = <&gpio0 23>; + va-supply = <&vdd_5v0_reg>; + vd-supply = <&vdd_3v3_reg>; +}; diff --git a/bindings/sound/ak5558.txt b/bindings/sound/ak5558.txt new file mode 100644 index 00000000..7d67ca6c --- /dev/null +++ b/bindings/sound/ak5558.txt @@ -0,0 +1,22 @@ +AK5558 8 channel differential 32-bit delta-sigma ADC + +This device supports I2C mode only. + +Required properties: + +- compatible : "asahi-kasei,ak5558" +- reg : The I2C address of the device. + +Optional properties: + +- reset-gpios: A GPIO specifier for the power down & reset pin. + +Example: + +&i2c { + ak5558: adc@10 { + compatible = "asahi-kasei,ak5558"; + reg = <0x10>; + reset-gpios = <&gpio1 10 GPIO_ACTIVE_LOW>; + }; +}; diff --git a/bindings/sound/alc5623.txt b/bindings/sound/alc5623.txt new file mode 100644 index 00000000..26c86c98 --- /dev/null +++ b/bindings/sound/alc5623.txt @@ -0,0 +1,25 @@ +ALC5621/ALC5622/ALC5623 audio Codec + +Required properties: + + - compatible: "realtek,alc5623" + - reg: the I2C address of the device. + +Optional properties: + + - add-ctrl: Default register value for Reg-40h, Additional Control + Register. If absent or has the value of 0, the + register is untouched. + + - jack-det-ctrl: Default register value for Reg-5Ah, Jack Detect + Control Register. If absent or has value 0, the + register is untouched. + +Example: + + alc5621: alc5621@1a { + compatible = "alc5621"; + reg = <0x1a>; + add-ctrl = <0x3700>; + jack-det-ctrl = <0x4810>; + }; diff --git a/bindings/sound/alc5632.txt b/bindings/sound/alc5632.txt new file mode 100644 index 00000000..ffd886d1 --- /dev/null +++ b/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/bindings/sound/amlogic,axg-fifo.txt b/bindings/sound/amlogic,axg-fifo.txt new file mode 100644 index 00000000..3dfc2515 --- /dev/null +++ b/bindings/sound/amlogic,axg-fifo.txt @@ -0,0 +1,23 @@ +* Amlogic Audio FIFO controllers + +Required properties: +- compatible: 'amlogic,axg-toddr' or + 'amlogic,axg-frddr' +- reg: physical base address of the controller and length of memory + mapped region. +- interrupts: interrupt specifier for the fifo. +- clocks: phandle to the fifo peripheral clock provided by the audio + clock controller. +- resets: phandle to memory ARB line provided by the arb reset controller. +- #sound-dai-cells: must be 0. + +Example of FRDDR A on the A113 SoC: + +frddr_a: audio-controller@1c0 { + compatible = "amlogic,axg-frddr"; + reg = <0x0 0x1c0 0x0 0x1c>; + #sound-dai-cells = <0>; + interrupts = <GIC_SPI 88 IRQ_TYPE_EDGE_RISING>; + clocks = <&clkc_audio AUD_CLKID_FRDDR_A>; + resets = <&arb AXG_ARB_FRDDR_A>; +}; diff --git a/bindings/sound/amlogic,axg-sound-card.txt b/bindings/sound/amlogic,axg-sound-card.txt new file mode 100644 index 00000000..80b41129 --- /dev/null +++ b/bindings/sound/amlogic,axg-sound-card.txt @@ -0,0 +1,124 @@ +Amlogic AXG sound card: + +Required properties: + +- compatible: "amlogic,axg-sound-card" +- model : User specified audio sound card name, one string + +Optional properties: + +- audio-aux-devs : List of phandles pointing to auxiliary devices +- audio-widgets : Please refer to widgets.txt. +- audio-routing : A list of the connections between audio components. + +Subnodes: + +- dai-link: Container for dai-link level properties and the CODEC + sub-nodes. There should be at least one (and probably more) + subnode of this type. + +Required dai-link properties: + +- sound-dai: phandle and port of the CPU DAI. + +Required TDM Backend dai-link properties: +- dai-format : CPU/CODEC common audio format + +Optional TDM Backend dai-link properties: +- dai-tdm-slot-rx-mask-{0,1,2,3}: Receive direction slot masks +- dai-tdm-slot-tx-mask-{0,1,2,3}: Transmit direction slot masks + When omitted, mask is assumed to have to no + slots. A valid must have at one slot, so at + least one these mask should be provided with + an enabled slot. +- dai-tdm-slot-num : Please refer to tdm-slot.txt. + If omitted, slot number is set to accommodate the largest + mask provided. +- dai-tdm-slot-width : Please refer to tdm-slot.txt. default to 32 if omitted. +- mclk-fs : Multiplication factor between stream rate and mclk + +Backend dai-link subnodes: + +- codec: dai-link representing backend links should have at least one subnode. + One subnode for each codec of the dai-link. + dai-link representing frontend links have no codec, therefore have no + subnodes + +Required codec subnodes properties: + +- sound-dai: phandle and port of the CODEC DAI. + +Optional codec subnodes properties: + +- dai-tdm-slot-tx-mask : Please refer to tdm-slot.txt. +- dai-tdm-slot-rx-mask : Please refer to tdm-slot.txt. + +Example: + +sound { + compatible = "amlogic,axg-sound-card"; + model = "AXG-S420"; + audio-aux-devs = <&tdmin_a>, <&tdmout_c>; + audio-widgets = "Line", "Lineout", + "Line", "Linein", + "Speaker", "Speaker1 Left", + "Speaker", "Speaker1 Right"; + "Speaker", "Speaker2 Left", + "Speaker", "Speaker2 Right"; + audio-routing = "TDMOUT_C IN 0", "FRDDR_A OUT 2", + "SPDIFOUT IN 0", "FRDDR_A OUT 3", + "TDM_C Playback", "TDMOUT_C OUT", + "TDMIN_A IN 2", "TDM_C Capture", + "TDMIN_A IN 5", "TDM_C Loopback", + "TODDR_A IN 0", "TDMIN_A OUT", + "Lineout", "Lineout AOUTL", + "Lineout", "Lineout AOUTR", + "Speaker1 Left", "SPK1 OUT_A", + "Speaker2 Left", "SPK2 OUT_A", + "Speaker1 Right", "SPK1 OUT_B", + "Speaker2 Right", "SPK2 OUT_B", + "Linein AINL", "Linein", + "Linein AINR", "Linein"; + + dai-link@0 { + sound-dai = <&frddr_a>; + }; + + dai-link@1 { + sound-dai = <&toddr_a>; + }; + + dai-link@2 { + sound-dai = <&tdmif_c>; + dai-format = "i2s"; + dai-tdm-slot-tx-mask-2 = <1 1>; + dai-tdm-slot-tx-mask-3 = <1 1>; + dai-tdm-slot-rx-mask-1 = <1 1>; + mclk-fs = <256>; + + codec@0 { + sound-dai = <&lineout>; + }; + + codec@1 { + sound-dai = <&speaker_amp1>; + }; + + codec@2 { + sound-dai = <&speaker_amp2>; + }; + + codec@3 { + sound-dai = <&linein>; + }; + + }; + + dai-link@3 { + sound-dai = <&spdifout>; + + codec { + sound-dai = <&spdif_dit>; + }; + }; +}; diff --git a/bindings/sound/amlogic,axg-spdifout.txt b/bindings/sound/amlogic,axg-spdifout.txt new file mode 100644 index 00000000..521c38ad --- /dev/null +++ b/bindings/sound/amlogic,axg-spdifout.txt @@ -0,0 +1,20 @@ +* Amlogic Audio SPDIF Output + +Required properties: +- compatible: 'amlogic,axg-spdifout' +- clocks: list of clock phandle, one for each entry clock-names. +- clock-names: should contain the following: + * "pclk" : peripheral clock. + * "mclk" : master clock +- #sound-dai-cells: must be 0. + +Example on the A113 SoC: + +spdifout: audio-controller@480 { + compatible = "amlogic,axg-spdifout"; + reg = <0x0 0x480 0x0 0x50>; + #sound-dai-cells = <0>; + clocks = <&clkc_audio AUD_CLKID_SPDIFOUT>, + <&clkc_audio AUD_CLKID_SPDIFOUT_CLK>; + clock-names = "pclk", "mclk"; +}; diff --git a/bindings/sound/amlogic,axg-tdm-formatters.txt b/bindings/sound/amlogic,axg-tdm-formatters.txt new file mode 100644 index 00000000..1c1b7490 --- /dev/null +++ b/bindings/sound/amlogic,axg-tdm-formatters.txt @@ -0,0 +1,28 @@ +* Amlogic Audio TDM formatters + +Required properties: +- compatible: 'amlogic,axg-tdmin' or + 'amlogic,axg-tdmout' +- reg: physical base address of the controller and length of memory + mapped region. +- clocks: list of clock phandle, one for each entry clock-names. +- clock-names: should contain the following: + * "pclk" : peripheral clock. + * "sclk" : bit clock. + * "sclk_sel" : bit clock input multiplexer. + * "lrclk" : sample clock + * "lrclk_sel": sample clock input multiplexer + +Example of TDMOUT_A on the A113 SoC: + +tdmout_a: audio-controller@500 { + compatible = "amlogic,axg-tdmout"; + reg = <0x0 0x500 0x0 0x40>; + clocks = <&clkc_audio AUD_CLKID_TDMOUT_A>, + <&clkc_audio AUD_CLKID_TDMOUT_A_SCLK>, + <&clkc_audio AUD_CLKID_TDMOUT_A_SCLK_SEL>, + <&clkc_audio AUD_CLKID_TDMOUT_A_LRCLK>, + <&clkc_audio AUD_CLKID_TDMOUT_A_LRCLK>; + clock-names = "pclk", "sclk", "sclk_sel", + "lrclk", "lrclk_sel"; +}; diff --git a/bindings/sound/amlogic,axg-tdm-iface.txt b/bindings/sound/amlogic,axg-tdm-iface.txt new file mode 100644 index 00000000..cabfb26a --- /dev/null +++ b/bindings/sound/amlogic,axg-tdm-iface.txt @@ -0,0 +1,22 @@ +* Amlogic Audio TDM Interfaces + +Required properties: +- compatible: 'amlogic,axg-tdm-iface' +- clocks: list of clock phandle, one for each entry clock-names. +- clock-names: should contain the following: + * "sclk" : bit clock. + * "lrclk": sample clock + * "mclk" : master clock + -> optional if the interface is in clock slave mode. +- #sound-dai-cells: must be 0. + +Example of TDM_A on the A113 SoC: + +tdmif_a: audio-controller@0 { + compatible = "amlogic,axg-tdm-iface"; + #sound-dai-cells = <0>; + clocks = <&clkc_audio AUD_CLKID_MST_A_MCLK>, + <&clkc_audio AUD_CLKID_MST_A_SCLK>, + <&clkc_audio AUD_CLKID_MST_A_LRCLK>; + clock-names = "mclk", "sclk", "lrclk"; +}; diff --git a/bindings/sound/armada-370db-audio.txt b/bindings/sound/armada-370db-audio.txt new file mode 100644 index 00000000..953c092d --- /dev/null +++ b/bindings/sound/armada-370db-audio.txt @@ -0,0 +1,26 @@ +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>; + }; diff --git a/bindings/sound/arndale.txt b/bindings/sound/arndale.txt new file mode 100644 index 00000000..0e769463 --- /dev/null +++ b/bindings/sound/arndale.txt @@ -0,0 +1,24 @@ +Audio Binding for Arndale boards + +Required properties: +- compatible : Can be the following, + "samsung,arndale-rt5631" + +- samsung,audio-cpu: The phandle of the Samsung I2S controller +- samsung,audio-codec: The phandle of the audio codec + +Optional: +- samsung,model: The name of the sound-card + +Arndale Boards has many audio daughter cards, one of them is +rt5631/alc5631. Below example shows audio bindings for rt5631/ +alc5631 based codec. + +Example: + +sound { + compatible = "samsung,arndale-rt5631"; + + samsung,audio-cpu = <&i2s0> + samsung,audio-codec = <&rt5631>; +}; diff --git a/bindings/sound/atmel-at91sam9g20ek-wm8731-audio.txt b/bindings/sound/atmel-at91sam9g20ek-wm8731-audio.txt new file mode 100644 index 00000000..9c5a9947 --- /dev/null +++ b/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/bindings/sound/atmel-classd.txt b/bindings/sound/atmel-classd.txt new file mode 100644 index 00000000..89855107 --- /dev/null +++ b/bindings/sound/atmel-classd.txt @@ -0,0 +1,55 @@ +* Atmel ClassD driver under ALSA SoC architecture + +Required properties: +- compatible + Should be "atmel,sama5d2-classd". +- reg + Should contain ClassD registers location and length. +- interrupts + Should contain the IRQ line for the ClassD. +- dmas + One DMA specifiers as described in atmel-dma.txt and dma.txt files. +- dma-names + Must be "tx". +- clock-names + Tuple listing input clock names. + Required elements: "pclk" and "gclk". +- clocks + Please refer to clock-bindings.txt. +- assigned-clocks + Should be <&classd_gclk>. + +Optional properties: +- pinctrl-names, pinctrl-0 + Please refer to pinctrl-bindings.txt. +- atmel,model + The user-visible name of this sound complex. + The default value is "CLASSD". +- atmel,pwm-type + PWM modulation type, "single" or "diff". + The default value is "single". +- atmel,non-overlap-time + Set non-overlapping time, the unit is nanosecond(ns). + There are four values, + <5>, <10>, <15>, <20>, the default value is <10>. + Non-overlapping will be disabled if not specified. + +Example: +classd: classd@fc048000 { + compatible = "atmel,sama5d2-classd"; + reg = <0xfc048000 0x100>; + interrupts = <59 IRQ_TYPE_LEVEL_HIGH 7>; + dmas = <&dma0 + (AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1) + | AT91_XDMAC_DT_PERID(47))>; + dma-names = "tx"; + clocks = <&classd_clk>, <&classd_gclk>; + clock-names = "pclk", "gclk"; + assigned-clocks = <&classd_gclk>; + + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_classd_default>; + atmel,model = "classd @ SAMA5D2-Xplained"; + atmel,pwm-type = "diff"; + atmel,non-overlap-time = <10>; +}; diff --git a/bindings/sound/atmel-i2s.txt b/bindings/sound/atmel-i2s.txt new file mode 100644 index 00000000..40549f49 --- /dev/null +++ b/bindings/sound/atmel-i2s.txt @@ -0,0 +1,46 @@ +* Atmel I2S controller + +Required properties: +- compatible: Should be "atmel,sama5d2-i2s". +- reg: Should be the physical base address of the controller and the + length of memory mapped region. +- interrupts: Should contain the interrupt for the controller. +- dmas: Should be one per channel name listed in the dma-names property, + as described in atmel-dma.txt and dma.txt files. +- dma-names: Two dmas have to be defined, "tx" and "rx". + This IP also supports one shared channel for both rx and tx; + if this mode is used, one "rx-tx" name must be used. +- clocks: Must contain an entry for each entry in clock-names. + Please refer to clock-bindings.txt. +- clock-names: Should be one of each entry matching the clocks phandles list: + - "pclk" (peripheral clock) Required. + - "gclk" (generated clock) Optional (1). + - "muxclk" (I2S mux clock) Optional (1). + +Optional properties: +- pinctrl-0: Should specify pin control groups used for this controller. +- princtrl-names: Should contain only one value - "default". + + +(1) : Only the peripheral clock is required. The generated clock and the I2S + mux clock are optional and should only be set together, when Master Mode + is required. + +Example: + + i2s@f8050000 { + compatible = "atmel,sama5d2-i2s"; + reg = <0xf8050000 0x300>; + interrupts = <54 IRQ_TYPE_LEVEL_HIGH 7>; + dmas = <&dma0 + (AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1) | + AT91_XDMAC_DT_PERID(31))>, + <&dma0 + (AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1) | + AT91_XDMAC_DT_PERID(32))>; + dma-names = "tx", "rx"; + clocks = <&i2s0_clk>, <&i2s0_gclk>, <&i2s0muxck>; + clock-names = "pclk", "gclk", "muxclk"; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_i2s0_default>; + }; diff --git a/bindings/sound/atmel-pdmic.txt b/bindings/sound/atmel-pdmic.txt new file mode 100644 index 00000000..e0875f17 --- /dev/null +++ b/bindings/sound/atmel-pdmic.txt @@ -0,0 +1,55 @@ +* Atmel PDMIC driver under ALSA SoC architecture + +Required properties: +- compatible + Should be "atmel,sama5d2-pdmic". +- reg + Should contain PDMIC registers location and length. +- interrupts + Should contain the IRQ line for the PDMIC. +- dmas + One DMA specifiers as described in atmel-dma.txt and dma.txt files. +- dma-names + Must be "rx". +- clock-names + Required elements: + - "pclk" peripheral clock + - "gclk" generated clock +- clocks + Must contain an entry for each required entry in clock-names. + Please refer to clock-bindings.txt. +- atmel,mic-min-freq + The minimal frequency that the micphone supports. +- atmel,mic-max-freq + The maximal frequency that the micphone supports. + +Optional properties: +- pinctrl-names, pinctrl-0 + Please refer to pinctrl-bindings.txt. +- atmel,model + The user-visible name of this sound card. + The default value is "PDMIC". +- atmel,mic-offset + The offset that should be added. + The range is from -32768 to 32767. + The default value is 0. + +Example: + pdmic@f8018000 { + compatible = "atmel,sama5d2-pdmic"; + reg = <0xf8018000 0x124>; + interrupts = <48 IRQ_TYPE_LEVEL_HIGH 7>; + dmas = <&dma0 + (AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1) + | AT91_XDMAC_DT_PERID(50))>; + dma-names = "rx"; + clocks = <&pdmic_clk>, <&pdmic_gclk>; + clock-names = "pclk", "gclk"; + + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_pdmic_default>; + atmel,model = "PDMIC @ sama5d2_xplained"; + atmel,mic-min-freq = <1000000>; + atmel,mic-max-freq = <3246000>; + atmel,mic-offset = <0x0>; + }; diff --git a/bindings/sound/atmel-sam9x5-wm8731-audio.txt b/bindings/sound/atmel-sam9x5-wm8731-audio.txt new file mode 100644 index 00000000..07208570 --- /dev/null +++ b/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/bindings/sound/atmel-wm8904.txt b/bindings/sound/atmel-wm8904.txt new file mode 100644 index 00000000..8bbe50c8 --- /dev/null +++ b/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/bindings/sound/atmel_ac97c.txt b/bindings/sound/atmel_ac97c.txt new file mode 100644 index 00000000..b151bd90 --- /dev/null +++ b/bindings/sound/atmel_ac97c.txt @@ -0,0 +1,20 @@ +* Atmel AC97 controller + +Required properties: + - compatible: "atmel,at91sam9263-ac97c" + - reg: Address and length of the register set for the device + - interrupts: Should contain AC97 interrupt + - ac97-gpios: Please refer to soc-ac97link.txt, only ac97-reset is used +Optional properties: + - pinctrl-names, pinctrl-0: Please refer to pinctrl-bindings.txt + +Example: +sound@fffa0000 { + compatible = "atmel,at91sam9263-ac97c"; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_ac97>; + reg = <0xfffa0000 0x4000>; + interrupts = <18 IRQ_TYPE_LEVEL_HIGH 5>; + + ac97-gpios = <&pioB 0 0 &pioB 2 0 &pioC 29 GPIO_ACTIVE_LOW>; +}; diff --git a/bindings/sound/audio-graph-card.txt b/bindings/sound/audio-graph-card.txt new file mode 100644 index 00000000..7e63e53a --- /dev/null +++ b/bindings/sound/audio-graph-card.txt @@ -0,0 +1,132 @@ +Audio Graph Card: + +Audio Graph Card specifies audio DAI connections of SoC <-> codec. +It is based on common bindings for device graphs. +see ${LINUX}/Documentation/devicetree/bindings/graph.txt + +Basically, Audio Graph Card property is same as Simple Card. +see ${LINUX}/Documentation/devicetree/bindings/sound/simple-card.txt + +Below are same as Simple-Card. + +- label +- widgets +- routing +- dai-format +- frame-master +- bitclock-master +- bitclock-inversion +- frame-inversion +- mclk-fs +- hp-det-gpio +- mic-det-gpio +- dai-tdm-slot-num +- dai-tdm-slot-width +- clocks / system-clock-frequency + +Required properties: + +- compatible : "audio-graph-card"; +- dais : list of CPU DAI port{s} + +Optional properties: +- pa-gpios: GPIO used to control external amplifier. + +Example: Single DAI case + + sound_card { + compatible = "audio-graph-card"; + + dais = <&cpu_port>; + }; + + dai-controller { + ... + cpu_port: port { + cpu_endpoint: endpoint { + remote-endpoint = <&codec_endpoint>; + + dai-format = "left_j"; + ... + }; + }; + }; + + audio-codec { + ... + port { + codec_endpoint: endpoint { + remote-endpoint = <&cpu_endpoint>; + }; + }; + }; + +Example: Multi DAI case + + sound-card { + compatible = "audio-graph-card"; + + label = "sound-card"; + + dais = <&cpu_port0 + &cpu_port1 + &cpu_port2>; + }; + + audio-codec@0 { + ... + port { + codec0_endpoint: endpoint { + remote-endpoint = <&cpu_endpoint0>; + }; + }; + }; + + audio-codec@1 { + ... + port { + codec1_endpoint: endpoint { + remote-endpoint = <&cpu_endpoint1>; + }; + }; + }; + + audio-codec@2 { + ... + port { + codec2_endpoint: endpoint { + remote-endpoint = <&cpu_endpoint2>; + }; + }; + }; + + dai-controller { + ... + ports { + cpu_port0: port@0 { + cpu_endpoint0: endpoint { + remote-endpoint = <&codec0_endpoint>; + + dai-format = "left_j"; + ... + }; + }; + cpu_port1: port@1 { + cpu_endpoint1: endpoint { + remote-endpoint = <&codec1_endpoint>; + + dai-format = "i2s"; + ... + }; + }; + cpu_port2: port@2 { + cpu_endpoint2: endpoint { + remote-endpoint = <&codec2_endpoint>; + + dai-format = "i2s"; + ... + }; + }; + }; + }; + diff --git a/bindings/sound/audio-graph-scu-card.txt b/bindings/sound/audio-graph-scu-card.txt new file mode 100644 index 00000000..441dd6f2 --- /dev/null +++ b/bindings/sound/audio-graph-scu-card.txt @@ -0,0 +1,123 @@ +Audio-Graph-SCU-Card: + +Audio-Graph-SCU-Card is "Audio-Graph-Card" + "ALSA DPCM". + +It is based on common bindings for device graphs. +see ${LINUX}/Documentation/devicetree/bindings/graph.txt + +Basically, Audio-Graph-SCU-Card property is same as +Simple-Card / Simple-SCU-Card / Audio-Graph-Card. +see ${LINUX}/Documentation/devicetree/bindings/sound/simple-card.txt + ${LINUX}/Documentation/devicetree/bindings/sound/simple-scu-card.txt + ${LINUX}/Documentation/devicetree/bindings/sound/audio-graph-card.txt + +Below are same as Simple-Card / Audio-Graph-Card. + +- label +- dai-format +- frame-master +- bitclock-master +- bitclock-inversion +- frame-inversion +- dai-tdm-slot-num +- dai-tdm-slot-width +- clocks / system-clock-frequency + +Below are same as Simple-SCU-Card. + +- convert-rate +- convert-channels +- prefix +- routing + +Required properties: + +- compatible : "audio-graph-scu-card"; +- dais : list of CPU DAI port{s} + +Example 1. Sampling Rate Conversion + + sound_card { + compatible = "audio-graph-scu-card"; + + label = "sound-card"; + prefix = "codec"; + routing = "codec Playback", "DAI0 Playback", + "DAI0 Capture", "codec Capture"; + convert-rate = <48000>; + + dais = <&cpu_port>; + }; + + audio-codec { + ... + + port { + codec_endpoint: endpoint { + remote-endpoint = <&cpu_endpoint>; + }; + }; + }; + + dai-controller { + ... + cpu_port: port { + cpu_endpoint: endpoint { + remote-endpoint = <&codec_endpoint>; + + dai-format = "left_j"; + ... + }; + }; + }; + +Example 2. 2 CPU 1 Codec (Mixing) + + sound_card { + compatible = "audio-graph-scu-card"; + + label = "sound-card"; + prefix = "codec"; + routing = "codec Playback", "DAI0 Playback", + "codec Playback", "DAI1 Playback", + "DAI0 Capture", "codec Capture"; + convert-rate = <48000>; + + dais = <&cpu_port0 + &cpu_port1>; + }; + + audio-codec { + ... + + port { + codec_endpoint0: endpoint { + remote-endpoint = <&cpu_endpoint0>; + }; + codec_endpoint1: endpoint { + remote-endpoint = <&cpu_endpoint1>; + }; + }; + }; + + dai-controller { + ... + ports { + cpu_port0: port { + cpu_endpoint0: endpoint { + remote-endpoint = <&codec_endpoint0>; + + dai-format = "left_j"; + ... + }; + }; + cpu_port1: port { + cpu_endpoint1: endpoint { + remote-endpoint = <&codec_endpoint1>; + + dai-format = "left_j"; + ... + }; + }; + }; + }; diff --git a/bindings/sound/axentia,tse850-pcm5142.txt b/bindings/sound/axentia,tse850-pcm5142.txt new file mode 100644 index 00000000..9d049d4b --- /dev/null +++ b/bindings/sound/axentia,tse850-pcm5142.txt @@ -0,0 +1,92 @@ +Devicetree bindings for the Axentia TSE-850 audio complex + +Required properties: + - compatible: "axentia,tse850-pcm5142" + - axentia,cpu-dai: The phandle of the cpu dai. + - axentia,audio-codec: The phandle of the PCM5142 codec. + - axentia,add-gpios: gpio specifier that controls the mixer. + - axentia,loop1-gpios: gpio specifier that controls loop relays on channel 1. + - axentia,loop2-gpios: gpio specifier that controls loop relays on channel 2. + - axentia,ana-supply: Regulator that supplies the output amplifier. Must + support voltages in the 2V - 20V range, in 1V steps. + +The schematics explaining the gpios are as follows: + + loop1 relays + IN1 +---o +------------+ o---+ OUT1 + \ / + + + + | / | + +--o +--. | + | add | | + | V | + | .---. | + DAC +----------->|Sum|---+ + | '---' | + | | + + + + + IN2 +---o--+------------+--o---+ OUT2 + loop2 relays + +The 'loop1' gpio pin controlls two relays, which are either in loop position, +meaning that input and output are directly connected, or they are in mixer +position, meaning that the signal is passed through the 'Sum' mixer. Similarly +for 'loop2'. + +In the above, the 'loop1' relays are inactive, thus feeding IN1 to the mixer +(if 'add' is active) and feeding the mixer output to OUT1. The 'loop2' relays +are active, short-cutting the TSE-850 from channel 2. IN1, IN2, OUT1 and OUT2 +are TSE-850 connectors and DAC is the PCB name of the (filtered) output from +the PCM5142 codec. + +Example: + + &ssc0 { + #sound-dai-cells = <0>; + + }; + + &i2c { + codec: pcm5142@4c { + compatible = "ti,pcm5142"; + + reg = <0x4c>; + + AVDD-supply = <®_3v3>; + DVDD-supply = <®_3v3>; + CPVDD-supply = <®_3v3>; + + clocks = <&sck>; + + pll-in = <3>; + pll-out = <6>; + }; + }; + + ana: ana-reg { + compatible = "pwm-regulator"; + + regulator-name = "ANA"; + + pwms = <&pwm0 2 1000 PWM_POLARITY_INVERTED>; + pwm-dutycycle-unit = <1000>; + pwm-dutycycle-range = <100 1000>; + + regulator-min-microvolt = <2000000>; + regulator-max-microvolt = <20000000>; + regulator-ramp-delay = <1000>; + }; + + sound { + compatible = "axentia,tse850-pcm5142"; + + axentia,cpu-dai = <&ssc0>; + axentia,audio-codec = <&codec>; + + axentia,add-gpios = <&pioA 8 GPIO_ACTIVE_LOW>; + axentia,loop1-gpios = <&pioA 10 GPIO_ACTIVE_LOW>; + axentia,loop2-gpios = <&pioA 11 GPIO_ACTIVE_LOW>; + + axentia,ana-supply = <&ana>; + }; diff --git a/bindings/sound/brcm,bcm2835-i2s.txt b/bindings/sound/brcm,bcm2835-i2s.txt new file mode 100644 index 00000000..7bb03628 --- /dev/null +++ b/bindings/sound/brcm,bcm2835-i2s.txt @@ -0,0 +1,24 @@ +* Broadcom BCM2835 SoC I2S/PCM module + +Required properties: +- compatible: "brcm,bcm2835-i2s" +- reg: Should contain PCM registers location and length. +- clocks: the (PCM) clock to use +- 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 0x24>; + clocks = <&clocks BCM2835_CLOCK_PCM>; + + dmas = <&dma 2>, + <&dma 3>; + dma-names = "tx", "rx"; +}; diff --git a/bindings/sound/brcm,cygnus-audio.txt b/bindings/sound/brcm,cygnus-audio.txt new file mode 100644 index 00000000..630bf7c0 --- /dev/null +++ b/bindings/sound/brcm,cygnus-audio.txt @@ -0,0 +1,63 @@ +BROADCOM Cygnus Audio I2S/TDM/SPDIF controller + +Required properties: + - compatible : "brcm,cygnus-audio" + - #address-cells: 32bit valued, 1 cell. + - #size-cells: 32bit valued, 0 cell. + - reg : Should contain audio registers location and length + - reg-names: names of the registers listed in "reg" property + Valid names are "aud" and "i2s_in". "aud" contains a + set of DMA, I2S_OUT and SPDIF registers. "i2s_in" contains + a set of I2S_IN registers. + - clocks: PLL and leaf clocks used by audio ports + - assigned-clocks: PLL and leaf clocks + - assigned-clock-parents: parent clocks of the assigned clocks + (usually the PLL) + - assigned-clock-rates: List of clock frequencies of the + assigned clocks + - clock-names: names of 3 leaf clocks used by audio ports + Valid names are "ch0_audio", "ch1_audio", "ch2_audio" + - interrupts: audio DMA interrupt number + +SSP Subnode properties: +- reg: The index of ssp port interface to use + Valid value are 0, 1, 2, or 3 (for spdif) + +Example: + cygnus_audio: audio@180ae000 { + compatible = "brcm,cygnus-audio"; + #address-cells = <1>; + #size-cells = <0>; + reg = <0x180ae000 0xafd>, <0x180aec00 0x1f8>; + reg-names = "aud", "i2s_in"; + clocks = <&audiopll BCM_CYGNUS_AUDIOPLL_CH0>, + <&audiopll BCM_CYGNUS_AUDIOPLL_CH1>, + <&audiopll BCM_CYGNUS_AUDIOPLL_CH2>; + assigned-clocks = <&audiopll BCM_CYGNUS_AUDIOPLL>, + <&audiopll BCM_CYGNUS_AUDIOPLL_CH0>, + <&audiopll BCM_CYGNUS_AUDIOPLL_CH1>, + <&audiopll BCM_CYGNUS_AUDIOPLL_CH2>; + assigned-clock-parents = <&audiopll BCM_CYGNUS_AUDIOPLL>; + assigned-clock-rates = <1769470191>, + <0>, + <0>, + <0>; + clock-names = "ch0_audio", "ch1_audio", "ch2_audio"; + interrupts = <GIC_SPI 143 IRQ_TYPE_LEVEL_HIGH>; + + ssp0: ssp_port@0 { + reg = <0>; + }; + + ssp1: ssp_port@1 { + reg = <1>; + }; + + ssp2: ssp_port@2 { + reg = <2>; + }; + + spdif: spdif_port@3 { + reg = <3>; + }; + }; diff --git a/bindings/sound/bt-sco.txt b/bindings/sound/bt-sco.txt new file mode 100644 index 00000000..641edf75 --- /dev/null +++ b/bindings/sound/bt-sco.txt @@ -0,0 +1,13 @@ +Bluetooth-SCO audio CODEC + +This device support generic Bluetooth SCO link. + +Required properties: + + - compatible : "delta,dfbmcs320" or "linux,bt-sco" + +Example: + +codec: bt_sco { + compatible = "delta,dfbmcs320"; +}; diff --git a/bindings/sound/cdns,xtfpga-i2s.txt b/bindings/sound/cdns,xtfpga-i2s.txt new file mode 100644 index 00000000..860fc0da --- /dev/null +++ b/bindings/sound/cdns,xtfpga-i2s.txt @@ -0,0 +1,18 @@ +Bindings for I2S controller built into xtfpga Xtensa bitstreams. + +Required properties: +- compatible: shall be "cdns,xtfpga-i2s". +- reg: memory region (address and length) with device registers. +- interrupts: interrupt for the device. +- clocks: phandle to the clk used as master clock. I2S bus clock + is derived from it. + +Examples: + + i2s0: xtfpga-i2s@d080000 { + #sound-dai-cells = <0>; + compatible = "cdns,xtfpga-i2s"; + reg = <0x0d080000 0x40>; + interrupts = <2 1>; + clocks = <&cdce706 4>; + }; diff --git a/bindings/sound/cs35l32.txt b/bindings/sound/cs35l32.txt new file mode 100644 index 00000000..1417d3f5 --- /dev/null +++ b/bindings/sound/cs35l32.txt @@ -0,0 +1,62 @@ +CS35L32 audio CODEC + +Required properties: + + - compatible : "cirrus,cs35l32" + + - reg : the I2C address of the device for I2C. Address is determined by the level + of the AD0 pin. Level 0 is 0x40 while Level 1 is 0x41. + + - VA-supply, VP-supply : power supplies for the device, + as covered in Documentation/devicetree/bindings/regulator/regulator.txt. + +Optional properties: + + - reset-gpios : a GPIO spec for the reset pin. If specified, it will be + deasserted before communication to the codec starts. + + - cirrus,boost-manager : Boost voltage control. + 0 = Automatically managed. Boost-converter output voltage is the higher + of the two: Class G or adaptive LED voltage. + 1 = Automatically managed irrespective of audio, adapting for low-power + dissipation when LEDs are ON, and operating in Fixed-Boost Bypass Mode + if LEDs are OFF (VBST = VP). + 2 = (Default) Boost voltage fixed in Bypass Mode (VBST = VP). + 3 = Boost voltage fixed at 5 V. + + - cirrus,sdout-datacfg : Data configuration for dual CS35L32 applications only. + Determines the data packed in a two-CS35L32 configuration. + 0 = Left/right channels VMON[11:0], IMON[11:0], VPMON[7:0]. + 1 = Left/right channels VMON[11:0], IMON[11:0], STATUS. + 2 = (Default) left/right channels VMON[15:0], IMON [15:0]. + 3 = Left/right channels VPMON[7:0], STATUS. + + - cirrus,sdout-share : SDOUT sharing. Determines whether one or two CS35L32 + devices are on board sharing SDOUT. + 0 = (Default) One IC. + 1 = Two IC's. + + - cirrus,battery-recovery : Low battery nominal recovery threshold, rising VP. + 0 = 3.1V + 1 = 3.2V + 2 = 3.3V (Default) + 3 = 3.4V + + - cirrus,battery-threshold : Low battery nominal threshold, falling VP. + 0 = 3.1V + 1 = 3.2V + 2 = 3.3V + 3 = 3.4V (Default) + 4 = 3.5V + 5 = 3.6V + +Example: + +codec: codec@40 { + compatible = "cirrus,cs35l32"; + reg = <0x40>; + reset-gpios = <&gpio 10 0>; + cirrus,boost-manager = <0x03>; + cirrus,sdout-datacfg = <0x02>; + VA-supply = <®_audio>; +}; diff --git a/bindings/sound/cs35l33.txt b/bindings/sound/cs35l33.txt new file mode 100644 index 00000000..dc5a355d --- /dev/null +++ b/bindings/sound/cs35l33.txt @@ -0,0 +1,124 @@ +CS35L33 Speaker Amplifier + +Required properties: + + - compatible : "cirrus,cs35l33" + + - reg : the I2C address of the device for I2C + + - VA-supply, VP-supply : power supplies for the device, + as covered in + Documentation/devicetree/bindings/regulator/regulator.txt. + +Optional properties: + + - reset-gpios : gpio used to reset the amplifier + + - interrupts : IRQ line info CS35L33. + (See Documentation/devicetree/bindings/interrupt-controller/interrupts.txt + for further information relating to interrupt properties) + + - cirrus,boost-ctl : Booster voltage use to supply the amp. If the value is + 0, then VBST = VP. If greater than 0, the boost voltage will be 3300mV with + a value of 1 and will increase at a step size of 100mV until a maximum of + 8000mV. + + - cirrus,ramp-rate : On power up, it affects the time from when the power + up sequence begins to the time the audio reaches a full-scale output. + On power down, it affects the time from when the power-down sequence + begins to when the amplifier disables the PWM outputs. If this property + is not set then soft ramping will be disabled and ramp time would be + 20ms. If this property is set to 0,1,2,3 then ramp times would be 40ms, + 60ms,100ms,175ms respectively for 48KHz sample rate. + + - cirrus,boost-ipk : The maximum current allowed for the boost converter. + The range starts at 1850000uA and goes to a maximum of 3600000uA + with a step size of 15625uA. The default is 2500000uA. + + - cirrus,imon-adc-scale : Configures the scaling of data bits from the IMON + ADC data word. This property can be set as a value of 0 for bits 15 down + to 0, 6 for 21 down to 6, 7, for 22 down to 7, 8 for 23 down to 8. + + +Optional H/G Algorithm sub-node: + +The cs35l33 node can have a single "cirrus,hg-algo" sub-node that will enable +the internal H/G Algorithm. + + - cirrus,hg-algo : Sub-node for internal Class H/G algorithm that + controls the amplifier supplies. + +Optional properties for the "cirrus,hg-algo" sub-node: + + - cirrus,mem-depth : Memory depth for the Class H/G algorithm measured in + LRCLK cycles. If this property is set to 0, 1, 2, or 3 then the memory + depths will be 1, 4, 8, 16 LRCLK cycles. The default is 16 LRCLK cycles. + + cirrus,release-rate : The number of consecutive LRCLK periods before + allowing release condition tracking updates. The number of LRCLK periods + start at 3 to a maximum of 255. + + - cirrus,ldo-thld : Configures the signal threshold at which the PWM output + stage enters LDO operation. Starts as a default value of 50mV for a value + of 1 and increases with a step size of 50mV to a maximum of 750mV (value of + 0xF). + + - cirrus,ldo-path-disable : This is a boolean property. If present, the H/G + algorithm uses the max detection path. If not present, the LDO + detection path is used. + + - cirrus,ldo-entry-delay : The LDO entry delay in milliseconds before the H/G + algorithm switches to the LDO voltage. This property can be set to values + from 0 to 7 for delays of 5ms, 10ms, 50ms, 100ms, 200ms, 500ms, 1000ms. + The default is 100ms. + + - cirrus,vp-hg-auto : This is a boolean property. When set, class H/G VPhg + automatic updating is enabled. + + - cirrus,vp-hg : Class H/G algorithm VPhg. Controls the H/G algorithm's + reference to the VP voltage for when to start generating a boosted VBST. + The reference voltage starts at 3000mV with a value of 0x3 and is increased + by 100mV per step to a maximum of 5500mV. + + - cirrus,vp-hg-rate : The rate (number of LRCLK periods) at which the VPhg is + allowed to increase to a higher voltage when using VPhg automatic + tracking. This property can be set to values from 0 to 3 with rates of 128 + periods, 2048 periods, 32768 periods, and 524288 periods. + The default is 32768 periods. + + - cirrus,vp-hg-va : VA calculation reference for automatic VPhg tracking + using VPMON. This property can be set to values from 0 to 6 starting at + 1800mV with a step size of 50mV up to a maximum value of 1750mV. + Default is 1800mV. + +Example: + +cs35l33: cs35l33@40 { + compatible = "cirrus,cs35l33"; + reg = <0x40>; + + VA-supply = <&ldo5_reg>; + VP-supply = <&ldo5_reg>; + + interrupt-parent = <&gpio8>; + interrupts = <3 IRQ_TYPE_LEVEL_LOW>; + + reset-gpios = <&cs47l91 34 0>; + + cirrus,ramp-rate = <0x0>; + cirrus,boost-ctl = <0x30>; /* VBST = 8000mV */ + cirrus,boost-ipk = <0xE0>; /* 3600mA */ + cirrus,imon-adc-scale = <0> /* Bits 15 down to 0 */ + + cirrus,hg-algo { + cirrus,mem-depth = <0x3>; + cirrus,release-rate = <0x3>; + cirrus,ldo-thld = <0x1>; + cirrus,ldo-path-disable = <0x0>; + cirrus,ldo-entry-delay=<0x4>; + cirrus,vp-hg-auto; + cirrus,vp-hg=<0xF>; + cirrus,vp-hg-rate=<0x2>; + cirrus,vp-hg-va=<0x0>; + }; +}; diff --git a/bindings/sound/cs35l34.txt b/bindings/sound/cs35l34.txt new file mode 100644 index 00000000..2f7606b7 --- /dev/null +++ b/bindings/sound/cs35l34.txt @@ -0,0 +1,62 @@ +CS35L34 Speaker Amplifier + +Required properties: + + - compatible : "cirrus,cs35l34" + + - reg : the I2C address of the device for I2C. + + - VA-supply, VP-supply : power supplies for the device, + as covered in + Documentation/devicetree/bindings/regulator/regulator.txt. + + - cirrus,boost-vtge-millivolt : Boost Voltage Value. Configures the boost + converter's output voltage in mV. The range is from VP to 8V with + increments of 100mV. + + - cirrus,boost-nanohenry: Inductor value for boost converter. The value is + in nH and they can be values of 1000nH, 1100nH, 1200nH, 1500nH, and 2200nH. + +Optional properties: + + - reset-gpios: GPIO used to reset the amplifier. + + - interrupts : IRQ line info CS35L34. + (See Documentation/devicetree/bindings/interrupt-controller/interrupts.txt + for further information relating to interrupt properties) + + - cirrus,boost-peak-milliamp : Boost converter peak current limit in mA. The + range starts at 1200mA and goes to a maximum of 3840mA with increments of + 80mA. The default value is 2480mA. + + - cirrus,i2s-sdinloc : ADSP SDIN I2S channel location. Indicates whether the + received mono data is in the left or right portion of the I2S frame + according to the AD0 pin or directly via this configuration. + 0x0 (Default) = Selected by AD0 input (if AD0 = LOW, use left channel), + 0x2 = Left, + 0x1 = Selected by the inversion of the AD0 input (if AD0 = LOW, use right + channel), + 0x3 = Right. + + - cirrus,gain-zc-disable: Boolean property. If set, the gain change will take + effect without waiting for a zero cross. + + - cirrus,tdm-rising-edge: Boolean property. If set, data is on the rising edge of + SCLK. Otherwise, data is on the falling edge of SCLK. + + +Example: + +cs35l34: cs35l34@40 { + compatible = "cirrus,cs35l34"; + reg = <0x40>; + + interrupt-parent = <&gpio8>; + interrupts = <3 IRQ_TYPE_LEVEL_LOW>; + + reset-gpios = <&gpio 10 0>; + + cirrus,boost-vtge-milltvolt = <8000>; /* 8V */ + cirrus,boost-ind-nanohenry = <1000>; /* 1uH */ + cirrus,boost-peak-milliamp = <3000>; /* 3A */ +}; diff --git a/bindings/sound/cs35l35.txt b/bindings/sound/cs35l35.txt new file mode 100644 index 00000000..7915897f --- /dev/null +++ b/bindings/sound/cs35l35.txt @@ -0,0 +1,181 @@ +CS35L35 Boosted Speaker Amplifier + +Required properties: + + - compatible : "cirrus,cs35l35" + + - reg : the I2C address of the device for I2C + + - VA-supply, VP-supply : power supplies for the device, + as covered in + Documentation/devicetree/bindings/regulator/regulator.txt. + + - interrupts : IRQ line info CS35L35. + (See Documentation/devicetree/bindings/interrupt-controller/interrupts.txt + for further information relating to interrupt properties) + + - cirrus,boost-ind-nanohenry: Inductor value for boost converter. The value is + in nH and they can be values of 1000nH, 1200nH, 1500nH, and 2200nH. + +Optional properties: + - reset-gpios : gpio used to reset the amplifier + + - cirrus,stereo-config : Boolean to determine if there are 2 AMPs for a + Stereo configuration + + - cirrus,audio-channel : Set Location of Audio Signal on Serial Port + 0 = Data Packet received on Left I2S Channel + 1 = Data Packet received on Right I2S Channel + + - cirrus,advisory-channel : Set Location of Advisory Signal on Serial Port + 0 = Data Packet received on Left I2S Channel + 1 = Data Packet received on Right I2S Channel + + - cirrus,shared-boost : Boolean to enable ClassH tracking of Advisory Signal + if 2 Devices share Boost BST_CTL + + - cirrus,external-boost : Boolean to specify the device is using an external + boost supply, note that sharing a boost from another cs35l35 would constitute + using an external supply for the slave device + + - cirrus,sp-drv-strength : Value for setting the Serial Port drive strength + Table 3-10 of the datasheet lists drive-strength specifications + 0 = 1x (Default) + 1 = .5x + - cirrus,sp-drv-unused : Determines how unused slots should be driven on the + Serial Port. + 0 - Hi-Z + 2 - Drive 0's (Default) + 3 - Drive 1's + + - cirrus,bst-pdn-fet-on : Boolean to determine if the Boost PDN control + powers down with a rectification FET On or Off. If VSPK is supplied + externally then FET is off. + + - cirrus,boost-ctl-millivolt : Boost Voltage Value. Configures the boost + converter's output voltage in mV. The range is from 2600mV to 9000mV with + increments of 100mV. + (Default) VP + + - cirrus,boost-peak-milliamp : Boost-converter peak current limit in mA. + Configures the peak current by monitoring the current through the boost FET. + Range starts at 1680mA and goes to a maximum of 4480mA with increments of + 110mA. + (Default) 2.46 Amps + + - cirrus,amp-gain-zc : Boolean to determine if to use Amplifier gain-change + zero-cross + +Optional H/G Algorithm sub-node: + + The cs35l35 node can have a single "cirrus,classh-internal-algo" sub-node + that will disable automatic control of the internal H/G Algorithm. + + It is strongly recommended that the Datasheet be referenced when adjusting + or using these Class H Algorithm controls over the internal Algorithm. + Serious damage can occur to the Device and surrounding components. + + - cirrus,classh-internal-algo : Sub-node for the Internal Class H Algorithm + See Section 4.3 Internal Class H Algorithm in the Datasheet. + If not used, the device manages the ClassH Algorithm internally. + +Optional properties for the "cirrus,classh-internal-algo" Sub-node + + Section 7.29 Class H Control + - cirrus,classh-bst-overide : Boolean + - cirrus,classh-bst-max-limit + - cirrus,classh-mem-depth + + Section 7.30 Class H Headroom Control + - cirrus,classh-headroom + + Section 7.31 Class H Release Rate + - cirrus,classh-release-rate + + Section 7.32 Class H Weak FET Drive Control + - cirrus,classh-wk-fet-disable + - cirrus,classh-wk-fet-delay + - cirrus,classh-wk-fet-thld + + Section 7.34 Class H VP Control + - cirrus,classh-vpch-auto + - cirrus,classh-vpch-rate + - cirrus,classh-vpch-man + +Optional Monitor Signal Format sub-node: + + The cs35l35 node can have a single "cirrus,monitor-signal-format" sub-node + for adjusting the Depth, Location and Frame of the Monitoring Signals + for Algorithms. + + See Sections 4.8.2 through 4.8.4 Serial-Port Control in the Datasheet + + -cirrus,monitor-signal-format : Sub-node for the Monitor Signaling Formating + on the I2S Port. Each of the 3 8 bit values in the array contain the settings + for depth, location, and frame. + + If not used, the defaults for the 6 monitor signals is used. + + Sections 7.44 - 7.53 lists values for the depth, location, and frame + for each monitoring signal. + + - cirrus,imon : 4 8 bit values to set the depth, location, frame and ADC + scale of the IMON monitor signal. + + - cirrus,vmon : 3 8 bit values to set the depth, location, and frame + of the VMON monitor signal. + + - cirrus,vpmon : 3 8 bit values to set the depth, location, and frame + of the VPMON monitor signal. + + - cirrus,vbstmon : 3 8 bit values to set the depth, location, and frame + of the VBSTMON monitor signal + + - cirrus,vpbrstat : 3 8 bit values to set the depth, location, and frame + of the VPBRSTAT monitor signal + + - cirrus,zerofill : 3 8 bit values to set the depth, location, and frame\ + of the ZEROFILL packet in the monitor signal + +Example: + +cs35l35: cs35l35@20 { + compatible = "cirrus,cs35l35"; + reg = <0x20>; + VA-supply = <&dummy_vreg>; + VP-supply = <&dummy_vreg>; + reset-gpios = <&axi_gpio 54 0>; + interrupt-parent = <&gpio8>; + interrupts = <3 IRQ_TYPE_LEVEL_LOW>; + cirrus,boost-ctl-millivolt = <9000>; + + cirrus,stereo-config; + cirrus,audio-channel = <0x00>; + cirrus,advisory-channel = <0x01>; + cirrus,shared-boost; + + cirrus,classh-internal-algo { + cirrus,classh-bst-overide; + cirrus,classh-bst-max-limit = <0x01>; + cirrus,classh-mem-depth = <0x01>; + cirrus,classh-release-rate = <0x08>; + cirrus,classh-headroom-millivolt = <0x0B>; + cirrus,classh-wk-fet-disable = <0x01>; + cirrus,classh-wk-fet-delay = <0x04>; + cirrus,classh-wk-fet-thld = <0x01>; + cirrus,classh-vpch-auto = <0x01>; + cirrus,classh-vpch-rate = <0x02>; + cirrus,classh-vpch-man = <0x05>; + }; + + /* Depth, Location, Frame */ + cirrus,monitor-signal-format { + cirrus,imon = /bits/ 8 <0x03 0x00 0x01>; + cirrus,vmon = /bits/ 8 <0x03 0x00 0x00>; + cirrus,vpmon = /bits/ 8 <0x03 0x04 0x00>; + cirrus,vbstmon = /bits/ 8 <0x03 0x04 0x01>; + cirrus,vpbrstat = /bits/ 8 <0x00 0x04 0x00>; + cirrus,zerofill = /bits/ 8 <0x00 0x00 0x00>; + }; + +}; diff --git a/bindings/sound/cs4265.txt b/bindings/sound/cs4265.txt new file mode 100644 index 00000000..380fff8e --- /dev/null +++ b/bindings/sound/cs4265.txt @@ -0,0 +1,29 @@ +CS4265 audio CODEC + +This device supports I2C only. + +Required properties: + + - compatible : "cirrus,cs4265" + + - reg : the I2C address of the device for I2C. The I2C address depends on + the state of the AD0 pin. If AD0 is high, the i2c address is 0x4f. + If it is low, the i2c address is 0x4e. + +Optional properties: + + - reset-gpios : a GPIO spec for the reset pin. If specified, it will be + deasserted before communication to the codec starts. + +Examples: + +codec_ad0_high: cs4265@4f { /* AD0 Pin is high */ + compatible = "cirrus,cs4265"; + reg = <0x4f>; +}; + + +codec_ad0_low: cs4265@4e { /* AD0 Pin is low */ + compatible = "cirrus,cs4265"; + reg = <0x4e>; +}; diff --git a/bindings/sound/cs4270.txt b/bindings/sound/cs4270.txt new file mode 100644 index 00000000..6b222f9b --- /dev/null +++ b/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/bindings/sound/cs4271.txt b/bindings/sound/cs4271.txt new file mode 100644 index 00000000..6e699cea --- /dev/null +++ b/bindings/sound/cs4271.txt @@ -0,0 +1,57 @@ +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. + + - vd-supply: Digital power + - vl-supply: Logic power + - va-supply: Analog Power + +Examples: + + codec_i2c: cs4271@10 { + compatible = "cirrus,cs4271"; + reg = <0x10>; + reset-gpio = <&gpio 23 0>; + vd-supply = <&vdd_3v3_reg>; + vl-supply = <&vdd_3v3_reg>; + va-supply = <&vdd_3v3_reg>; + }; + + codec_spi: cs4271@0 { + compatible = "cirrus,cs4271"; + reg = <0x0>; + reset-gpio = <&gpio 23 0>; + spi-max-frequency = <6000000>; + }; + diff --git a/bindings/sound/cs42l42.txt b/bindings/sound/cs42l42.txt new file mode 100644 index 00000000..7dfaa2ab --- /dev/null +++ b/bindings/sound/cs42l42.txt @@ -0,0 +1,107 @@ +CS42L42 audio CODEC + +Required properties: + + - compatible : "cirrus,cs42l42" + + - reg : the I2C address of the device for I2C. + + - VP-supply, VCP-supply, VD_FILT-supply, VL-supply, VA-supply : + power supplies for the device, as covered in + Documentation/devicetree/bindings/regulator/regulator.txt. + +Optional properties: + + - reset-gpios : a GPIO spec for the reset pin. If specified, it will be + deasserted before communication to the codec starts. + + - interrupts : IRQ line info CS42L42. + (See Documentation/devicetree/bindings/interrupt-controller/interrupts.txt + for further information relating to interrupt properties) + + - cirrus,ts-inv : Boolean property. For jacks that invert the tip sense + polarity. Normal jacks will short tip sense pin to HS1 when headphones are + plugged in and leave tip sense floating when not plugged in. Inverting jacks + short tip sense when unplugged and float when plugged in. + + 0 = (Default) Non-inverted + 1 = Inverted + + - cirrus,ts-dbnc-rise : Debounce the rising edge of TIP_SENSE_PLUG. With no + debounce, the tip sense pin might be noisy on a plug event. + + 0 - 0ms, + 1 - 125ms, + 2 - 250ms, + 3 - 500ms, + 4 - 750ms, + 5 - (Default) 1s, + 6 - 1.25s, + 7 - 1.5s, + + - cirrus,ts-dbnc-fall : Debounce the falling edge of TIP_SENSE_UNPLUG. + With no debounce, the tip sense pin might be noisy on an unplug event. + + 0 - 0ms, + 1 - 125ms, + 2 - 250ms, + 3 - 500ms, + 4 - 750ms, + 5 - (Default) 1s, + 6 - 1.25s, + 7 - 1.5s, + + - cirrus,btn-det-init-dbnce : This sets how long the driver sleeps after + enabling button detection interrupts. After auto-detection and before + servicing button interrupts, the HS bias needs time to settle. If you + don't wait, there is possibility for erroneous button interrupt. + + 0ms - 200ms, + Default = 100ms + + - cirrus,btn-det-event-dbnce : This sets how long the driver delays after + receiving a button press interrupt. With level detect interrupts, you want + to wait a small amount of time to make sure the button press is making a + clean connection with the bias resistors. + + 0ms - 20ms, + Default = 10ms + + - cirrus,bias-lvls : For a level-detect headset button scheme, each button + will bias the mic pin to a certain voltage. To determine which button was + pressed, the driver will compare this biased voltage to sequential, + decreasing voltages and will stop when a comparator is tripped, + indicating a comparator voltage < bias voltage. This value represents a + percentage of the internally generated HS bias voltage. For different + hardware setups, a designer might want to tweak this. This is an array of + descending values for the comparator voltage. + + Array of 4 values + Each 0-63 + < x1 x2 x3 x4 > + Default = < 15 8 4 1> + + +Example: + +cs42l42: cs42l42@48 { + compatible = "cirrus,cs42l42"; + reg = <0x48>; + VA-supply = <&dummy_vreg>; + VP-supply = <&dummy_vreg>; + VCP-supply = <&dummy_vreg>; + VD_FILT-supply = <&dummy_vreg>; + VL-supply = <&dummy_vreg>; + + reset-gpios = <&axi_gpio_0 1 0>; + interrupt-parent = <&gpio0>; + interrupts = <55 8> + + cirrus,ts-inv = <0x00>; + cirrus,ts-dbnc-rise = <0x05>; + cirrus,ts-dbnc-fall = <0x00>; + cirrus,btn-det-init-dbnce = <100>; + cirrus,btn-det-event-dbnce = <10>; + cirrus,bias-lvls = <0x0F 0x08 0x04 0x01>; + cirrus,hs-bias-ramp-rate = <0x02>; +}; diff --git a/bindings/sound/cs42l52.txt b/bindings/sound/cs42l52.txt new file mode 100644 index 00000000..bc03c931 --- /dev/null +++ b/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/bindings/sound/cs42l56.txt b/bindings/sound/cs42l56.txt new file mode 100644 index 00000000..4ba520a2 --- /dev/null +++ b/bindings/sound/cs42l56.txt @@ -0,0 +1,63 @@ +CS42L52 audio CODEC + +Required properties: + + - compatible : "cirrus,cs42l56" + + - reg : the I2C address of the device for I2C + + - VA-supply, VCP-supply, VLDO-supply : power supplies for the device, + as covered in Documentation/devicetree/bindings/regulator/regulator.txt. + +Optional properties: + + - cirrus,gpio-nreset : 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 = MCLK / 4 * (N+2) + N = chgfreq_val + MCLK = Where MCLK is the frequency of the mclk signal after the MCLKDIV2 circuit. + + - cirrus,ain1a-ref-cfg, ain1b-ref-cfg : boolean, If present, AIN1A or AIN1B are configured + as a pseudo-differential input referenced to AIN1REF/AIN3A. + + - cirrus,ain2a-ref-cfg, ain2b-ref-cfg : boolean, If present, AIN2A or AIN2B are configured + as a pseudo-differential input referenced to AIN2REF/AIN3B. + + - 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 + + - cirrus,adaptive-pwr-cfg : Configures how the power to the Headphone and Lineout + Amplifiers adapt to the output signal levels. + 0 = Adapt to Volume Mode. Voltage level determined by the sum of the relevant volume settings. + 1 = Fixed - Headphone and Line Amp supply = + or - VCP/2. + 2 = Fixed - Headphone and Line Amp supply = + or - VCP. + 3 = Adapted to Signal; Voltage level is dynamically determined by the output signal. + + - cirrus,hpf-left-freq, hpf-right-freq : Sets the corner frequency (-3dB point) for the internal High-Pass + Filter. + 0 = 1.8Hz + 1 = 119Hz + 2 = 236Hz + 3 = 464Hz + + +Example: + +codec: codec@4b { + compatible = "cirrus,cs42l56"; + reg = <0x4b>; + cirrus,gpio-nreset = <&gpio 10 0>; + cirrus,chgfreq-divisor = <0x05>; + cirrus.ain1_ref_cfg; + cirrus,micbias-lvl = <5>; + VA-supply = <®_audio>; +}; diff --git a/bindings/sound/cs42l73.txt b/bindings/sound/cs42l73.txt new file mode 100644 index 00000000..80ae910d --- /dev/null +++ b/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/bindings/sound/cs42xx8.txt b/bindings/sound/cs42xx8.txt new file mode 100644 index 00000000..8619a156 --- /dev/null +++ b/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: + +cs42888: codec@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/bindings/sound/cs43130.txt b/bindings/sound/cs43130.txt new file mode 100644 index 00000000..8b1dd5ae --- /dev/null +++ b/bindings/sound/cs43130.txt @@ -0,0 +1,67 @@ +CS43130 DAC + +Required properties: + + - compatible : "cirrus,cs43130", "cirrus,cs4399", "cirrus,cs43131", + "cirrus,cs43198" + + - reg : the I2C address of the device for I2C + + - VA-supply, VP-supply, VL-supply, VCP-supply, VD-supply: + power supplies for the device, as covered in + Documentation/devicetree/bindings/regulator/regulator.txt. + + +Optional properties: + + - reset-gpios : Active low GPIO used to reset the device + + - cirrus,xtal-ibias: + When external MCLK is generated by external crystal + oscillator, CS43130 can be used to provide bias current + for external crystal. Amount of bias current sent is + set as: + 1 = 7.5uA + 2 = 12.5uA + 3 = 15uA + + - cirrus,dc-measure: + Boolean, define to enable headphone DC impedance measurement. + + - cirrus,ac-measure: + Boolean, define to enable headphone AC impedance measurement. + DC impedance must also be enabled for AC impedance measurement. + + - cirrus,dc-threshold: + Define 2 DC impedance thresholds in ohms for HP output control. + Default values are 50 and 120 Ohms. + + - cirrus,ac-freq: + Define the frequencies at which to measure HP AC impedance. + Only used if "cirrus,dc-measure" is defined. + Exactly 10 frequencies must be defined. + If this properties is undefined, by default, + following frequencies are used: + <24 43 93 200 431 928 2000 4309 9283 20000> + The above frequencies are logarithmically equally spaced. + Log base is 10. + +Example: + +cs43130: audio-codec@30 { + compatible = "cirrus,cs43130"; + reg = <0x30>; + reset-gpios = <&axi_gpio 54 0>; + VA-supply = <&dummy_vreg>; + VP-supply = <&dummy_vreg>; + VL-supply = <&dummy_vreg>; + VCP-supply = <&dummy_vreg>; + VD-supply = <&dummy_vreg>; + cirrus,xtal-ibias = <2>; + interrupt-parent = <&gpio0>; + interrupts = <55 8>; + cirrus,dc-measure; + cirrus,ac-measure; + cirrus,dc-threshold = /bits/ 16 <20 100>; + cirrus,ac-freq = /bits/ 16 <24 43 93 200 431 928 2000 4309 9283 20000>; +}; diff --git a/bindings/sound/cs4349.txt b/bindings/sound/cs4349.txt new file mode 100644 index 00000000..54c117b5 --- /dev/null +++ b/bindings/sound/cs4349.txt @@ -0,0 +1,19 @@ +CS4349 audio CODEC + +Required properties: + + - compatible : "cirrus,cs4349" + + - reg : the I2C address of the device for I2C + +Optional properties: + + - reset-gpios : a GPIO spec for the reset pin. + +Example: + +codec: cs4349@48 { + compatible = "cirrus,cs4349"; + reg = <0x48>; + reset-gpios = <&gpio 54 0>; +}; diff --git a/bindings/sound/cs53l30.txt b/bindings/sound/cs53l30.txt new file mode 100644 index 00000000..4dbfb827 --- /dev/null +++ b/bindings/sound/cs53l30.txt @@ -0,0 +1,44 @@ +CS53L30 audio CODEC + +Required properties: + + - compatible : "cirrus,cs53l30" + + - reg : the I2C address of the device + + - VA-supply, VP-supply : power supplies for the device, + as covered in Documentation/devicetree/bindings/regulator/regulator.txt. + +Optional properties: + + - reset-gpios : a GPIO spec for the reset pin. + + - mute-gpios : a GPIO spec for the MUTE pin. The active state can be either + GPIO_ACTIVE_HIGH or GPIO_ACTIVE_LOW, which would be handled + by the driver automatically. + + - cirrus,micbias-lvl : Set the output voltage level on the MICBIAS Pin. + 0 = Hi-Z + 1 = 1.80 V + 2 = 2.75 V + + - cirrus,use-sdout2 : This is a boolean property. If present, it indicates + the hardware design connects both SDOUT1 and SDOUT2 + pins to output data. Otherwise, it indicates that + only SDOUT1 is connected for data output. + * CS53l30 supports 4-channel data output in the same + * frame using two different ways: + * 1) Normal I2S mode on two data pins -- each SDOUT + * carries 2-channel data in the same time. + * 2) TDM mode on one signle data pin -- SDOUT1 carries + * 4-channel data per frame. + +Example: + +codec: cs53l30@48 { + compatible = "cirrus,cs53l30"; + reg = <0x48>; + reset-gpios = <&gpio 54 0>; + VA-supply = <&cs53l30_va>; + VP-supply = <&cs53l30_vp>; +}; diff --git a/bindings/sound/da7213.txt b/bindings/sound/da7213.txt new file mode 100644 index 00000000..58902802 --- /dev/null +++ b/bindings/sound/da7213.txt @@ -0,0 +1,41 @@ +Dialog Semiconductor DA7213 Audio Codec bindings + +====== + +Required properties: +- compatible : Should be "dlg,da7213" +- reg: Specifies the I2C slave address + +Optional properties: +- clocks : phandle and clock specifier for codec MCLK. +- clock-names : Clock name string for 'clocks' attribute, should be "mclk". + +- dlg,micbias1-lvl : Voltage (mV) for Mic Bias 1 + [<1600>, <2200>, <2500>, <3000>] +- dlg,micbias2-lvl : Voltage (mV) for Mic Bias 2 + [<1600>, <2200>, <2500>, <3000>] +- dlg,dmic-data-sel : DMIC channel select based on clock edge. + ["lrise_rfall", "lfall_rrise"] +- dlg,dmic-samplephase : When to sample audio from DMIC. + ["on_clkedge", "between_clkedge"] +- dlg,dmic-clkrate : DMIC clock frequency (Hz). + [<1500000>, <3000000>] + +====== + +Example: + + codec_i2c: da7213@1a { + compatible = "dlg,da7213"; + reg = <0x1a>; + + clocks = <&clks 201>; + clock-names = "mclk"; + + dlg,micbias1-lvl = <2500>; + dlg,micbias2-lvl = <2500>; + + dlg,dmic-data-sel = "lrise_rfall"; + dlg,dmic-samplephase = "between_clkedge"; + dlg,dmic-clkrate = <3000000>; + }; diff --git a/bindings/sound/da7218.txt b/bindings/sound/da7218.txt new file mode 100644 index 00000000..2cf30899 --- /dev/null +++ b/bindings/sound/da7218.txt @@ -0,0 +1,102 @@ +Dialog Semiconductor DA7218 Audio Codec bindings + +DA7218 is an audio codec with HP detect feature. + +====== + +Required properties: +- compatible : Should be "dlg,da7217" or "dlg,da7218" +- reg: Specifies the I2C slave address + +- VDD-supply: VDD power supply for the device +- VDDMIC-supply: VDDMIC power supply for the device +- VDDIO-supply: VDDIO power supply for the device + (See Documentation/devicetree/bindings/regulator/regulator.txt for further + information relating to regulators) + +Optional properties: +- interrupts: IRQ line info for DA7218 chip. + (See Documentation/devicetree/bindings/interrupt-controller/interrupts.txt for + further information relating to interrupt properties) +- interrupt-names : Name associated with interrupt line. Should be "wakeup" if + interrupt is to be used to wake system, otherwise "irq" should be used. +- wakeup-source: Flag to indicate this device can wake system (suspend/resume). + +- clocks : phandle and clock specifier for codec MCLK. +- clock-names : Clock name string for 'clocks' attribute, should be "mclk". + +- dlg,micbias1-lvl-millivolt : Voltage (mV) for Mic Bias 1 + [<1200>, <1600>, <1800>, <2000>, <2200>, <2400>, <2600>, <2800>, <3000>] +- dlg,micbias2-lvl-millivolt : Voltage (mV) for Mic Bias 2 + [<1200>, <1600>, <1800>, <2000>, <2200>, <2400>, <2600>, <2800>, <3000>] +- dlg,mic1-amp-in-sel : Mic1 input source type + ["diff", "se_p", "se_n"] +- dlg,mic2-amp-in-sel : Mic2 input source type + ["diff", "se_p", "se_n"] +- dlg,dmic1-data-sel : DMIC1 channel select based on clock edge. + ["lrise_rfall", "lfall_rrise"] +- dlg,dmic1-samplephase : When to sample audio from DMIC1. + ["on_clkedge", "between_clkedge"] +- dlg,dmic1-clkrate-hz : DMic1 clock frequency (Hz). + [<1500000>, <3000000>] +- dlg,dmic2-data-sel : DMic2 channel select based on clock edge. + ["lrise_rfall", "lfall_rrise"] +- dlg,dmic2-samplephase : When to sample audio from DMic2. + ["on_clkedge", "between_clkedge"] +- dlg,dmic2-clkrate-hz : DMic2 clock frequency (Hz). + [<1500000>, <3000000>] +- dlg,hp-diff-single-supply : Boolean flag, use single supply for HP + (DA7217 only) + +====== + +Optional Child node - 'da7218_hpldet' (DA7218 only): + +Optional properties: +- dlg,jack-rate-us : Time between jack detect measurements (us) + [<5>, <10>, <20>, <40>, <80>, <160>, <320>, <640>] +- dlg,jack-debounce : Number of debounce measurements taken for jack detect + [<0>, <2>, <3>, <4>] +- dlg,jack-threshold-pct : Threshold level for jack detection (% of VDD) + [<84>, <88>, <92>, <96>] +- dlg,comp-inv : Boolean flag, invert comparator output +- dlg,hyst : Boolean flag, enable hysteresis +- dlg,discharge : Boolean flag, auto discharge of Mic Bias on jack removal + +====== + +Example: + + codec: da7218@1a { + compatible = "dlg,da7218"; + reg = <0x1a>; + interrupt-parent = <&gpio6>; + interrupts = <11 IRQ_TYPE_LEVEL_LOW>; + wakeup-source; + + VDD-supply = <®_audio>; + VDDMIC-supply = <®_audio>; + VDDIO-supply = <®_audio>; + + clocks = <&clks 201>; + clock-names = "mclk"; + + dlg,micbias1-lvl-millivolt = <2600>; + dlg,micbias2-lvl-millivolt = <2600>; + dlg,mic1-amp-in-sel = "diff"; + dlg,mic2-amp-in-sel = "diff"; + + dlg,dmic1-data-sel = "lrise_rfall"; + dlg,dmic1-samplephase = "on_clkedge"; + dlg,dmic1-clkrate-hz = <3000000>; + dlg,dmic2-data-sel = "lrise_rfall"; + dlg,dmic2-samplephase = "on_clkedge"; + dlg,dmic2-clkrate-hz = <3000000>; + + da7218_hpldet { + dlg,jack-rate-us = <40>; + dlg,jack-debounce = <2>; + dlg,jack-threshold-pct = <84>; + dlg,hyst; + }; + }; diff --git a/bindings/sound/da7219.txt b/bindings/sound/da7219.txt new file mode 100644 index 00000000..e9d0baeb --- /dev/null +++ b/bindings/sound/da7219.txt @@ -0,0 +1,112 @@ +Dialog Semiconductor DA7219 Audio Codec bindings + +DA7219 is an audio codec with advanced accessory detect features. + +====== + +Required properties: +- compatible : Should be "dlg,da7219" +- reg: Specifies the I2C slave address + +- interrupts : IRQ line info for DA7219. + (See Documentation/devicetree/bindings/interrupt-controller/interrupts.txt for + further information relating to interrupt properties) + +- VDD-supply: VDD power supply for the device +- VDDMIC-supply: VDDMIC power supply for the device +- VDDIO-supply: VDDIO power supply for the device + (See Documentation/devicetree/bindings/regulator/regulator.txt for further + information relating to regulators) + +Optional properties: +- interrupt-names : Name associated with interrupt line. Should be "wakeup" if + interrupt is to be used to wake system, otherwise "irq" should be used. +- wakeup-source: Flag to indicate this device can wake system (suspend/resume). + +- #clock-cells : Should be set to '<0>', only one clock source provided; +- clock-output-names : Name given for DAI clocks output; + +- clocks : phandle and clock specifier for codec MCLK. +- clock-names : Clock name string for 'clocks' attribute, should be "mclk". + +- dlg,micbias-lvl : Voltage (mV) for Mic Bias + [<1600>, <1800>, <2000>, <2200>, <2400>, <2600>] +- dlg,mic-amp-in-sel : Mic input source type + ["diff", "se_p", "se_n"] + +Deprecated properties: +- dlg,ldo-lvl : Required internal LDO voltage (mV) level for digital engine + (LDO unavailable in production HW so property no longer required). + +====== + +Child node - 'da7219_aad': + +Optional properties: +- dlg,micbias-pulse-lvl : Mic bias higher voltage pulse level (mV). + [<2800>, <2900>] +- dlg,micbias-pulse-time : Mic bias higher voltage pulse duration (ms) +- dlg,btn-cfg : Periodic button press measurements for 4-pole jack (ms) + [<2>, <5>, <10>, <50>, <100>, <200>, <500>] +- dlg,mic-det-thr : Impedance threshold for mic detection measurement (Ohms) + [<200>, <500>, <750>, <1000>] +- dlg,jack-ins-deb : Debounce time for jack insertion (ms) + [<5>, <10>, <20>, <50>, <100>, <200>, <500>, <1000>] +- dlg,jack-det-rate: Jack type detection latency (3/4 pole) + ["32ms_64ms", "64ms_128ms", "128ms_256ms", "256ms_512ms"] +- dlg,jack-rem-deb : Debounce time for jack removal (ms) + [<1>, <5>, <10>, <20>] +- dlg,a-d-btn-thr : Impedance threshold between buttons A and D + [0x0 - 0xFF] +- dlg,d-b-btn-thr : Impedance threshold between buttons D and B + [0x0 - 0xFF] +- dlg,b-c-btn-thr : Impedance threshold between buttons B and C + [0x0 - 0xFF] +- dlg,c-mic-btn-thr : Impedance threshold between button C and Mic + [0x0 - 0xFF] +- dlg,btn-avg : Number of 8-bit readings for averaged button measurement + [<1>, <2>, <4>, <8>] +- dlg,adc-1bit-rpt : Repeat count for 1-bit button measurement + [<1>, <2>, <4>, <8>] + +====== + +Example: + + codec: da7219@1a { + compatible = "dlg,da7219"; + reg = <0x1a>; + + interrupt-parent = <&gpio6>; + interrupts = <11 IRQ_TYPE_LEVEL_LOW>; + + VDD-supply = <®_audio>; + VDDMIC-supply = <®_audio>; + VDDIO-supply = <®_audio>; + + #clock-cells = <0>; + clock-output-names = "dai-clks"; + + clocks = <&clks 201>; + clock-names = "mclk"; + + dlg,ldo-lvl = <1200>; + dlg,micbias-lvl = <2600>; + dlg,mic-amp-in-sel = "diff"; + + da7219_aad { + dlg,btn-cfg = <50>; + dlg,mic-det-thr = <500>; + dlg,jack-ins-deb = <20>; + dlg,jack-det-rate = "32ms_64ms"; + dlg,jack-rem-deb = <1>; + + dlg,a-d-btn-thr = <0xa>; + dlg,d-b-btn-thr = <0x16>; + dlg,b-c-btn-thr = <0x21>; + dlg,c-mic-btn-thr = <0x3E>; + + dlg,btn-avg = <4>; + dlg,adc-1bit-rpt = <1>; + }; + }; diff --git a/bindings/sound/da9055.txt b/bindings/sound/da9055.txt new file mode 100644 index 00000000..ed1b7cc6 --- /dev/null +++ b/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/bindings/sound/davinci-evm-audio.txt b/bindings/sound/davinci-evm-audio.txt new file mode 100644 index 00000000..963e1005 --- /dev/null +++ b/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/bindings/sound/davinci-mcasp-audio.txt b/bindings/sound/davinci-mcasp-audio.txt new file mode 100644 index 00000000..46bc9829 --- /dev/null +++ b/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 +- 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>; + interrupt-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/bindings/sound/davinci-mcbsp.txt b/bindings/sound/davinci-mcbsp.txt new file mode 100644 index 00000000..3ffc2562 --- /dev/null +++ b/bindings/sound/davinci-mcbsp.txt @@ -0,0 +1,50 @@ +Texas Instruments DaVinci McBSP module +~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + +This binding describes the "Multi-channel Buffered Serial Port" (McBSP) +audio interface found in some TI DaVinci processors like the OMAP-L138 or AM180x. + + +Required properties: +~~~~~~~~~~~~~~~~~~~~ +- compatible : + "ti,da850-mcbsp" : for DA850, AM180x and OPAM-L138 platforms + +- reg : physical base address and length of the controller memory mapped + region(s). +- reg-names : Should contain: + * "mpu" for the main registers (required). + * "dat" for the data FIFO (optional). + +- dmas: three element list of DMA controller phandles, DMA request line and + TC channel ordered triplets. +- 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: +~~~~~~~~~~~~~~~~~~~~ +- interrupts : Interrupt numbers for McBSP +- interrupt-names : Known interrupt names are "rx" and "tx" + +- 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 + +Example (AM1808): +~~~~~~~~~~~~~~~~~ + +mcbsp0: mcbsp@1d10000 { + compatible = "ti,da850-mcbsp"; + pinctrl-names = "default"; + pinctrl-0 = <&mcbsp0_pins>; + + reg = <0x00110000 0x1000>, + <0x00310000 0x1000>; + reg-names = "mpu", "dat"; + interrupts = <97 98>; + interrupt-names = "rx", "tx"; + dmas = <&edma0 3 1 + &edma0 2 1>; + dma-names = "tx", "rx"; +}; diff --git a/bindings/sound/designware-i2s.txt b/bindings/sound/designware-i2s.txt new file mode 100644 index 00000000..6a536d57 --- /dev/null +++ b/bindings/sound/designware-i2s.txt @@ -0,0 +1,35 @@ +DesignWare I2S controller + +Required properties: + - compatible : Must be "snps,designware-i2s" + - reg : Must contain the I2S core's registers location and length + - clocks : Pairs of phandle and specifier referencing the controller's + clocks. The controller expects one clock: the clock used as the sampling + rate reference clock sample. + - clock-names : "i2sclk" 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 one or two dma channels: one for transmit and + one for receive. + - dma-names : "tx" for the transmit channel, "rx" for the receive channel. + +Optional properties: + - interrupts: The interrupt line number for the I2S controller. Add this + parameter if the I2S controller that you are using does not support DMA. + +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: + + soc_i2s: i2s@7ff90000 { + compatible = "snps,designware-i2s"; + reg = <0x0 0x7ff90000 0x0 0x1000>; + clocks = <&scpi_i2sclk 0>; + clock-names = "i2sclk"; + #sound-dai-cells = <0>; + dmas = <&dma0 5>; + dma-names = "tx"; + }; diff --git a/bindings/sound/dmic.txt b/bindings/sound/dmic.txt new file mode 100644 index 00000000..e957b413 --- /dev/null +++ b/bindings/sound/dmic.txt @@ -0,0 +1,20 @@ +Device-Tree bindings for Digital microphone (DMIC) codec + +This device support generic PDM digital microphone. + +Required properties: + - compatible: should be "dmic-codec". + +Optional properties: + - dmicen-gpios: GPIO specifier for dmic to control start and stop + - num-channels: Number of microphones on this DAI + - wakeup-delay-ms: Delay (in ms) after enabling the DMIC + +Example node: + + dmic_codec: dmic@0 { + compatible = "dmic-codec"; + dmicen-gpios = <&gpio4 3 GPIO_ACTIVE_HIGH>; + num-channels = <1>; + wakeup-delay-ms <50>; + }; diff --git a/bindings/sound/es8328.txt b/bindings/sound/es8328.txt new file mode 100644 index 00000000..33fbf058 --- /dev/null +++ b/bindings/sound/es8328.txt @@ -0,0 +1,38 @@ +Everest ES8328 audio CODEC + +This device supports both I2C and SPI. + +Required properties: + + - compatible : Should be "everest,es8328" or "everest,es8388" + - DVDD-supply : Regulator providing digital core supply voltage 1.8 - 3.6V + - AVDD-supply : Regulator providing analog supply voltage 3.3V + - PVDD-supply : Regulator providing digital IO supply voltage 1.8 - 3.6V + - IPVDD-supply : Regulator providing analog output voltage 3.3V + - clocks : A 22.5792 or 11.2896 MHz clock + - 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 + * LINPUT1 + * RINPUT1 + * LINPUT2 + * RINPUT2 + * Mic Bias + + +Example: + +codec: es8328@11 { + compatible = "everest,es8328"; + DVDD-supply = <®_3p3v>; + AVDD-supply = <®_3p3v>; + PVDD-supply = <®_3p3v>; + HPVDD-supply = <®_3p3v>; + clocks = <&clks 169>; + reg = <0x11>; +}; diff --git a/bindings/sound/eukrea-tlv320.txt b/bindings/sound/eukrea-tlv320.txt new file mode 100644 index 00000000..6dfa88c4 --- /dev/null +++ b/bindings/sound/eukrea-tlv320.txt @@ -0,0 +1,26 @@ +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/bindings/sound/everest,es7134.txt b/bindings/sound/everest,es7134.txt new file mode 100644 index 00000000..09166606 --- /dev/null +++ b/bindings/sound/everest,es7134.txt @@ -0,0 +1,15 @@ +ES7134 i2s DA converter + +Required properties: +- compatible : "everest,es7134" or + "everest,es7144" or + "everest,es7154" +- VDD-supply : regulator phandle for the VDD supply +- PVDD-supply: regulator phandle for the PVDD supply for the es7154 + +Example: + +i2s_codec: external-codec { + compatible = "everest,es7134"; + VDD-supply = <&vcc_5v>; +}; diff --git a/bindings/sound/everest,es7241.txt b/bindings/sound/everest,es7241.txt new file mode 100644 index 00000000..28f82cf4 --- /dev/null +++ b/bindings/sound/everest,es7241.txt @@ -0,0 +1,28 @@ +ES7241 i2s AD converter + +Required properties: +- compatible : "everest,es7241" +- VDDP-supply: regulator phandle for the VDDA supply +- VDDA-supply: regulator phandle for the VDDP supply +- VDDD-supply: regulator phandle for the VDDD supply + +Optional properties: +- reset-gpios: gpio connected to the reset pin +- m0-gpios : gpio connected to the m0 pin +- m1-gpios : gpio connected to the m1 pin +- everest,sdout-pull-down: + Format used by the serial interface is controlled by pulling + the sdout. If the sdout is pulled down, leftj format is used. + If this property is not provided, sdout is assumed to pulled + up and i2s format is used + +Example: + +linein: audio-codec@2 { + #sound-dai-cells = <0>; + compatible = "everest,es7241"; + VDDA-supply = <&vcc_3v3>; + VDDP-supply = <&vcc_3v3>; + VDDD-supply = <&vcc_3v3>; + reset-gpios = <&gpio GPIOH_42>; +}; diff --git a/bindings/sound/fsl,asrc.txt b/bindings/sound/fsl,asrc.txt new file mode 100644 index 00000000..1d4d9f93 --- /dev/null +++ b/bindings/sound/fsl,asrc.txt @@ -0,0 +1,66 @@ +Freescale Asynchronous Sample Rate Converter (ASRC) Controller + +The Asynchronous Sample Rate Converter (ASRC) converts the sampling rate of a +signal associated with an input clock into a signal associated with a different +output clock. The driver currently works as a Front End of DPCM with other Back +Ends Audio controller such as ESAI, SSI and SAI. It has three pairs to support +three substreams within totally 10 channels. + +Required properties: + + - compatible : Contains "fsl,imx35-asrc" or "fsl,imx53-asrc". + + - 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 : Contains "rxa", "rxb", "rxc", "txa", "txb" and "txc". + + - clocks : Contains an entry for each entry in clock-names. + + - clock-names : Contains the following entries + "mem" Peripheral access clock to access registers. + "ipg" Peripheral clock to driver module. + "asrck_<0-f>" Clock sources for input and output clock. + "spba" The spba clock is required when ASRC is placed as a + bus slave of the Shared Peripheral Bus and when two + or more bus masters (CPU, DMA or DSP) try to access + it. This property is optional depending on the SoC + design. + + - fsl,asrc-rate : Defines a mutual sample rate used by DPCM Back Ends. + + - fsl,asrc-width : Defines a mutual sample width used by DPCM Back Ends. + +Optional properties: + + - big-endian : If this property is absent, the little endian mode + will be in use as default. Otherwise, the big endian + mode will be in use for all the device registers. + +Example: + +asrc: asrc@2034000 { + compatible = "fsl,imx53-asrc"; + reg = <0x02034000 0x4000>; + interrupts = <0 50 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&clks 107>, <&clks 107>, <&clks 0>, + <&clks 0>, <&clks 0>, <&clks 0>, <&clks 0>, + <&clks 0>, <&clks 0>, <&clks 0>, <&clks 0>, + <&clks 0>, <&clks 0>, <&clks 0>, <&clks 0>, + <&clks 107>, <&clks 0>, <&clks 0>; + clock-names = "mem", "ipg", "asrck0", + "asrck_1", "asrck_2", "asrck_3", "asrck_4", + "asrck_5", "asrck_6", "asrck_7", "asrck_8", + "asrck_9", "asrck_a", "asrck_b", "asrck_c", + "asrck_d", "asrck_e", "asrck_f"; + dmas = <&sdma 17 23 1>, <&sdma 18 23 1>, <&sdma 19 23 1>, + <&sdma 20 23 1>, <&sdma 21 23 1>, <&sdma 22 23 1>; + dma-names = "rxa", "rxb", "rxc", + "txa", "txb", "txc"; + fsl,asrc-rate = <48000>; + fsl,asrc-width = <16>; +}; diff --git a/bindings/sound/fsl,esai.txt b/bindings/sound/fsl,esai.txt new file mode 100644 index 00000000..5b991436 --- /dev/null +++ b/bindings/sound/fsl,esai.txt @@ -0,0 +1,64 @@ +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" or + "fsl,vf610-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. + "spba" The spba clock is required when ESAI is placed as a + bus slave of the Shared Peripheral Bus and when two + or more bus masters (CPU, DMA or DSP) try to access + it. This property is optional depending on the SoC + design. + + - 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. + +Optional properties: + + - big-endian : If this property is absent, the native endian mode + will be in use as default, or the big endian mode + will be in use for all the device registers. + +Example: + +esai: esai@2024000 { + 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; +}; diff --git a/bindings/sound/fsl,spdif.txt b/bindings/sound/fsl,spdif.txt new file mode 100644 index 00000000..8b324f82 --- /dev/null +++ b/bindings/sound/fsl,spdif.txt @@ -0,0 +1,64 @@ +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. + "spba" The spba clock is required when SPDIF is placed as a + bus slave of the Shared Peripheral Bus and when two + or more bus masters (CPU, DMA or DSP) try to access + it. This property is optional depending on the SoC + design. + +Optional properties: + + - big-endian : If this property is absent, the native endian mode + will be in use as default, or the big endian mode + will be in use for all the device registers. + +Example: + +spdif: spdif@2004000 { + 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; +}; diff --git a/bindings/sound/fsl,ssi.txt b/bindings/sound/fsl,ssi.txt new file mode 100644 index 00000000..7e15a85c --- /dev/null +++ b/bindings/sound/fsl,ssi.txt @@ -0,0 +1,87 @@ +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. +- fsl,fifo-depth: The number of elements in the transmit and receive FIFOs. + This number is the maximum allowed value for SFCSR[TFWM0]. + - 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 AC97 interface only. + "ac97-slave" - AC97 mode, SSI is clock slave + "ac97-master" - AC97 mode, SSI is clock master +- 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. +- 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. + Only used on Power Architecture. +- 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. + Only used on Power Architecture. + +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/bindings/sound/fsl-asoc-card.txt b/bindings/sound/fsl-asoc-card.txt new file mode 100644 index 00000000..c60a5732 --- /dev/null +++ b/bindings/sound/fsl-asoc-card.txt @@ -0,0 +1,94 @@ +Freescale Generic ASoC Sound Card with ASRC support + +The Freescale Generic ASoC Sound Card can be used, ideally, for all Freescale +SoCs connecting with external CODECs. + +The idea of this generic sound card is a bit like ASoC Simple Card. However, +for Freescale SoCs (especially those released in recent years), most of them +have ASRC (Documentation/devicetree/bindings/sound/fsl,asrc.txt) inside. And +this is a specific feature that might be painstakingly controlled and merged +into the Simple Card. + +So having this generic sound card allows all Freescale SoC users to benefit +from the simplification of a new card support and the capability of the wide +sample rates support through ASRC. + +Note: The card is initially designed for those sound cards who use AC'97, I2S + and PCM DAI formats. However, it'll be also possible to support those non + AC'97/I2S/PCM type sound cards, such as S/PDIF audio and HDMI audio, as + long as the driver has been properly upgraded. + + +The compatible list for this generic sound card currently: + "fsl,imx-audio-ac97" + + "fsl,imx-audio-cs42888" + + "fsl,imx-audio-cs427x" + (compatible with CS4271 and CS4272) + + "fsl,imx-audio-wm8962" + + "fsl,imx-audio-sgtl5000" + (compatible with Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt) + + "fsl,imx-audio-wm8960" + +Required properties: + + - compatible : Contains one of entries in the compatible list. + + - model : The user-visible name of this sound complex + + - audio-cpu : The phandle of an CPU DAI controller + + - audio-codec : The phandle of an 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. There're a few pre-designed board connectors: + * Line Out Jack + * Line In Jack + * Headphone Jack + * Mic Jack + * Ext Spk + * AMIC (stands for Analog Microphone Jack) + * DMIC (stands for Digital Microphone Jack) + + Note: The "Mic Jack" and "AMIC" are redundant while + coexisting in order to support the old bindings + of wm8962 and sgtl5000. + +Optional properties: + + - audio-asrc : The phandle of ASRC. It can be absent if there's no + need to add ASRC support via DPCM. + +Optional unless SSI is selected as a CPU DAI: + + - 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 + +Example: +sound-cs42888 { + compatible = "fsl,imx-audio-cs42888"; + model = "cs42888-audio"; + audio-cpu = <&esai>; + audio-asrc = <&asrc>; + audio-codec = <&cs42888>; + audio-routing = + "Line Out Jack", "AOUT1L", + "Line Out Jack", "AOUT1R", + "Line Out Jack", "AOUT2L", + "Line Out Jack", "AOUT2R", + "Line Out Jack", "AOUT3L", + "Line Out Jack", "AOUT3R", + "Line Out Jack", "AOUT4L", + "Line Out Jack", "AOUT4R", + "AIN1L", "Line In Jack", + "AIN1R", "Line In Jack", + "AIN2L", "Line In Jack", + "AIN2R", "Line In Jack"; +}; diff --git a/bindings/sound/fsl-sai.txt b/bindings/sound/fsl-sai.txt new file mode 100644 index 00000000..dd9e5973 --- /dev/null +++ b/bindings/sound/fsl-sai.txt @@ -0,0 +1,80 @@ +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", + "fsl,imx6sx-sai" or "fsl,imx6ul-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 "bus" for register access and + "mclk1", "mclk2", "mclk3" for bit clock and frame + clock providing. + - 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. + + - lsb-first : Configures whether the LSB or the MSB is transmitted + first for the fifo data. If this property is absent, + the MSB is transmitted first as default, or the LSB + is transmitted first. + + - fsl,sai-synchronous-rx: This is a boolean property. If present, indicating + that SAI will work in the synchronous mode (sync Tx + with Rx) which means both the transimitter and the + receiver will send and receive data by following + receiver's bit clocks and frame sync clocks. + + - fsl,sai-asynchronous: This is a boolean property. If present, indicating + that SAI will work in the asynchronous mode, which + means both transimitter and receiver will send and + receive data by following their own bit clocks and + frame sync clocks separately. + +Optional properties: + + - big-endian : Boolean property, required if all the SAI + registers are big-endian rather than little-endian. + +Optional properties (for mx6ul): + + - fsl,sai-mclk-direction-output: This is a boolean property. If present, + indicates that SAI will output the SAI MCLK clock. + +Note: +- If both fsl,sai-asynchronous and fsl,sai-synchronous-rx are absent, the + default synchronous mode (sync Rx with Tx) will be used, which means both + transimitter and receiver will send and receive data by following clocks + of transimitter. +- fsl,sai-asynchronous and fsl,sai-synchronous-rx are exclusive. + +Example: +sai2: sai@40031000 { + compatible = "fsl,vf610-sai"; + reg = <0x40031000 0x1000>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_sai2_1>; + clocks = <&clks VF610_CLK_PLATFORM_BUS>, + <&clks VF610_CLK_SAI2>, + <&clks 0>, <&clks 0>; + clock-names = "bus", "mclk1", "mclk2", "mclk3"; + dma-names = "tx", "rx"; + dmas = <&edma0 0 VF610_EDMA_MUXID0_SAI2_TX>, + <&edma0 0 VF610_EDMA_MUXID0_SAI2_RX>; + big-endian; + lsb-first; +}; diff --git a/bindings/sound/gtm601.txt b/bindings/sound/gtm601.txt new file mode 100644 index 00000000..5efc8c06 --- /dev/null +++ b/bindings/sound/gtm601.txt @@ -0,0 +1,13 @@ +GTM601 UMTS modem audio interface CODEC + +This device has no configuration interface. Sample rate is fixed - 8kHz. + +Required properties: + + - compatible : "option,gtm601" + +Example: + +codec: gtm601_codec { + compatible = "option,gtm601"; +}; diff --git a/bindings/sound/hdmi.txt b/bindings/sound/hdmi.txt new file mode 100644 index 00000000..56407c30 --- /dev/null +++ b/bindings/sound/hdmi.txt @@ -0,0 +1,16 @@ +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"; + }; diff --git a/bindings/sound/hisilicon,hi6210-i2s.txt b/bindings/sound/hisilicon,hi6210-i2s.txt new file mode 100644 index 00000000..7a296784 --- /dev/null +++ b/bindings/sound/hisilicon,hi6210-i2s.txt @@ -0,0 +1,42 @@ +* Hisilicon 6210 i2s controller + +Required properties: + +- compatible: should be one of the following: + - "hisilicon,hi6210-i2s" +- reg: physical base address of the i2s controller unit and length of + memory mapped region. +- interrupts: should contain the i2s interrupt. +- clocks: a list of phandle + clock-specifier pairs, one for each entry + in clock-names. +- clock-names: should contain following: + - "dacodec" + - "i2s-base" +- dmas: DMA specifiers for tx dma. See the DMA client binding, + Documentation/devicetree/bindings/dma/dma.txt +- dma-names: should be "tx" and "rx" +- hisilicon,sysctrl-syscon: phandle to sysctrl syscon +- #sound-dai-cells: Should be set to 1 (for multi-dai) + - The dai cell indexes reference the following interfaces: + 0: S2 interface + (Currently that is the only one available, but more may be + supported in the future) + +Example for the hi6210 i2s controller: + +i2s0: i2s@f7118000{ + compatible = "hisilicon,hi6210-i2s"; + reg = <0x0 0xf7118000 0x0 0x8000>; /* i2s unit */ + interrupts = <GIC_SPI 123 IRQ_TYPE_LEVEL_HIGH>; /* 155 "DigACodec_intr"-32 */ + clocks = <&sys_ctrl HI6220_DACODEC_PCLK>, + <&sys_ctrl HI6220_BBPPLL0_DIV>; + clock-names = "dacodec", "i2s-base"; + dmas = <&dma0 15 &dma0 14>; + dma-names = "rx", "tx"; + hisilicon,sysctrl-syscon = <&sys_ctrl>; + #sound-dai-cells = <1>; +}; + +Then when referencing the i2s controller: + sound-dai = <&i2s0 0>; /* index 0 => S2 interface */ + diff --git a/bindings/sound/ics43432.txt b/bindings/sound/ics43432.txt new file mode 100644 index 00000000..b02e3a6c --- /dev/null +++ b/bindings/sound/ics43432.txt @@ -0,0 +1,17 @@ +Invensense ICS-43432 MEMS microphone with I2S output. + +There are no software configuration options for this device, indeed, the only +host connection is the I2S interface. Apart from requirements on clock +frequency (460 kHz to 3.379 MHz according to the data sheet) there must be +64 clock cycles in each stereo output frame; 24 of the 32 available bits +contain audio data. A hardware pin determines if the device outputs data +on the left or right channel of the I2S frame. + +Required properties: + - compatible : Must be "invensense,ics43432" + +Example: + + ics43432: ics43432 { + compatible = "invensense,ics43432"; + }; diff --git a/bindings/sound/img,i2s-in.txt b/bindings/sound/img,i2s-in.txt new file mode 100644 index 00000000..423265cf --- /dev/null +++ b/bindings/sound/img,i2s-in.txt @@ -0,0 +1,47 @@ +Imagination Technologies I2S Input Controller + +Required Properties: + + - compatible : Compatible list, must contain "img,i2s-in" + + - #sound-dai-cells : Must be equal to 0 + + - reg : Offset and length of the register set for the device + + - clocks : Contains an entry for each entry in clock-names + + - clock-names : Must include the following entry: + "sys" The system clock + + - dmas: Contains an entry for each entry in dma-names. + + - dma-names: Must include the following entry: + "rx" Single DMA channel used by all active I2S channels + + - img,i2s-channels : Number of I2S channels instantiated in the I2S in block + +Optional Properties: + + - interrupts : Contains the I2S in interrupts. Depending on + the configuration, there may be no interrupts, one interrupt, + or an interrupt per I2S channel. For the case where there is + one interrupt per channel, the interrupts should be listed + in ascending channel order + + - resets: Contains a phandle to the I2S in reset signal + + - reset-names: Contains the reset signal name "rst" + +Example: + +i2s_in: i2s-in@18100800 { + compatible = "img,i2s-in"; + reg = <0x18100800 0x200>; + interrupts = <GIC_SHARED 7 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&mdc 30 0xffffffff 0>; + dma-names = "rx"; + clocks = <&cr_periph SYS_CLK_I2S_IN>; + clock-names = "sys"; + img,i2s-channels = <6>; + #sound-dai-cells = <0>; +}; diff --git a/bindings/sound/img,i2s-out.txt b/bindings/sound/img,i2s-out.txt new file mode 100644 index 00000000..6b0ee9b7 --- /dev/null +++ b/bindings/sound/img,i2s-out.txt @@ -0,0 +1,51 @@ +Imagination Technologies I2S Output Controller + +Required Properties: + + - compatible : Compatible list, must contain "img,i2s-out" + + - #sound-dai-cells : Must be equal to 0 + + - reg : Offset and length of the register set for the device + + - clocks : Contains an entry for each entry in clock-names + + - clock-names : Must include the following entries: + "sys" The system clock + "ref" The reference clock + + - dmas: Contains an entry for each entry in dma-names. + + - dma-names: Must include the following entry: + "tx" Single DMA channel used by all active I2S channels + + - img,i2s-channels : Number of I2S channels instantiated in the I2S out block + + - resets: Contains a phandle to the I2S out reset signal + + - reset-names: Contains the reset signal name "rst" + +Optional Properties: + + - interrupts : Contains the I2S out interrupts. Depending on + the configuration, there may be no interrupts, one interrupt, + or an interrupt per I2S channel. For the case where there is + one interrupt per channel, the interrupts should be listed + in ascending channel order + +Example: + +i2s_out: i2s-out@18100a00 { + compatible = "img,i2s-out"; + reg = <0x18100A00 0x200>; + interrupts = <GIC_SHARED 13 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&mdc 23 0xffffffff 0>; + dma-names = "tx"; + clocks = <&cr_periph SYS_CLK_I2S_OUT>, + <&clk_core CLK_I2S>; + clock-names = "sys", "ref"; + img,i2s-channels = <6>; + resets = <&pistachio_reset PISTACHIO_RESET_I2S_OUT>; + reset-names = "rst"; + #sound-dai-cells = <0>; +}; diff --git a/bindings/sound/img,parallel-out.txt b/bindings/sound/img,parallel-out.txt new file mode 100644 index 00000000..37a3f94c --- /dev/null +++ b/bindings/sound/img,parallel-out.txt @@ -0,0 +1,44 @@ +Imagination Technologies Parallel Output Controller + +Required Properties: + + - compatible : Compatible list, must contain "img,parallel-out". + + - #sound-dai-cells : Must be equal to 0 + + - reg : Offset and length of the register set for the device. + + - dmas: Contains an entry for each entry in dma-names. + + - dma-names: Must include the following entry: + "tx" + + - clocks : Contains an entry for each entry in clock-names. + + - clock-names : Includes the following entries: + "sys" The system clock + "ref" The reference clock + + - resets: Contains a phandle to the parallel out reset signal + + - reset-names: Contains the reset signal name "rst" + +Optional Properties: + + - interrupts : Contains the parallel out interrupt, if present + +Example: + +parallel_out: parallel-out@18100c00 { + compatible = "img,parallel-out"; + reg = <0x18100C00 0x100>; + interrupts = <GIC_SHARED 19 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&mdc 16 0xffffffff 0>; + dma-names = "tx"; + clocks = <&cr_periph SYS_CLK_PAUD_OUT>, + <&clk_core CLK_AUDIO_DAC>; + clock-names = "sys", "ref"; + resets = <&pistachio_reset PISTACHIO_RESET_PRL_OUT>; + reset-names = "rst"; + #sound-dai-cells = <0>; +}; diff --git a/bindings/sound/img,pistachio-internal-dac.txt b/bindings/sound/img,pistachio-internal-dac.txt new file mode 100644 index 00000000..4cc18fc0 --- /dev/null +++ b/bindings/sound/img,pistachio-internal-dac.txt @@ -0,0 +1,18 @@ +Pistachio internal DAC DT bindings + +Required properties: + + - compatible: "img,pistachio-internal-dac" + + - img,cr-top : Must contain a phandle to the top level control syscon + node which contains the internal dac control registers + + - VDD-supply : Digital power supply regulator (+1.8V or +3.3V) + +Examples: + +internal_dac: internal-dac { + compatible = "img,pistachio-internal-dac"; + img,cr-top = <&cr_top>; + VDD-supply = <&supply3v3>; +}; diff --git a/bindings/sound/img,spdif-in.txt b/bindings/sound/img,spdif-in.txt new file mode 100644 index 00000000..f7ea8c87 --- /dev/null +++ b/bindings/sound/img,spdif-in.txt @@ -0,0 +1,41 @@ +Imagination Technologies SPDIF Input Controller + +Required Properties: + + - compatible : Compatible list, must contain "img,spdif-in" + + - #sound-dai-cells : Must be equal to 0 + + - reg : Offset and length of the register set for the device + + - dmas: Contains an entry for each entry in dma-names. + + - dma-names: Must include the following entry: + "rx" + + - clocks : Contains an entry for each entry in clock-names + + - clock-names : Includes the following entries: + "sys" The system clock + +Optional Properties: + + - resets: Should contain a phandle to the spdif in reset signal, if any + + - reset-names: Should contain the reset signal name "rst", if a + reset phandle is given + + - interrupts : Contains the spdif in interrupt, if present + +Example: + +spdif_in: spdif-in@18100e00 { + compatible = "img,spdif-in"; + reg = <0x18100E00 0x100>; + interrupts = <GIC_SHARED 20 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&mdc 15 0xffffffff 0>; + dma-names = "rx"; + clocks = <&cr_periph SYS_CLK_SPDIF_IN>; + clock-names = "sys"; + #sound-dai-cells = <0>; +}; diff --git a/bindings/sound/img,spdif-out.txt b/bindings/sound/img,spdif-out.txt new file mode 100644 index 00000000..413ed8b0 --- /dev/null +++ b/bindings/sound/img,spdif-out.txt @@ -0,0 +1,44 @@ +Imagination Technologies SPDIF Output Controller + +Required Properties: + + - compatible : Compatible list, must contain "img,spdif-out" + + - #sound-dai-cells : Must be equal to 0 + + - reg : Offset and length of the register set for the device + + - dmas: Contains an entry for each entry in dma-names. + + - dma-names: Must include the following entry: + "tx" + + - clocks : Contains an entry for each entry in clock-names. + + - clock-names : Includes the following entries: + "sys" The system clock + "ref" The reference clock + + - resets: Contains a phandle to the spdif out reset signal + + - reset-names: Contains the reset signal name "rst" + +Optional Properties: + + - interrupts : Contains the parallel out interrupt, if present + +Example: + +spdif_out: spdif-out@18100d00 { + compatible = "img,spdif-out"; + reg = <0x18100D00 0x100>; + interrupts = <GIC_SHARED 21 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&mdc 14 0xffffffff 0>; + dma-names = "tx"; + clocks = <&cr_periph SYS_CLK_SPDIF_OUT>, + <&clk_core CLK_SPDIF>; + clock-names = "sys", "ref"; + resets = <&pistachio_reset PISTACHIO_RESET_SPDIF_OUT>; + reset-names = "rst"; + #sound-dai-cells = <0>; +}; diff --git a/bindings/sound/imx-audio-es8328.txt b/bindings/sound/imx-audio-es8328.txt new file mode 100644 index 00000000..07b68ab2 --- /dev/null +++ b/bindings/sound/imx-audio-es8328.txt @@ -0,0 +1,60 @@ +Freescale i.MX audio complex with ES8328 codec + +Required properties: +- compatible : "fsl,imx-audio-es8328" +- model : The user-visible name of this sound complex +- ssi-controller : The phandle of the i.MX SSI controller +- jack-gpio : Optional GPIO for headphone jack +- audio-amp-supply : Power regulator for speaker amps +- audio-codec : The phandle of the ES8328 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, ES8328 + pins, and the jacks on the board: + + Power supplies: + * audio-amp + + ES8328 pins: + * LOUT1 + * LOUT2 + * ROUT1 + * ROUT2 + * LINPUT1 + * LINPUT2 + * RINPUT1 + * RINPUT2 + * Mic PGA + + Board connectors: + * Headphone + * Speaker + * Mic Jack +- 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 (AUDMIX) + +Note: The AUDMUX port numbering should start at 1, which is consistent with +hardware manual. + +Example: + +sound { + compatible = "fsl,imx-audio-es8328"; + model = "imx-audio-es8328"; + ssi-controller = <&ssi1>; + audio-codec = <&codec>; + jack-gpio = <&gpio5 15 0>; + audio-amp-supply = <®_audio_amp>; + audio-routing = + "Speaker", "LOUT2", + "Speaker", "ROUT2", + "Speaker", "audio-amp", + "Headphone", "ROUT1", + "Headphone", "LOUT1", + "LINPUT1", "Mic Jack", + "RINPUT1", "Mic Jack", + "Mic Jack", "Mic Bias"; + mux-int-port = <1>; + mux-ext-port = <3>; +}; diff --git a/bindings/sound/imx-audio-sgtl5000.txt b/bindings/sound/imx-audio-sgtl5000.txt new file mode 100644 index 00000000..2f89db88 --- /dev/null +++ b/bindings/sound/imx-audio-sgtl5000.txt @@ -0,0 +1,56 @@ +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/bindings/sound/imx-audio-spdif.txt b/bindings/sound/imx-audio-spdif.txt new file mode 100644 index 00000000..da84a442 --- /dev/null +++ b/bindings/sound/imx-audio-spdif.txt @@ -0,0 +1,36 @@ +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 + or jack on the board or it's connecting to some + other IP block, such as an HDMI encoder or + display-controller. + + - spdif-in : This is a boolean property. If present, the receiving + function of S/PDIF will be enabled, indicating there + is 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/bindings/sound/imx-audmux.txt b/bindings/sound/imx-audmux.txt new file mode 100644 index 00000000..2db4dcbe --- /dev/null +++ b/bindings/sound/imx-audmux.txt @@ -0,0 +1,28 @@ +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@21d8000 { + compatible = "fsl,imx6q-audmux", "fsl,imx31-audmux"; + reg = <0x021d8000 0x4000>; +}; diff --git a/bindings/sound/ingenic,jz4740-i2s.txt b/bindings/sound/ingenic,jz4740-i2s.txt new file mode 100644 index 00000000..b623d500 --- /dev/null +++ b/bindings/sound/ingenic,jz4740-i2s.txt @@ -0,0 +1,23 @@ +Ingenic JZ4740 I2S controller + +Required properties: +- compatible : "ingenic,jz4740-i2s" or "ingenic,jz4780-i2s" +- reg : I2S registers location and length +- clocks : AIC and I2S PLL clock specifiers. +- clock-names: "aic" and "i2s" +- dmas: DMA controller phandle and DMA request line for I2S Tx and Rx channels +- dma-names: Must be "tx" and "rx" + +Example: + +i2s: i2s@10020000 { + compatible = "ingenic,jz4740-i2s"; + reg = <0x10020000 0x94>; + + clocks = <&cgu JZ4740_CLK_AIC>, <&cgu JZ4740_CLK_I2SPLL>; + clock-names = "aic", "i2s"; + + dmas = <&dma 2>, <&dma 3>; + dma-names = "tx", "rx"; + +}; diff --git a/bindings/sound/inno-rk3036.txt b/bindings/sound/inno-rk3036.txt new file mode 100644 index 00000000..758de8e2 --- /dev/null +++ b/bindings/sound/inno-rk3036.txt @@ -0,0 +1,20 @@ +Inno audio codec for RK3036 + +Inno audio codec is integrated inside RK3036 SoC. + +Required properties: +- compatible : Should be "rockchip,rk3036-codec". +- reg : The registers of codec. +- clock-names : Should be "acodec_pclk". +- clocks : The clock of codec. +- rockchip,grf : The phandle of grf device node. + +Example: + + acodec: acodec-ana@20030000 { + compatible = "rk3036-codec"; + reg = <0x20030000 0x4000>; + rockchip,grf = <&grf>; + clock-names = "acodec_pclk"; + clocks = <&cru ACLK_VCODEC>; + }; diff --git a/bindings/sound/marvell,pxa2xx-ac97.txt b/bindings/sound/marvell,pxa2xx-ac97.txt new file mode 100644 index 00000000..2ea85d5b --- /dev/null +++ b/bindings/sound/marvell,pxa2xx-ac97.txt @@ -0,0 +1,27 @@ +Marvell PXA2xx audio complex + +This descriptions matches the AC97 controller found in pxa2xx and pxa3xx series. + +Required properties: + - compatible: should be one of the following: + "marvell,pxa250-ac97" + "marvell,pxa270-ac97" + "marvell,pxa300-ac97" + - reg: device MMIO address space + - interrupts: single interrupt generated by AC97 IP + - clocks: input clock of the AC97 IP, refer to clock-bindings.txt + +Optional properties: + - pinctrl-names, pinctrl-0: refer to pinctrl-bindings.txt + - reset-gpios: gpio used for AC97 reset, refer to gpio.txt + +Example: + ac97: sound@40500000 { + compatible = "marvell,pxa250-ac97"; + reg = < 0x40500000 0x1000 >; + interrupts = <14>; + reset-gpios = <&gpio 113 GPIO_ACTIVE_HIGH>; + #sound-dai-cells = <1>; + pinctrl-names = "default"; + pinctrl-0 = < &pmux_ac97_default >; + }; diff --git a/bindings/sound/max98090.txt b/bindings/sound/max98090.txt new file mode 100644 index 00000000..7e1bbd5c --- /dev/null +++ b/bindings/sound/max98090.txt @@ -0,0 +1,59 @@ +MAX98090 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : "maxim,max98090" or "maxim,max98091". + +- reg : The I2C address of the device. + +- interrupts : The CODEC's interrupt output. + +Optional properties: + +- clocks: The phandle of the master clock to the CODEC + +- clock-names: Should be "mclk" + +- #sound-dai-cells : should be 0. + +- maxim,dmic-freq: Frequency at which to clock DMIC + +- maxim,micbias: Micbias voltage applies to the analog mic, valid voltages value are: + 0 - 2.2v + 1 - 2.55v + 2 - 2.4v + 3 - 2.8v + +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/bindings/sound/max98095.txt b/bindings/sound/max98095.txt new file mode 100644 index 00000000..318a4c82 --- /dev/null +++ b/bindings/sound/max98095.txt @@ -0,0 +1,22 @@ +MAX98095 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : "maxim,max98095". + +- reg : The I2C address of the device. + +Optional properties: + +- clocks: The phandle of the master clock to the CODEC + +- clock-names: Should be "mclk" + +Example: + +max98095: codec@11 { + compatible = "maxim,max98095"; + reg = <0x11>; +}; diff --git a/bindings/sound/max98357a.txt b/bindings/sound/max98357a.txt new file mode 100644 index 00000000..28645a2f --- /dev/null +++ b/bindings/sound/max98357a.txt @@ -0,0 +1,18 @@ +Maxim MAX98357A audio DAC + +This node models the Maxim MAX98357A DAC. + +Required properties: +- compatible : "maxim,max98357a" + +Optional properties: +- sdmode-gpios : GPIO specifier for the chip's SD_MODE pin. + If this option is not specified then driver does not manage + the pin state (e.g. chip is always on). + +Example: + +max98357a { + compatible = "maxim,max98357a"; + sdmode-gpios = <&qcom_pinmux 25 0>; +}; diff --git a/bindings/sound/max98371.txt b/bindings/sound/max98371.txt new file mode 100644 index 00000000..8b2b2704 --- /dev/null +++ b/bindings/sound/max98371.txt @@ -0,0 +1,17 @@ +max98371 codec + +This device supports I2C mode only. + +Required properties: + +- compatible : "maxim,max98371" +- reg : The chip select number on the I2C bus + +Example: + +&i2c { + max98371: max98371@31 { + compatible = "maxim,max98371"; + reg = <0x31>; + }; +}; diff --git a/bindings/sound/max98373.txt b/bindings/sound/max98373.txt new file mode 100644 index 00000000..456cb1c5 --- /dev/null +++ b/bindings/sound/max98373.txt @@ -0,0 +1,40 @@ +Maxim Integrated MAX98373 Speaker Amplifier + +This device supports I2C. + +Required properties: + + - compatible : "maxim,max98373" + + - reg : the I2C address of the device. + +Optional properties: + + - maxim,vmon-slot-no : slot number used to send voltage information + or in inteleave mode this will be used as + interleave slot. + slot range : 0 ~ 15, Default : 0 + + - maxim,imon-slot-no : slot number used to send current information + slot range : 0 ~ 15, Default : 0 + + - maxim,spkfb-slot-no : slot number used to send speaker feedback information + slot range : 0 ~ 15, Default : 0 + + - maxim,interleave-mode : For cases where a single combined channel + for the I/V sense data is not sufficient, the device can also be configured + to share a single data output channel on alternating frames. + In this configuration, the current and voltage data will be frame interleaved + on a single output channel. + Boolean, define to enable the interleave mode, Default : false + +Example: + +codec: max98373@31 { + compatible = "maxim,max98373"; + reg = <0x31>; + maxim,vmon-slot-no = <0>; + maxim,imon-slot-no = <1>; + maxim,spkfb-slot-no = <2>; + maxim,interleave-mode; +}; diff --git a/bindings/sound/max98504.txt b/bindings/sound/max98504.txt new file mode 100644 index 00000000..583ed5fd --- /dev/null +++ b/bindings/sound/max98504.txt @@ -0,0 +1,44 @@ +Maxim MAX98504 class D mono speaker amplifier + +This device supports I2C control interface and an IRQ output signal. It features +a PCM and PDM digital audio interface (DAI) and a differential analog input. + +Required properties: + + - compatible : "maxim,max98504" + - reg : should contain the I2C slave device address + - DVDD-supply, DIOVDD-supply, PVDD-supply: power supplies for the device, + as covered in ../regulator/regulator.txt + - interrupts : should specify the interrupt line the device is connected to, + as described in ../interrupt-controller/interrupts.txt + +Optional properties: + + - maxim,brownout-threshold - the PVDD brownout threshold, the value must be + from 0, 1...21 range, corresponding to 2.6V, 2.65V...3.65V voltage range + - maxim,brownout-attenuation - the brownout attenuation to the speaker gain + applied during the "attack hold" and "timed hold" phase, the value must be + from 0...6 (dB) range + - maxim,brownout-attack-hold-ms - the brownout attack hold phase time in ms, + 0...255 (VBATBROWN_ATTK_HOLD, register 0x0018) + - maxim,brownout-timed-hold-ms - the brownout timed hold phase time in ms, + 0...255 (VBATBROWN_TIME_HOLD, register 0x0019) + - maxim,brownout-release-rate-ms - the brownout release phase step time in ms, + 0...255 (VBATBROWN_RELEASE, register 0x001A) + +The default value when the above properties are not specified is 0, +the maxim,brownout-threshold property must be specified to actually enable +the PVDD brownout protection. + +Example: + + max98504@31 { + compatible = "maxim,max98504"; + reg = <0x31>; + interrupt-parent = <&gpio_bank_0>; + interrupts = <2 0>; + + DVDD-supply = <®ulator>; + DIOVDD-supply = <®ulator>; + PVDD-supply = <®ulator>; +}; diff --git a/bindings/sound/max9860.txt b/bindings/sound/max9860.txt new file mode 100644 index 00000000..e0d4e95e --- /dev/null +++ b/bindings/sound/max9860.txt @@ -0,0 +1,28 @@ +MAX9860 Mono Audio Voice Codec + +Required properties: + + - compatible : "maxim,max9860" + + - reg : the I2C address of the device + + - AVDD-supply, DVDD-supply and DVDDIO-supply : power supplies for + the device, as covered in bindings/regulator/regulator.txt + + - clock-names : Required element: "mclk". + + - clocks : A clock specifier for the clock connected as MCLK. + +Examples: + + max9860: max9860@10 { + compatible = "maxim,max9860"; + reg = <0x10>; + + AVDD-supply = <®_1v8>; + DVDD-supply = <®_1v8>; + DVDDIO-supply = <®_3v0>; + + clock-names = "mclk"; + clocks = <&pck2>; + }; diff --git a/bindings/sound/max9867.txt b/bindings/sound/max9867.txt new file mode 100644 index 00000000..b8bd914e --- /dev/null +++ b/bindings/sound/max9867.txt @@ -0,0 +1,17 @@ +max9867 codec + +This device supports I2C mode only. + +Required properties: + +- compatible : "maxim,max9867" +- reg : The chip select number on the I2C bus + +Example: + +&i2c { + max9867: max9867@18 { + compatible = "maxim,max9867"; + reg = <0x18>; + }; +}; diff --git a/bindings/sound/max9892x.txt b/bindings/sound/max9892x.txt new file mode 100644 index 00000000..f6171591 --- /dev/null +++ b/bindings/sound/max9892x.txt @@ -0,0 +1,41 @@ +Maxim Integrated MAX98925/MAX98926/MAX98927 Speaker Amplifier + +This device supports I2C. + +Required properties: + + - compatible : should be one of the following + - "maxim,max98925" + - "maxim,max98926" + - "maxim,max98927" + + - vmon-slot-no : slot number used to send voltage information + or in inteleave mode this will be used as + interleave slot. + MAX98925/MAX98926 slot range : 0 ~ 30, Default : 0 + MAX98927 slot range : 0 ~ 15, Default : 0 + + - imon-slot-no : slot number used to send current information + MAX98925/MAX98926 slot range : 0 ~ 30, Default : 0 + MAX98927 slot range : 0 ~ 15, Default : 0 + + - interleave-mode : When using two MAX9892X in a system it is + possible to create ADC data that that will + overflow the frame size. Digital Audio Interleave + mode provides a means to output VMON and IMON data + from two devices on a single DOUT line when running + smaller frames sizes such as 32 BCLKS per LRCLK or + 48 BCLKS per LRCLK. + Range : 0 (off), 1 (on), Default : 0 + + - reg : the I2C address of the device for I2C + +Example: + +codec: max98927@3a { + compatible = "maxim,max98927"; + vmon-slot-no = <0>; + imon-slot-no = <1>; + interleave-mode = <0>; + reg = <0x3a>; +}; diff --git a/bindings/sound/maxim,max9759.txt b/bindings/sound/maxim,max9759.txt new file mode 100644 index 00000000..737a9963 --- /dev/null +++ b/bindings/sound/maxim,max9759.txt @@ -0,0 +1,18 @@ +Maxim MAX9759 Speaker Amplifier +=============================== + +Required properties: +- compatible : "maxim,max9759" +- shutdown-gpios : the gpio connected to the shutdown pin +- mute-gpios : the gpio connected to the mute pin +- gain-gpios : the 2 gpios connected to the g1 and g2 pins + +Example: + +max9759: analog-amplifier { + compatible = "maxim,max9759"; + shutdown-gpios = <&gpio3 20 GPIO_ACTIVE_LOW>; + mute-gpios = <&gpio3 19 GPIO_ACTIVE_LOW>; + gain-gpios = <&gpio3 23 GPIO_ACTIVE_LOW>, + <&gpio3 25 GPIO_ACTIVE_LOW>; +}; diff --git a/bindings/sound/mrvl,pxa-ssp.txt b/bindings/sound/mrvl,pxa-ssp.txt new file mode 100644 index 00000000..feef39b4 --- /dev/null +++ b/bindings/sound/mrvl,pxa-ssp.txt @@ -0,0 +1,34 @@ +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 + +Optional properties: + + clock-names + clocks Through "clock-names" and "clocks", external clocks + can be configured. If a clock names "extclk" exists, + it will be set to the mclk rate of the audio stream + and be used as clock provider of the DAI. + +Example: + + /* upstream device */ + + ssp1: ssp@41000000 { + compatible = "mrvl,pxa3xx-ssp"; + reg = <0x41000000 0x40>; + interrupts = <24>; + clock-names = "pxa27x-ssp.0"; + }; + + /* DAI as user */ + + ssp_dai0: ssp_dai@0 { + compatible = "mrvl,pxa-ssp-dai"; + port = <&ssp1>; + #sound-dai-cells = <0>; + }; + diff --git a/bindings/sound/mt2701-afe-pcm.txt b/bindings/sound/mt2701-afe-pcm.txt new file mode 100644 index 00000000..560762e0 --- /dev/null +++ b/bindings/sound/mt2701-afe-pcm.txt @@ -0,0 +1,146 @@ +Mediatek AFE PCM controller for mt2701 + +Required properties: +- compatible: should be one of the followings. + - "mediatek,mt2701-audio" + - "mediatek,mt7622-audio" +- interrupts: should contain AFE and ASYS interrupts +- interrupt-names: should be "afe" and "asys" +- power-domains: should define the power domain +- clocks: Must contain an entry for each entry in clock-names + See ../clocks/clock-bindings.txt for details +- clock-names: should have these clock names: + "infra_sys_audio_clk", + "top_audio_mux1_sel", + "top_audio_mux2_sel", + "top_audio_a1sys_hp", + "top_audio_a2sys_hp", + "i2s0_src_sel", + "i2s1_src_sel", + "i2s2_src_sel", + "i2s3_src_sel", + "i2s0_src_div", + "i2s1_src_div", + "i2s2_src_div", + "i2s3_src_div", + "i2s0_mclk_en", + "i2s1_mclk_en", + "i2s2_mclk_en", + "i2s3_mclk_en", + "i2so0_hop_ck", + "i2so1_hop_ck", + "i2so2_hop_ck", + "i2so3_hop_ck", + "i2si0_hop_ck", + "i2si1_hop_ck", + "i2si2_hop_ck", + "i2si3_hop_ck", + "asrc0_out_ck", + "asrc1_out_ck", + "asrc2_out_ck", + "asrc3_out_ck", + "audio_afe_pd", + "audio_afe_conn_pd", + "audio_a1sys_pd", + "audio_a2sys_pd", + "audio_mrgif_pd"; +- assigned-clocks: list of input clocks and dividers for the audio system. + See ../clocks/clock-bindings.txt for details. +- assigned-clocks-parents: parent of input clocks of assigned clocks. +- assigned-clock-rates: list of clock frequencies of assigned clocks. + +Must be a subnode of MediaTek audsys device tree node. +See ../arm/mediatek/mediatek,audsys.txt for details about the parent node. + +Example: + + audsys: audio-subsystem@11220000 { + compatible = "mediatek,mt2701-audsys", "syscon"; + ... + + afe: audio-controller { + compatible = "mediatek,mt2701-audio"; + interrupts = <GIC_SPI 104 IRQ_TYPE_LEVEL_LOW>, + <GIC_SPI 132 IRQ_TYPE_LEVEL_LOW>; + interrupt-names = "afe", "asys"; + power-domains = <&scpsys MT2701_POWER_DOMAIN_IFR_MSC>; + + clocks = <&infracfg CLK_INFRA_AUDIO>, + <&topckgen CLK_TOP_AUD_MUX1_SEL>, + <&topckgen CLK_TOP_AUD_MUX2_SEL>, + <&topckgen CLK_TOP_AUD_48K_TIMING>, + <&topckgen CLK_TOP_AUD_44K_TIMING>, + <&topckgen CLK_TOP_AUD_K1_SRC_SEL>, + <&topckgen CLK_TOP_AUD_K2_SRC_SEL>, + <&topckgen CLK_TOP_AUD_K3_SRC_SEL>, + <&topckgen CLK_TOP_AUD_K4_SRC_SEL>, + <&topckgen CLK_TOP_AUD_K1_SRC_DIV>, + <&topckgen CLK_TOP_AUD_K2_SRC_DIV>, + <&topckgen CLK_TOP_AUD_K3_SRC_DIV>, + <&topckgen CLK_TOP_AUD_K4_SRC_DIV>, + <&topckgen CLK_TOP_AUD_I2S1_MCLK>, + <&topckgen CLK_TOP_AUD_I2S2_MCLK>, + <&topckgen CLK_TOP_AUD_I2S3_MCLK>, + <&topckgen CLK_TOP_AUD_I2S4_MCLK>, + <&audsys CLK_AUD_I2SO1>, + <&audsys CLK_AUD_I2SO2>, + <&audsys CLK_AUD_I2SO3>, + <&audsys CLK_AUD_I2SO4>, + <&audsys CLK_AUD_I2SIN1>, + <&audsys CLK_AUD_I2SIN2>, + <&audsys CLK_AUD_I2SIN3>, + <&audsys CLK_AUD_I2SIN4>, + <&audsys CLK_AUD_ASRCO1>, + <&audsys CLK_AUD_ASRCO2>, + <&audsys CLK_AUD_ASRCO3>, + <&audsys CLK_AUD_ASRCO4>, + <&audsys CLK_AUD_AFE>, + <&audsys CLK_AUD_AFE_CONN>, + <&audsys CLK_AUD_A1SYS>, + <&audsys CLK_AUD_A2SYS>, + <&audsys CLK_AUD_AFE_MRGIF>; + + clock-names = "infra_sys_audio_clk", + "top_audio_mux1_sel", + "top_audio_mux2_sel", + "top_audio_a1sys_hp", + "top_audio_a2sys_hp", + "i2s0_src_sel", + "i2s1_src_sel", + "i2s2_src_sel", + "i2s3_src_sel", + "i2s0_src_div", + "i2s1_src_div", + "i2s2_src_div", + "i2s3_src_div", + "i2s0_mclk_en", + "i2s1_mclk_en", + "i2s2_mclk_en", + "i2s3_mclk_en", + "i2so0_hop_ck", + "i2so1_hop_ck", + "i2so2_hop_ck", + "i2so3_hop_ck", + "i2si0_hop_ck", + "i2si1_hop_ck", + "i2si2_hop_ck", + "i2si3_hop_ck", + "asrc0_out_ck", + "asrc1_out_ck", + "asrc2_out_ck", + "asrc3_out_ck", + "audio_afe_pd", + "audio_afe_conn_pd", + "audio_a1sys_pd", + "audio_a2sys_pd", + "audio_mrgif_pd"; + + assigned-clocks = <&topckgen CLK_TOP_AUD_MUX1_SEL>, + <&topckgen CLK_TOP_AUD_MUX2_SEL>, + <&topckgen CLK_TOP_AUD_MUX1_DIV>, + <&topckgen CLK_TOP_AUD_MUX2_DIV>; + assigned-clock-parents = <&topckgen CLK_TOP_AUD1PLL_98M>, + <&topckgen CLK_TOP_AUD2PLL_90M>; + assigned-clock-rates = <0>, <0>, <49152000>, <45158400>; + }; + }; diff --git a/bindings/sound/mt2701-cs42448.txt b/bindings/sound/mt2701-cs42448.txt new file mode 100644 index 00000000..05574446 --- /dev/null +++ b/bindings/sound/mt2701-cs42448.txt @@ -0,0 +1,43 @@ +MT2701 with CS42448 CODEC + +Required properties: +- compatible: "mediatek,mt2701-cs42448-machine" +- mediatek,platform: the phandle of MT2701 ASoC platform +- audio-routing: a list of the connections between audio +- mediatek,audio-codec: the phandles of cs42448 codec +- mediatek,audio-codec-bt-mrg the phandles of bt-sco dummy codec +- pinctrl-names: Should contain only one value - "default" +- pinctrl-0: Should specify pin control groups used for this controller. +- i2s1-in-sel-gpio1, i2s1-in-sel-gpio2: Should specify two gpio pins to + control I2S1-in mux. + +Example: + + sound:sound { + compatible = "mediatek,mt2701-cs42448-machine"; + mediatek,platform = <&afe>; + /* CS42448 Machine name */ + audio-routing = + "Line Out Jack", "AOUT1L", + "Line Out Jack", "AOUT1R", + "Line Out Jack", "AOUT2L", + "Line Out Jack", "AOUT2R", + "Line Out Jack", "AOUT3L", + "Line Out Jack", "AOUT3R", + "Line Out Jack", "AOUT4L", + "Line Out Jack", "AOUT4R", + "AIN1L", "AMIC", + "AIN1R", "AMIC", + "AIN2L", "Tuner In", + "AIN2R", "Tuner In", + "AIN3L", "Satellite Tuner In", + "AIN3R", "Satellite Tuner In", + "AIN3L", "AUX In", + "AIN3R", "AUX In"; + mediatek,audio-codec = <&cs42448>; + mediatek,audio-codec-bt-mrg = <&bt_sco_codec>; + pinctrl-names = "default"; + pinctrl-0 = <&aud_pins_default>; + i2s1-in-sel-gpio1 = <&pio 53 0>; + i2s1-in-sel-gpio2 = <&pio 54 0>; + }; diff --git a/bindings/sound/mt2701-wm8960.txt b/bindings/sound/mt2701-wm8960.txt new file mode 100644 index 00000000..809b609e --- /dev/null +++ b/bindings/sound/mt2701-wm8960.txt @@ -0,0 +1,24 @@ +MT2701 with WM8960 CODEC + +Required properties: +- compatible: "mediatek,mt2701-wm8960-machine" +- mediatek,platform: the phandle of MT2701 ASoC platform +- audio-routing: a list of the connections between audio +- mediatek,audio-codec: the phandles of wm8960 codec +- pinctrl-names: Should contain only one value - "default" +- pinctrl-0: Should specify pin control groups used for this controller. + +Example: + + sound:sound { + compatible = "mediatek,mt2701-wm8960-machine"; + mediatek,platform = <&afe>; + audio-routing = + "Headphone", "HP_L", + "Headphone", "HP_R", + "LINPUT1", "AMIC", + "RINPUT1", "AMIC"; + mediatek,audio-codec = <&wm8960>; + pinctrl-names = "default"; + pinctrl-0 = <&aud_pins_default>; + }; diff --git a/bindings/sound/mt6351.txt b/bindings/sound/mt6351.txt new file mode 100644 index 00000000..7fb2cb99 --- /dev/null +++ b/bindings/sound/mt6351.txt @@ -0,0 +1,16 @@ +Mediatek MT6351 Audio Codec + +The communication between MT6351 and SoC is through Mediatek PMIC wrapper. +For more detail, please visit Mediatek PMIC wrapper documentation. + +Must be a child node of PMIC wrapper. + +Required properties: + +- compatible : "mediatek,mt6351-sound". + +Example: + +mt6351_snd { + compatible = "mediatek,mt6351-sound"; +}; diff --git a/bindings/sound/mt6797-afe-pcm.txt b/bindings/sound/mt6797-afe-pcm.txt new file mode 100644 index 00000000..0ae29de1 --- /dev/null +++ b/bindings/sound/mt6797-afe-pcm.txt @@ -0,0 +1,42 @@ +Mediatek AFE PCM controller for mt6797 + +Required properties: +- compatible = "mediatek,mt6797-audio"; +- reg: register location and size +- interrupts: should contain AFE interrupt +- power-domains: should define the power domain +- clocks: Must contain an entry for each entry in clock-names +- clock-names: should have these clock names: + "infra_sys_audio_clk", + "infra_sys_audio_26m", + "mtkaif_26m_clk", + "top_mux_audio", + "top_mux_aud_intbus", + "top_sys_pll3_d4", + "top_sys_pll1_d4", + "top_clk26m_clk"; + +Example: + + afe: mt6797-afe-pcm@11220000 { + compatible = "mediatek,mt6797-audio"; + reg = <0 0x11220000 0 0x1000>; + interrupts = <GIC_SPI 151 IRQ_TYPE_LEVEL_LOW>; + power-domains = <&scpsys MT6797_POWER_DOMAIN_AUDIO>; + clocks = <&infrasys CLK_INFRA_AUDIO>, + <&infrasys CLK_INFRA_AUDIO_26M>, + <&infrasys CLK_INFRA_AUDIO_26M_PAD_TOP>, + <&topckgen CLK_TOP_MUX_AUDIO>, + <&topckgen CLK_TOP_MUX_AUD_INTBUS>, + <&topckgen CLK_TOP_SYSPLL3_D4>, + <&topckgen CLK_TOP_SYSPLL1_D4>, + <&clk26m>; + clock-names = "infra_sys_audio_clk", + "infra_sys_audio_26m", + "mtkaif_26m_clk", + "top_mux_audio", + "top_mux_aud_intbus", + "top_sys_pll3_d4", + "top_sys_pll1_d4", + "top_clk26m_clk"; + }; diff --git a/bindings/sound/mt6797-mt6351.txt b/bindings/sound/mt6797-mt6351.txt new file mode 100644 index 00000000..1d95a884 --- /dev/null +++ b/bindings/sound/mt6797-mt6351.txt @@ -0,0 +1,14 @@ +MT6797 with MT6351 CODEC + +Required properties: +- compatible: "mediatek,mt6797-mt6351-sound" +- mediatek,platform: the phandle of MT6797 ASoC platform +- mediatek,audio-codec: the phandles of MT6351 codec + +Example: + + sound { + compatible = "mediatek,mt6797-mt6351-sound"; + mediatek,audio-codec = <&mt6351_snd>; + mediatek,platform = <&afe>; + }; diff --git a/bindings/sound/mt8173-max98090.txt b/bindings/sound/mt8173-max98090.txt new file mode 100644 index 00000000..519e97c8 --- /dev/null +++ b/bindings/sound/mt8173-max98090.txt @@ -0,0 +1,15 @@ +MT8173 with MAX98090 CODEC + +Required properties: +- compatible : "mediatek,mt8173-max98090" +- mediatek,audio-codec: the phandle of the MAX98090 audio codec +- mediatek,platform: the phandle of MT8173 ASoC platform + +Example: + + sound { + compatible = "mediatek,mt8173-max98090"; + mediatek,audio-codec = <&max98090>; + mediatek,platform = <&afe>; + }; + diff --git a/bindings/sound/mt8173-rt5650-rt5514.txt b/bindings/sound/mt8173-rt5650-rt5514.txt new file mode 100644 index 00000000..e8b3c80c --- /dev/null +++ b/bindings/sound/mt8173-rt5650-rt5514.txt @@ -0,0 +1,15 @@ +MT8173 with RT5650 RT5514 CODECS + +Required properties: +- compatible : "mediatek,mt8173-rt5650-rt5514" +- mediatek,audio-codec: the phandles of rt5650 and rt5514 codecs +- mediatek,platform: the phandle of MT8173 ASoC platform + +Example: + + sound { + compatible = "mediatek,mt8173-rt5650-rt5514"; + mediatek,audio-codec = <&rt5650 &rt5514>; + mediatek,platform = <&afe>; + }; + diff --git a/bindings/sound/mt8173-rt5650-rt5676.txt b/bindings/sound/mt8173-rt5650-rt5676.txt new file mode 100644 index 00000000..ac28cdb4 --- /dev/null +++ b/bindings/sound/mt8173-rt5650-rt5676.txt @@ -0,0 +1,16 @@ +MT8173 with RT5650 RT5676 CODECS and HDMI via I2S + +Required properties: +- compatible : "mediatek,mt8173-rt5650-rt5676" +- mediatek,audio-codec: the phandles of rt5650 and rt5676 codecs + and of the hdmi encoder node +- mediatek,platform: the phandle of MT8173 ASoC platform + +Example: + + sound { + compatible = "mediatek,mt8173-rt5650-rt5676"; + mediatek,audio-codec = <&rt5650 &rt5676 &hdmi0>; + mediatek,platform = <&afe>; + }; + diff --git a/bindings/sound/mt8173-rt5650.txt b/bindings/sound/mt8173-rt5650.txt new file mode 100644 index 00000000..29dce2ac --- /dev/null +++ b/bindings/sound/mt8173-rt5650.txt @@ -0,0 +1,31 @@ +MT8173 with RT5650 CODECS and HDMI via I2S + +Required properties: +- compatible : "mediatek,mt8173-rt5650" +- mediatek,audio-codec: the phandles of rt5650 codecs + and of the hdmi encoder node +- mediatek,platform: the phandle of MT8173 ASoC platform + +Optional subnodes: +- codec-capture : the subnode of rt5650 codec capture +Required codec-capture subnode properties: +- sound-dai: audio codec dai name on capture path + <&rt5650 0> : Default setting. Connect rt5650 I2S1 for capture. (dai_name = rt5645-aif1) + <&rt5650 1> : Connect rt5650 I2S2 for capture. (dai_name = rt5645-aif2) + +- mediatek,mclk: the MCLK source + 0 : external oscillator, MCLK = 12.288M + 1 : internal source from mt8173, MCLK = sampling rate*256 + +Example: + + sound { + compatible = "mediatek,mt8173-rt5650"; + mediatek,audio-codec = <&rt5650 &hdmi0>; + mediatek,platform = <&afe>; + mediatek,mclk = <0>; + codec-capture { + sound-dai = <&rt5650 1>; + }; + }; + diff --git a/bindings/sound/mtk-afe-pcm.txt b/bindings/sound/mtk-afe-pcm.txt new file mode 100644 index 00000000..e302c7f4 --- /dev/null +++ b/bindings/sound/mtk-afe-pcm.txt @@ -0,0 +1,45 @@ +Mediatek AFE PCM controller + +Required properties: +- compatible = "mediatek,mt8173-afe-pcm"; +- reg: register location and size +- interrupts: Should contain AFE interrupt +- clock-names: should have these clock names: + "infra_sys_audio_clk", + "top_pdn_audio", + "top_pdn_aud_intbus", + "bck0", + "bck1", + "i2s0_m", + "i2s1_m", + "i2s2_m", + "i2s3_m", + "i2s3_b"; + +Example: + + afe: mt8173-afe-pcm@11220000 { + compatible = "mediatek,mt8173-afe-pcm"; + reg = <0 0x11220000 0 0x1000>; + interrupts = <GIC_SPI 134 IRQ_TYPE_EDGE_FALLING>; + clocks = <&infracfg INFRA_AUDIO>, + <&topckgen TOP_AUDIO_SEL>, + <&topckgen TOP_AUD_INTBUS_SEL>, + <&topckgen TOP_APLL1_DIV0>, + <&topckgen TOP_APLL2_DIV0>, + <&topckgen TOP_I2S0_M_CK_SEL>, + <&topckgen TOP_I2S1_M_CK_SEL>, + <&topckgen TOP_I2S2_M_CK_SEL>, + <&topckgen TOP_I2S3_M_CK_SEL>, + <&topckgen TOP_I2S3_B_CK_SEL>; + clock-names = "infra_sys_audio_clk", + "top_pdn_audio", + "top_pdn_aud_intbus", + "bck0", + "bck1", + "i2s0_m", + "i2s1_m", + "i2s2_m", + "i2s3_m", + "i2s3_b"; + }; diff --git a/bindings/sound/mvebu-audio.txt b/bindings/sound/mvebu-audio.txt new file mode 100644 index 00000000..cb8c07c8 --- /dev/null +++ b/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/bindings/sound/mxs-audio-sgtl5000.txt b/bindings/sound/mxs-audio-sgtl5000.txt new file mode 100644 index 00000000..4eb980bd --- /dev/null +++ b/bindings/sound/mxs-audio-sgtl5000.txt @@ -0,0 +1,42 @@ +* 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 +- 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 + +Example: + +sound { + compatible = "fsl,imx28-evk-sgtl5000", + "fsl,mxs-audio-sgtl5000"; + model = "imx28-evk-sgtl5000"; + saif-controllers = <&saif0 &saif1>; + audio-codec = <&sgtl5000>; + audio-routing = + "MIC_IN", "Mic Jack", + "Mic Jack", "Mic Bias", + "Headphone Jack", "HP_OUT"; +}; diff --git a/bindings/sound/mxs-saif.txt b/bindings/sound/mxs-saif.txt new file mode 100644 index 00000000..7ba07a11 --- /dev/null +++ b/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/bindings/sound/name-prefix.txt b/bindings/sound/name-prefix.txt new file mode 100644 index 00000000..64577590 --- /dev/null +++ b/bindings/sound/name-prefix.txt @@ -0,0 +1,24 @@ +Name prefix: + +Card implementing the routing property define the connection between +audio components as list of string pair. Component using the same +sink/source names may use the name prefix property to prepend the +name of their sinks/sources with the provided string. + +Optional name prefix property: +- sound-name-prefix : string using as prefix for the sink/source names of + the component. + +Example: Two instances of the same component. + +amp0: analog-amplifier@0 { + compatible = "simple-audio-amplifier"; + enable-gpios = <&gpio GPIOH_3 0>; + sound-name-prefix = "FRONT"; +}; + +amp1: analog-amplifier@1 { + compatible = "simple-audio-amplifier"; + enable-gpios = <&gpio GPIOH_4 0>; + sound-name-prefix = "BACK"; +}; diff --git a/bindings/sound/nau8540.txt b/bindings/sound/nau8540.txt new file mode 100644 index 00000000..307a7652 --- /dev/null +++ b/bindings/sound/nau8540.txt @@ -0,0 +1,16 @@ +NAU85L40 audio CODEC + +This device supports I2C only. + +Required properties: + + - compatible : "nuvoton,nau8540" + + - reg : the I2C address of the device. + +Example: + +codec: nau8540@1c { + compatible = "nuvoton,nau8540"; + reg = <0x1c>; +}; diff --git a/bindings/sound/nau8810.txt b/bindings/sound/nau8810.txt new file mode 100644 index 00000000..05830e47 --- /dev/null +++ b/bindings/sound/nau8810.txt @@ -0,0 +1,16 @@ +NAU8810 audio CODEC + +This device supports I2C only. + +Required properties: + + - compatible : "nuvoton,nau8810" + + - reg : the I2C address of the device. + +Example: + +codec: nau8810@1a { + compatible = "nuvoton,nau8810"; + reg = <0x1a>; +}; diff --git a/bindings/sound/nau8824.txt b/bindings/sound/nau8824.txt new file mode 100644 index 00000000..e0058b97 --- /dev/null +++ b/bindings/sound/nau8824.txt @@ -0,0 +1,88 @@ +Nuvoton NAU8824 audio codec + +This device supports I2C only. + +Required properties: + - compatible : Must be "nuvoton,nau8824" + + - reg : the I2C address of the device. This is either 0x1a (CSB=0) or 0x1b (CSB=1). + +Optional properties: + - nuvoton,jkdet-polarity: JKDET pin polarity. 0 - active high, 1 - active low. + + - nuvoton,vref-impedance: VREF Impedance selection + 0 - Open + 1 - 25 kOhm + 2 - 125 kOhm + 3 - 2.5 kOhm + + - nuvoton,micbias-voltage: Micbias voltage level. + 0 - VDDA + 1 - VDDA + 2 - VDDA * 1.1 + 3 - VDDA * 1.2 + 4 - VDDA * 1.3 + 5 - VDDA * 1.4 + 6 - VDDA * 1.53 + 7 - VDDA * 1.53 + + - nuvoton,sar-threshold-num: Number of buttons supported + - nuvoton,sar-threshold: Impedance threshold for each button. Array that contains up to 8 buttons configuration. SAR value is calculated as + SAR = 255 * MICBIAS / SAR_VOLTAGE * R / (2000 + R) + where MICBIAS is configured by 'nuvoton,micbias-voltage', SAR_VOLTAGE is configured by 'nuvoton,sar-voltage', R - button impedance. + Refer datasheet section 10.2 for more information about threshold calculation. + + - nuvoton,sar-hysteresis: Button impedance measurement hysteresis. + + - nuvoton,sar-voltage: Reference voltage for button impedance measurement. + 0 - VDDA + 1 - VDDA + 2 - VDDA * 1.1 + 3 - VDDA * 1.2 + 4 - VDDA * 1.3 + 5 - VDDA * 1.4 + 6 - VDDA * 1.53 + 7 - VDDA * 1.53 + + - nuvoton,sar-compare-time: SAR compare time + 0 - 500 ns + 1 - 1 us + 2 - 2 us + 3 - 4 us + + - nuvoton,sar-sampling-time: SAR sampling time + 0 - 2 us + 1 - 4 us + 2 - 8 us + 3 - 16 us + + - nuvoton,short-key-debounce: Button short key press debounce time. + 0 - 30 ms + 1 - 50 ms + 2 - 100 ms + + - nuvoton,jack-eject-debounce: Jack ejection debounce time. + 0 - 0 ms + 1 - 1 ms + 2 - 10 ms + + +Example: + + headset: nau8824@1a { + compatible = "nuvoton,nau8824"; + reg = <0x1a>; + interrupt-parent = <&gpio>; + interrupts = <TEGRA_GPIO(E, 6) IRQ_TYPE_LEVEL_LOW>; + nuvoton,vref-impedance = <2>; + nuvoton,micbias-voltage = <6>; + // Setup 4 buttons impedance according to Android specification + nuvoton,sar-threshold-num = <4>; + nuvoton,sar-threshold = <0xc 0x1e 0x38 0x60>; + nuvoton,sar-hysteresis = <0>; + nuvoton,sar-voltage = <6>; + nuvoton,sar-compare-time = <1>; + nuvoton,sar-sampling-time = <1>; + nuvoton,short-key-debounce = <0>; + nuvoton,jack-eject-debounce = <1>; + }; diff --git a/bindings/sound/nau8825.txt b/bindings/sound/nau8825.txt new file mode 100644 index 00000000..d16d9683 --- /dev/null +++ b/bindings/sound/nau8825.txt @@ -0,0 +1,105 @@ +Nuvoton NAU8825 audio codec + +This device supports I2C only. + +Required properties: + - compatible : Must be "nuvoton,nau8825" + + - reg : the I2C address of the device. This is either 0x1a (CSB=0) or 0x1b (CSB=1). + +Optional properties: + - nuvoton,jkdet-enable: Enable jack detection via JKDET pin. + - nuvoton,jkdet-pull-enable: Enable JKDET pin pull. If set - pin pull enabled, + otherwise pin in high impedance state. + - nuvoton,jkdet-pull-up: Pull-up JKDET pin. If set then JKDET pin is pull up, otherwise pull down. + - nuvoton,jkdet-polarity: JKDET pin polarity. 0 - active high, 1 - active low. + + - nuvoton,vref-impedance: VREF Impedance selection + 0 - Open + 1 - 25 kOhm + 2 - 125 kOhm + 3 - 2.5 kOhm + + - nuvoton,micbias-voltage: Micbias voltage level. + 0 - VDDA + 1 - VDDA + 2 - VDDA * 1.1 + 3 - VDDA * 1.2 + 4 - VDDA * 1.3 + 5 - VDDA * 1.4 + 6 - VDDA * 1.53 + 7 - VDDA * 1.53 + + - nuvoton,sar-threshold-num: Number of buttons supported + - nuvoton,sar-threshold: Impedance threshold for each button. Array that contains up to 8 buttons configuration. SAR value is calculated as + SAR = 255 * MICBIAS / SAR_VOLTAGE * R / (2000 + R) + where MICBIAS is configured by 'nuvoton,micbias-voltage', SAR_VOLTAGE is configured by 'nuvoton,sar-voltage', R - button impedance. + Refer datasheet section 10.2 for more information about threshold calculation. + + - nuvoton,sar-hysteresis: Button impedance measurement hysteresis. + + - nuvoton,sar-voltage: Reference voltage for button impedance measurement. + 0 - VDDA + 1 - VDDA + 2 - VDDA * 1.1 + 3 - VDDA * 1.2 + 4 - VDDA * 1.3 + 5 - VDDA * 1.4 + 6 - VDDA * 1.53 + 7 - VDDA * 1.53 + + - nuvoton,sar-compare-time: SAR compare time + 0 - 500 ns + 1 - 1 us + 2 - 2 us + 3 - 4 us + + - nuvoton,sar-sampling-time: SAR sampling time + 0 - 2 us + 1 - 4 us + 2 - 8 us + 3 - 16 us + + - nuvoton,short-key-debounce: Button short key press debounce time. + 0 - 30 ms + 1 - 50 ms + 2 - 100 ms + 3 - 30 ms + + - nuvoton,jack-insert-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms + - nuvoton,jack-eject-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms + + - nuvoton,crosstalk-enable: make crosstalk function enable if set. + + - clocks: list of phandle and clock specifier pairs according to common clock bindings for the + clocks described in clock-names + - clock-names: should include "mclk" for the MCLK master clock + +Example: + + headset: nau8825@1a { + compatible = "nuvoton,nau8825"; + reg = <0x1a>; + interrupt-parent = <&gpio>; + interrupts = <TEGRA_GPIO(E, 6) IRQ_TYPE_LEVEL_LOW>; + nuvoton,jkdet-enable; + nuvoton,jkdet-pull-enable; + nuvoton,jkdet-pull-up; + nuvoton,jkdet-polarity = <GPIO_ACTIVE_LOW>; + nuvoton,vref-impedance = <2>; + nuvoton,micbias-voltage = <6>; + // Setup 4 buttons impedance according to Android specification + nuvoton,sar-threshold-num = <4>; + nuvoton,sar-threshold = <0xc 0x1e 0x38 0x60>; + nuvoton,sar-hysteresis = <1>; + nuvoton,sar-voltage = <0>; + nuvoton,sar-compare-time = <0>; + nuvoton,sar-sampling-time = <0>; + nuvoton,short-key-debounce = <2>; + nuvoton,jack-insert-debounce = <7>; + nuvoton,jack-eject-debounce = <7>; + nuvoton,crosstalk-enable; + + clock-names = "mclk"; + clocks = <&tegra_car TEGRA210_CLK_CLK_OUT_2>; + }; diff --git a/bindings/sound/nokia,rx51.txt b/bindings/sound/nokia,rx51.txt new file mode 100644 index 00000000..72f93d99 --- /dev/null +++ b/bindings/sound/nokia,rx51.txt @@ -0,0 +1,27 @@ +* Nokia N900 audio setup + +Required properties: +- compatible: Should contain "nokia,n900-audio" +- nokia,cpu-dai: phandle for the McBSP node +- nokia,audio-codec: phandles for the main TLV320AIC3X node and the + auxiliary TLV320AIC3X node (in this order) +- nokia,headphone-amplifier: phandle for the TPA6130A2 node +- tvout-selection-gpios: GPIO for tvout selection +- jack-detection-gpios: GPIO for jack detection +- eci-switch-gpios: GPIO for ECI (Enhancement Control Interface) switch +- speaker-amplifier-gpios: GPIO for speaker amplifier + +Example: + +sound { + compatible = "nokia,n900-audio"; + + nokia,cpu-dai = <&mcbsp2>; + nokia,audio-codec = <&tlv320aic3x>, <&tlv320aic3x_aux>; + nokia,headphone-amplifier = <&tpa6130a2>; + + tvout-selection-gpios = <&gpio2 8 GPIO_ACTIVE_HIGH>; /* 40 */ + jack-detection-gpios = <&gpio6 17 GPIO_ACTIVE_HIGH>; /* 177 */ + eci-switch-gpios = <&gpio6 22 GPIO_ACTIVE_HIGH>; /* 182 */ + speaker-amplifier-gpios = <&twl_gpio 7 GPIO_ACTIVE_HIGH>; +}; diff --git a/bindings/sound/nvidia,tegra-audio-alc5632.txt b/bindings/sound/nvidia,tegra-audio-alc5632.txt new file mode 100644 index 00000000..57f40f93 --- /dev/null +++ b/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/bindings/sound/nvidia,tegra-audio-max98090.txt b/bindings/sound/nvidia,tegra-audio-max98090.txt new file mode 100644 index 00000000..c3495beb --- /dev/null +++ b/bindings/sound/nvidia,tegra-audio-max98090.txt @@ -0,0 +1,53 @@ +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 + * Int Mic + +- 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 +- nvidia,mic-det-gpios : The GPIO that detect microphones 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/bindings/sound/nvidia,tegra-audio-rt5640.txt b/bindings/sound/nvidia,tegra-audio-rt5640.txt new file mode 100644 index 00000000..7788808d --- /dev/null +++ b/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/bindings/sound/nvidia,tegra-audio-rt5677.txt b/bindings/sound/nvidia,tegra-audio-rt5677.txt new file mode 100644 index 00000000..a4589cda --- /dev/null +++ b/bindings/sound/nvidia,tegra-audio-rt5677.txt @@ -0,0 +1,67 @@ +NVIDIA Tegra audio complex, with RT5677 CODEC + +Required properties: +- compatible : "nvidia,tegra-audio-rt5677" +- 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 RT5677's pins (as documented in its binding), and the jacks + on the board: + + * Headphone + * Speaker + * Headset Mic + * Internal Mic 1 + * Internal Mic 2 + +- nvidia,i2s-controller : The phandle of the Tegra I2S controller that's + connected to the CODEC. +- nvidia,audio-codec : The phandle of the RT5677 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 +- nvidia,hp-en-gpios : The GPIO that enables headphone amplifier +- nvidia,mic-present-gpios: The GPIO that mic jack is plugged in +- nvidia,dmic-clk-en-gpios : The GPIO that gates DMIC clock signal + +Example: + +sound { + compatible = "nvidia,tegra-audio-rt5677-ryu", + "nvidia,tegra-audio-rt5677"; + nvidia,model = "NVIDIA Tegra Ryu"; + + nvidia,audio-routing = + "Headphone", "LOUT2", + "Headphone", "LOUT1", + "Headset Mic", "MICBIAS1", + "IN1P", "Headset Mic", + "IN1N", "Headset Mic", + "DMIC L1", "Internal Mic 1", + "DMIC R1", "Internal Mic 1", + "DMIC L2", "Internal Mic 2", + "DMIC R2", "Internal Mic 2", + "Speaker", "PDM1L", + "Speaker", "PDM1R"; + + nvidia,i2s-controller = <&tegra_i2s1>; + nvidia,audio-codec = <&rt5677>; + + nvidia,hp-det-gpios = <&gpio TEGRA_GPIO(R, 7) GPIO_ACTIVE_HIGH>; + nvidia,mic-present-gpios = <&gpio TEGRA_GPIO(O, 5) GPIO_ACTIVE_LOW>; + nvidia,hp-en-gpios = <&rt5677 1 GPIO_ACTIVE_HIGH>; + nvidia,dmic-clk-en-gpios = <&rt5677 2 GPIO_ACTIVE_HIGH>; + + 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/bindings/sound/nvidia,tegra-audio-sgtl5000.txt b/bindings/sound/nvidia,tegra-audio-sgtl5000.txt new file mode 100644 index 00000000..5da7da4e --- /dev/null +++ b/bindings/sound/nvidia,tegra-audio-sgtl5000.txt @@ -0,0 +1,42 @@ +NVIDIA Tegra audio complex, with SGTL5000 CODEC + +Required properties: +- compatible : "nvidia,tegra-audio-sgtl5000" +- 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 SGTL5000's pins (as documented in its binding), and the jacks + on the board: + + * Headphone Jack + * Line In Jack + * 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 SGTL5000 audio codec. + +Example: + +sound { + compatible = "toradex,tegra-audio-sgtl5000-apalis_t30", + "nvidia,tegra-audio-sgtl5000"; + nvidia,model = "Toradex Apalis T30"; + nvidia,audio-routing = + "Headphone Jack", "HP_OUT", + "LINE_IN", "Line In Jack", + "MIC_IN", "Mic Jack"; + nvidia,i2s-controller = <&tegra_i2s2>; + nvidia,audio-codec = <&sgtl5000>; + clocks = <&tegra_car TEGRA30_CLK_PLL_A>, + <&tegra_car TEGRA30_CLK_PLL_A_OUT0>, + <&tegra_car TEGRA30_CLK_EXTERN1>; + clock-names = "pll_a", "pll_a_out0", "mclk"; +}; diff --git a/bindings/sound/nvidia,tegra-audio-trimslice.txt b/bindings/sound/nvidia,tegra-audio-trimslice.txt new file mode 100644 index 00000000..ef1fe735 --- /dev/null +++ b/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/bindings/sound/nvidia,tegra-audio-wm8753.txt b/bindings/sound/nvidia,tegra-audio-wm8753.txt new file mode 100644 index 00000000..96f6a57d --- /dev/null +++ b/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/bindings/sound/nvidia,tegra-audio-wm8903.txt b/bindings/sound/nvidia,tegra-audio-wm8903.txt new file mode 100644 index 00000000..b795d282 --- /dev/null +++ b/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/bindings/sound/nvidia,tegra-audio-wm9712.txt b/bindings/sound/nvidia,tegra-audio-wm9712.txt new file mode 100644 index 00000000..436f6cd9 --- /dev/null +++ b/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/bindings/sound/nvidia,tegra20-ac97.txt b/bindings/sound/nvidia,tegra20-ac97.txt new file mode 100644 index 00000000..eaf00102 --- /dev/null +++ b/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/bindings/sound/nvidia,tegra20-das.txt b/bindings/sound/nvidia,tegra20-das.txt new file mode 100644 index 00000000..6de3a7ee --- /dev/null +++ b/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/bindings/sound/nvidia,tegra20-i2s.txt b/bindings/sound/nvidia,tegra20-i2s.txt new file mode 100644 index 00000000..dc30c6bf --- /dev/null +++ b/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/bindings/sound/nvidia,tegra30-ahub.txt b/bindings/sound/nvidia,tegra30-ahub.txt new file mode 100644 index 00000000..0e9a1895 --- /dev/null +++ b/bindings/sound/nvidia,tegra30-ahub.txt @@ -0,0 +1,88 @@ +NVIDIA Tegra30 AHUB (Audio Hub) + +Required properties: +- compatible : For Tegra30, must contain "nvidia,tegra30-ahub". For Tegra114, + must contain "nvidia,tegra114-ahub". For Tegra124, must contain + "nvidia,tegra124-ahub". Otherwise, must contain "nvidia,<chip>-ahub", + plus at least one of the above, where <chip> is tegra132. +- 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/bindings/sound/nvidia,tegra30-hda.txt b/bindings/sound/nvidia,tegra30-hda.txt new file mode 100644 index 00000000..44d27456 --- /dev/null +++ b/bindings/sound/nvidia,tegra30-hda.txt @@ -0,0 +1,30 @@ +NVIDIA Tegra30 HDA controller + +Required properties: +- compatible : For Tegra30, must contain "nvidia,tegra30-hda". Otherwise, + must contain '"nvidia,<chip>-hda", "nvidia,tegra30-hda"', where <chip> is + tegra114, tegra124, or tegra132. +- reg : Should contain the HDA registers location and length. +- interrupts : The interrupt from the HDA controller. +- clocks : Must contain an entry for each required entry in clock-names. + See ../clocks/clock-bindings.txt for details. +- clock-names : Must include the following entries: hda, hda2hdmi, hda2codec_2x +- resets : Must contain an entry for each entry in reset-names. + See ../reset/reset.txt for details. +- reset-names : Must include the following entries: hda, hda2hdmi, hda2codec_2x + +Example: + +hda@70030000 { + compatible = "nvidia,tegra124-hda", "nvidia,tegra30-hda"; + reg = <0x0 0x70030000 0x0 0x10000>; + interrupts = <GIC_SPI 81 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&tegra_car TEGRA124_CLK_HDA>, + <&tegra_car TEGRA124_CLK_HDA2HDMI>, + <&tegra_car TEGRA124_CLK_HDA2CODEC_2X>; + clock-names = "hda", "hda2hdmi", "hda2codec_2x"; + resets = <&tegra_car 125>, /* hda */ + <&tegra_car 128>, /* hda2hdmi */ + <&tegra_car 111>; /* hda2codec_2x */ + reset-names = "hda", "hda2hdmi", "hda2codec_2x"; +}; diff --git a/bindings/sound/nvidia,tegra30-i2s.txt b/bindings/sound/nvidia,tegra30-i2s.txt new file mode 100644 index 00000000..38caa936 --- /dev/null +++ b/bindings/sound/nvidia,tegra30-i2s.txt @@ -0,0 +1,27 @@ +NVIDIA Tegra30 I2S controller + +Required properties: +- compatible : For Tegra30, must contain "nvidia,tegra30-i2s". For Tegra124, + must contain "nvidia,tegra124-i2s". Otherwise, must contain + "nvidia,<chip>-i2s" plus at least one of the above, where <chip> is + tegra114 or tegra132. +- 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/bindings/sound/omap-abe-twl6040.txt b/bindings/sound/omap-abe-twl6040.txt new file mode 100644 index 00000000..462b04e8 --- /dev/null +++ b/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/bindings/sound/omap-dmic.txt b/bindings/sound/omap-dmic.txt new file mode 100644 index 00000000..418e30e7 --- /dev/null +++ b/bindings/sound/omap-dmic.txt @@ -0,0 +1,20 @@ +* 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 +- 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/bindings/sound/omap-mcbsp.txt b/bindings/sound/omap-mcbsp.txt new file mode 100644 index 00000000..ae8bf703 --- /dev/null +++ b/bindings/sound/omap-mcbsp.txt @@ -0,0 +1,36 @@ +* 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 +- 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/bindings/sound/omap-mcpdm.txt b/bindings/sound/omap-mcpdm.txt new file mode 100644 index 00000000..5f4e68ca --- /dev/null +++ b/bindings/sound/omap-mcpdm.txt @@ -0,0 +1,20 @@ +* 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 +- 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/bindings/sound/omap-twl4030.txt b/bindings/sound/omap-twl4030.txt new file mode 100644 index 00000000..f6a715e4 --- /dev/null +++ b/bindings/sound/omap-twl4030.txt @@ -0,0 +1,62 @@ +* 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 + +Optional properties: +- ti,codec: phandle for the twl4030 audio node +- 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>; +}; diff --git a/bindings/sound/pcm1789.txt b/bindings/sound/pcm1789.txt new file mode 100644 index 00000000..3c74ed22 --- /dev/null +++ b/bindings/sound/pcm1789.txt @@ -0,0 +1,22 @@ +Texas Instruments pcm1789 DT bindings + +PCM1789 is a simple audio codec that can be connected via +I2C or SPI. Currently, only I2C bus is supported. + +Required properties: + + - compatible: "ti,pcm1789" + +Required properties on I2C: + + - reg: the I2C address + - reset-gpios: GPIO to control the RESET pin + +Examples: + + audio-codec@4c { + compatible = "ti,pcm1789"; + reg = <0x4c>; + reset-gpios = <&gpio2 14 GPIO_ACTIVE_LOW>; + #sound-dai-cells = <0>; + }; diff --git a/bindings/sound/pcm179x.txt b/bindings/sound/pcm179x.txt new file mode 100644 index 00000000..436c2b24 --- /dev/null +++ b/bindings/sound/pcm179x.txt @@ -0,0 +1,27 @@ +Texas Instruments pcm179x DT bindings + +This driver supports both the I2C and SPI bus. + +Required properties: + + - compatible: "ti,pcm1792a" + +For required properties on SPI, please consult +Documentation/devicetree/bindings/spi/spi-bus.txt + +Required properties on I2C: + + - reg: the I2C address + + +Examples: + + codec_spi: 1792a@0 { + compatible = "ti,pcm1792a"; + spi-max-frequency = <600000>; + }; + + codec_i2c: 1792a@4c { + compatible = "ti,pcm1792a"; + reg = <0x4c>; + }; diff --git a/bindings/sound/pcm186x.txt b/bindings/sound/pcm186x.txt new file mode 100644 index 00000000..1087f485 --- /dev/null +++ b/bindings/sound/pcm186x.txt @@ -0,0 +1,42 @@ +Texas Instruments PCM186x Universal Audio ADC + +These devices support both I2C and SPI (configured with pin strapping +on the board). + +Required properties: + + - compatible : "ti,pcm1862", + "ti,pcm1863", + "ti,pcm1864", + "ti,pcm1865" + + - reg : The I2C address of the device for I2C, the chip select + number for SPI. + + - avdd-supply: Analog core power supply (3.3v) + - dvdd-supply: Digital core power supply + - iovdd-supply: Digital IO power supply + See regulator/regulator.txt for more information + +CODEC input pins: + * VINL1 + * VINR1 + * VINL2 + * VINR2 + * VINL3 + * VINR3 + * VINL4 + * VINR4 + +The pins can be used in referring sound node's audio-routing property. + +Example: + + pcm186x: audio-codec@4a { + compatible = "ti,pcm1865"; + reg = <0x4a>; + + avdd-supply = <®_3v3_analog>; + dvdd-supply = <®_3v3>; + iovdd-supply = <®_1v8>; + }; diff --git a/bindings/sound/pcm5102a.txt b/bindings/sound/pcm5102a.txt new file mode 100644 index 00000000..c63ab0b6 --- /dev/null +++ b/bindings/sound/pcm5102a.txt @@ -0,0 +1,13 @@ +PCM5102a audio CODECs + +These devices does not use I2C or SPI. + +Required properties: + + - compatible : set as "ti,pcm5102a" + +Examples: + + pcm5102a: pcm5102a { + compatible = "ti,pcm5102a"; + }; diff --git a/bindings/sound/pcm512x.txt b/bindings/sound/pcm512x.txt new file mode 100644 index 00000000..3aae3b41 --- /dev/null +++ b/bindings/sound/pcm512x.txt @@ -0,0 +1,52 @@ +PCM512x audio CODECs + +These devices support both I2C and SPI (configured with pin strapping +on the board). + +Required properties: + + - compatible : One of "ti,pcm5121", "ti,pcm5122", "ti,pcm5141" or + "ti,pcm5142" + + - 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. If pll-in + and pll-out are specified in addition to a clock, the device is + configured to accept clock input on a specified gpio pin. + + - pll-in, pll-out : gpio pins used to connect the pll using <1> + through <6>. The device will be configured for clock input on the + given pll-in pin and PLL output on the given pll-out pin. An + external connection from the pll-out pin to the SCLK pin is assumed. + +Examples: + + pcm5122: pcm5122@4c { + compatible = "ti,pcm5122"; + reg = <0x4c>; + + AVDD-supply = <®_3v3_analog>; + DVDD-supply = <®_1v8>; + CPVDD-supply = <®_3v3>; + }; + + + pcm5142: pcm5142@4c { + compatible = "ti,pcm5142"; + reg = <0x4c>; + + AVDD-supply = <®_3v3_analog>; + DVDD-supply = <®_1v8>; + CPVDD-supply = <®_3v3>; + + clocks = <&sck>; + pll-in = <3>; + pll-out = <6>; + }; diff --git a/bindings/sound/qcom,apq8016-sbc.txt b/bindings/sound/qcom,apq8016-sbc.txt new file mode 100644 index 00000000..84b28dbe --- /dev/null +++ b/bindings/sound/qcom,apq8016-sbc.txt @@ -0,0 +1,89 @@ +* Qualcomm Technologies APQ8016 SBC ASoC machine driver + +This node models the Qualcomm Technologies APQ8016 SBC ASoC machine driver + +Required properties: + +- compatible : "qcom,apq8016-sbc-sndcard" + +- pinctrl-N : One property must exist for each entry in + pinctrl-names. See ../pinctrl/pinctrl-bindings.txt + for details of the property values. +- pinctrl-names : Must contain a "default" entry. +- reg : Must contain an address for each entry in reg-names. +- reg-names : A list which must include the following entries: + * "mic-iomux" + * "spkr-iomux" +- qcom,model : Name of the sound card. + +- qcom,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, MicBias + of msm8x16_wcd codec and the jacks on the board: + + Power supplies: + * MIC BIAS External1 + * MIC BIAS External2 + * MIC BIAS Internal1 + * MIC BIAS Internal2 + + Board connectors: + * Headset Mic + * Secondary Mic + * DMIC + * Ext Spk + +Dai-link subnode properties and subnodes: + +Required dai-link subnodes: + +- cpu : CPU sub-node +- codec : CODEC sub-node + +Required CPU/CODEC subnodes properties: + +-link-name : Name of the dai link. +-sound-dai : phandle/s and port of CPU/CODEC + +Example: + +sound: sound { + compatible = "qcom,apq8016-sbc-sndcard"; + reg = <0x07702000 0x4>, <0x07702004 0x4>; + reg-names = "mic-iomux", "spkr-iomux"; + qcom,model = "DB410c"; + + qcom,audio-routing = + "MIC BIAS External1", "Handset Mic", + "MIC BIAS Internal2", "Headset Mic", + "MIC BIAS External1", "Secondary Mic", + "AMIC1", "MIC BIAS External1", + "AMIC2", "MIC BIAS Internal2", + "AMIC3", "MIC BIAS External1", + "DMIC1", "MIC BIAS Internal1", + "MIC BIAS Internal1", "Digital Mic1", + "DMIC2", "MIC BIAS Internal1", + "MIC BIAS Internal1", "Digital Mic2"; + + /* I2S - Internal codec */ + internal-dai-link@0 { + cpu { /* PRIMARY */ + sound-dai = <&lpass MI2S_PRIMARY>; + }; + codec { + sound-dai = <&lpass_codec 0>, <&wcd_codec 0>; + }; + }; + + /* External Primary or External Secondary -ADV7533 HDMI */ + external-dai-link@0 { + link-name = "ADV7533"; + cpu { /* QUAT */ + sound-dai = <&lpass MI2S_QUATERNARY>; + }; + codec { + sound-dai = <&adv_bridge 0>; + }; + }; +}; diff --git a/bindings/sound/qcom,apq8096.txt b/bindings/sound/qcom,apq8096.txt new file mode 100644 index 00000000..c814e867 --- /dev/null +++ b/bindings/sound/qcom,apq8096.txt @@ -0,0 +1,120 @@ +* Qualcomm Technologies APQ8096 ASoC sound card driver + +This binding describes the APQ8096 sound card, which uses qdsp for audio. + +- compatible: + Usage: required + Value type: <stringlist> + Definition: must be "qcom,apq8096-sndcard" + +- audio-routing: + Usage: Optional + Value type: <stringlist> + Definition: A list of the connections between audio components. + Each entry is a pair of strings, the first being the + connection's sink, the second being the connection's + source. Valid names could be power supplies, MicBias + of codec and the jacks on the board: + Valid names include: + + Board Connectors: + "Headphone Left" + "Headphone Right" + "Earphone" + "Line Out1" + "Line Out2" + "Line Out3" + "Line Out4" + "Analog Mic1" + "Analog Mic2" + "Analog Mic3" + "Analog Mic4" + "Analog Mic5" + "Analog Mic6" + "Digital Mic2" + "Digital Mic3" + + Audio pins and MicBias on WCD9335 Codec: + "MIC_BIAS1" + "MIC_BIAS2" + "MIC_BIAS3" + "MIC_BIAS4" + "AMIC1" + "AMIC2" + "AMIC3" + "AMIC4" + "AMIC5" + "AMIC6" + "AMIC6" + "DMIC1" + "DMIC2" + "DMIC3" + +- model: + Usage: required + Value type: <stringlist> + Definition: The user-visible name of this sound card. + += dailinks +Each subnode of sndcard represents either a dailink, and subnodes of each +dailinks would be cpu/codec/platform dais. + +- link-name: + Usage: required + Value type: <string> + Definition: User friendly name for dai link + += CPU, PLATFORM, CODEC dais subnodes +- cpu: + Usage: required + Value type: <subnode> + Definition: cpu dai sub-node + +- codec: + Usage: Optional + Value type: <subnode> + Definition: codec dai sub-node + +- platform: + Usage: Optional + Value type: <subnode> + Definition: platform dai sub-node + +- sound-dai: + Usage: required + Value type: <phandle with arguments> + Definition: dai phandle/s and port of CPU/CODEC/PLATFORM node. + +Obsolete: + qcom,model: String for soundcard name (Use model instead) + qcom,audio-routing: A list of the connections between audio components. + (Use audio-routing instead) + +Example: + +audio { + compatible = "qcom,apq8096-sndcard"; + model = "DB820c"; + + mm1-dai-link { + link-name = "MultiMedia1"; + cpu { + sound-dai = <&q6asmdai MSM_FRONTEND_DAI_MULTIMEDIA1>; + }; + }; + + hdmi-dai-link { + link-name = "HDMI Playback"; + cpu { + sound-dai = <&q6afe HDMI_RX>; + }; + + platform { + sound-dai = <&q6adm>; + }; + + codec { + sound-dai = <&hdmi 0>; + }; + }; +}; diff --git a/bindings/sound/qcom,lpass-cpu.txt b/bindings/sound/qcom,lpass-cpu.txt new file mode 100644 index 00000000..21c64832 --- /dev/null +++ b/bindings/sound/qcom,lpass-cpu.txt @@ -0,0 +1,54 @@ +* Qualcomm Technologies LPASS CPU DAI + +This node models the Qualcomm Technologies Low-Power Audio SubSystem (LPASS). + +Required properties: + +- compatible : "qcom,lpass-cpu" or "qcom,apq8016-lpass-cpu" +- clocks : Must contain an entry for each entry in clock-names. +- clock-names : A list which must include the following entries: + * "ahbix-clk" + * "mi2s-osr-clk" + * "mi2s-bit-clk" + : required clocks for "qcom,lpass-cpu-apq8016" + * "ahbix-clk" + * "mi2s-bit-clk0" + * "mi2s-bit-clk1" + * "mi2s-bit-clk2" + * "mi2s-bit-clk3" + * "pcnoc-mport-clk" + * "pcnoc-sway-clk" + +- interrupts : Must contain an entry for each entry in + interrupt-names. +- interrupt-names : A list which must include the following entries: + * "lpass-irq-lpaif" +- pinctrl-N : One property must exist for each entry in + pinctrl-names. See ../pinctrl/pinctrl-bindings.txt + for details of the property values. +- pinctrl-names : Must contain a "default" entry. +- reg : Must contain an address for each entry in reg-names. +- reg-names : A list which must include the following entries: + * "lpass-lpaif" + + + +Optional properties: + +- qcom,adsp : Phandle for the audio DSP node + +Example: + +lpass@28100000 { + compatible = "qcom,lpass-cpu"; + clocks = <&lcc AHBIX_CLK>, <&lcc MI2S_OSR_CLK>, <&lcc MI2S_BIT_CLK>; + clock-names = "ahbix-clk", "mi2s-osr-clk", "mi2s-bit-clk"; + interrupts = <0 85 1>; + interrupt-names = "lpass-irq-lpaif"; + pinctrl-names = "default", "idle"; + pinctrl-0 = <&mi2s_default>; + pinctrl-1 = <&mi2s_idle>; + reg = <0x28100000 0x10000>; + reg-names = "lpass-lpaif"; + qcom,adsp = <&adsp>; +}; diff --git a/bindings/sound/qcom,msm8916-wcd-analog.txt b/bindings/sound/qcom,msm8916-wcd-analog.txt new file mode 100644 index 00000000..fdcea3d1 --- /dev/null +++ b/bindings/sound/qcom,msm8916-wcd-analog.txt @@ -0,0 +1,100 @@ +msm8916 analog audio CODEC + +Bindings for codec Analog IP which is integrated in pmic pm8916, + +## Bindings for codec core on pmic: + +Required properties + - compatible = "qcom,pm8916-wcd-analog-codec"; + - reg: represents the slave base address provided to the peripheral. + - interrupts: List of interrupts in given SPMI peripheral. + - interrupt-names: Names specified to above list of interrupts in same + order. List of supported interrupt names are: + "cdc_spk_cnp_int" - Speaker click and pop interrupt. + "cdc_spk_clip_int" - Speaker clip interrupt. + "cdc_spk_ocp_int" - Speaker over current protect interrupt. + "mbhc_ins_rem_det1" - jack insert removal detect interrupt 1. + "mbhc_but_rel_det" - button release interrupt. + "mbhc_but_press_det" - button press event + "mbhc_ins_rem_det" - jack insert removal detect interrupt. + "mbhc_switch_int" - multi button headset interrupt. + "cdc_ear_ocp_int" - Earphone over current protect interrupt. + "cdc_hphr_ocp_int" - Headphone R over current protect interrupt. + "cdc_hphl_ocp_det" - Headphone L over current protect interrupt. + "cdc_ear_cnp_int" - earphone cnp interrupt. + "cdc_hphr_cnp_int" - hphr click and pop interrupt. + "cdc_hphl_cnp_int" - hphl click and pop interrupt. + + - clocks: Handle to mclk. + - clock-names: should be "mclk" + - vdd-cdc-io-supply: phandle to VDD_CDC_IO regulator DT node. + - vdd-cdc-tx-rx-cx-supply: phandle to VDD_CDC_TX/RX/CX regulator DT node. + - vdd-micbias-supply: phandle of VDD_MICBIAS supply's regulator DT node. +Optional Properties: + - qcom,mbhc-vthreshold-low: Array of 5 threshold voltages in mV for 5 buttons + detection on headset when the mbhc is powered up + by internal current source, this is a low power. + - qcom,mbhc-vthreshold-high: Array of 5 thresold voltages in mV for 5 buttons + detection on headset when mbhc is powered up + from micbias. +- qcom,micbias-lvl: Voltage (mV) for Mic Bias +- qcom,hphl-jack-type-normally-open: boolean, present if hphl pin on jack is a + NO (Normally Open). If not specified, then + its assumed that hphl pin on jack is NC + (Normally Closed). +- qcom,gnd-jack-type-normally-open: boolean, present if gnd pin on jack is + NO (Normally Open). If not specified, then + its assumed that gnd pin on jack is NC + (Normally Closed). +- qcom,micbias1-ext-cap: boolean, present if micbias1 has external capacitor + connected. +- qcom,micbias2-ext-cap: boolean, present if micbias2 has external capacitor + connected. + +Example: + +spmi_bus { + ... + audio-codec@f000{ + compatible = "qcom,pm8916-wcd-analog-codec"; + reg = <0xf000 0x200>; + reg-names = "pmic-codec-core"; + clocks = <&gcc GCC_CODEC_DIGCODEC_CLK>; + clock-names = "mclk"; + qcom,mbhc-vthreshold-low = <75 150 237 450 500>; + qcom,mbhc-vthreshold-high = <75 150 237 450 500>; + interrupt-parent = <&spmi_bus>; + interrupts = <0x1 0xf0 0x0 IRQ_TYPE_NONE>, + <0x1 0xf0 0x1 IRQ_TYPE_NONE>, + <0x1 0xf0 0x2 IRQ_TYPE_NONE>, + <0x1 0xf0 0x3 IRQ_TYPE_NONE>, + <0x1 0xf0 0x4 IRQ_TYPE_NONE>, + <0x1 0xf0 0x5 IRQ_TYPE_NONE>, + <0x1 0xf0 0x6 IRQ_TYPE_NONE>, + <0x1 0xf0 0x7 IRQ_TYPE_NONE>, + <0x1 0xf1 0x0 IRQ_TYPE_NONE>, + <0x1 0xf1 0x1 IRQ_TYPE_NONE>, + <0x1 0xf1 0x2 IRQ_TYPE_NONE>, + <0x1 0xf1 0x3 IRQ_TYPE_NONE>, + <0x1 0xf1 0x4 IRQ_TYPE_NONE>, + <0x1 0xf1 0x5 IRQ_TYPE_NONE>; + interrupt-names = "cdc_spk_cnp_int", + "cdc_spk_clip_int", + "cdc_spk_ocp_int", + "mbhc_ins_rem_det1", + "mbhc_but_rel_det", + "mbhc_but_press_det", + "mbhc_ins_rem_det", + "mbhc_switch_int", + "cdc_ear_ocp_int", + "cdc_hphr_ocp_int", + "cdc_hphl_ocp_det", + "cdc_ear_cnp_int", + "cdc_hphr_cnp_int", + "cdc_hphl_cnp_int"; + VDD-CDC-IO-supply = <&pm8916_l5>; + VDD-CDC-TX-RX-CX-supply = <&pm8916_l5>; + VDD-MICBIAS-supply = <&pm8916_l13>; + #sound-dai-cells = <1>; + }; +}; diff --git a/bindings/sound/qcom,msm8916-wcd-digital.txt b/bindings/sound/qcom,msm8916-wcd-digital.txt new file mode 100644 index 00000000..1c8e4cb2 --- /dev/null +++ b/bindings/sound/qcom,msm8916-wcd-digital.txt @@ -0,0 +1,20 @@ +msm8916 digital audio CODEC + +## Bindings for codec core in lpass: + +Required properties + - compatible = "qcom,msm8916-wcd-digital-codec"; + - reg: address space for lpass codec. + - clocks: Handle to mclk and ahbclk + - clock-names: should be "mclk", "ahbix-clk". + +Example: + +audio-codec@771c000{ + compatible = "qcom,msm8916-wcd-digital-codec"; + reg = <0x0771c000 0x400>; + clocks = <&gcc GCC_ULTAUDIO_AHBFABRIC_IXFABRIC_CLK>, + <&gcc GCC_CODEC_DIGCODEC_CLK>; + clock-names = "ahbix-clk", "mclk"; + #sound-dai-cells = <1>; +}; diff --git a/bindings/sound/qcom,q6adm.txt b/bindings/sound/qcom,q6adm.txt new file mode 100644 index 00000000..bbae426c --- /dev/null +++ b/bindings/sound/qcom,q6adm.txt @@ -0,0 +1,39 @@ +Qualcomm Audio Device Manager (Q6ADM) binding + +Q6ADM is one of the APR audio service on Q6DSP. +Please refer to qcom,apr.txt for details of the coommon apr service bindings +used by the apr service device. + +- but must contain the following property: + +- compatible: + Usage: required + Value type: <stringlist> + Definition: must be "qcom,q6adm-v<MAJOR-NUMBER>.<MINOR-NUMBER>". + Or "qcom,q6adm" where the version number can be queried + from DSP. + example "qcom,q6adm-v2.0" + + += ADM routing +"routing" subnode of the ADM node represents adm routing specific configuration + +- compatible: + Usage: required + Value type: <stringlist> + Definition: must be "qcom,q6adm-routing". + +- #sound-dai-cells + Usage: required + Value type: <u32> + Definition: Must be 0 + += EXAMPLE +q6adm@8 { + compatible = "qcom,q6adm"; + reg = <APR_SVC_ADM>; + q6routing: routing { + compatible = "qcom,q6adm-routing"; + #sound-dai-cells = <0>; + }; +}; diff --git a/bindings/sound/qcom,q6afe.txt b/bindings/sound/qcom,q6afe.txt new file mode 100644 index 00000000..a8179409 --- /dev/null +++ b/bindings/sound/qcom,q6afe.txt @@ -0,0 +1,178 @@ +Qualcomm Audio Front End (Q6AFE) binding + +AFE is one of the APR audio service on Q6DSP +Please refer to qcom,apr.txt for details of the common apr service bindings +used by all apr services. Must contain the following properties. + +- compatible: + Usage: required + Value type: <stringlist> + Definition: must be "qcom,q6afe-v<MAJOR-NUMBER>.<MINOR-NUMBER>" + Or "qcom,q6afe" where the version number can be queried + from DSP. + example "qcom,q6afe" + += AFE DAIs (Digial Audio Interface) +"dais" subnode of the AFE node. It represents afe dais, each afe dai is a +subnode of "dais" representing board specific dai setup. +"dais" node should have following properties followed by dai children. + +- compatible: + Usage: required + Value type: <stringlist> + Definition: must be "qcom,q6afe-dais" + +- #sound-dai-cells + Usage: required + Value type: <u32> + Definition: Must be 1 + +- #address-cells + Usage: required + Value type: <u32> + Definition: Must be 1 + +- #size-cells + Usage: required + Value type: <u32> + Definition: Must be 0 + +== AFE DAI is subnode of "dais" and represent a dai, it includes board specific +configuration of each dai. Must contain the following properties. + +- reg + Usage: required + Value type: <u32> + Definition: Must be dai id + +- qcom,sd-lines + Usage: required for mi2s interface + Value type: <prop-encoded-array> + Definition: Must be list of serial data lines used by this dai. + should be one or more of the 1-4 sd lines. + + - qcom,tdm-sync-mode: + Usage: required for tdm interface + Value type: <prop-encoded-array> + Definition: Synchronization mode. + 0 - Short sync bit mode + 1 - Long sync mode + 2 - Short sync slot mode + + - qcom,tdm-sync-src: + Usage: required for tdm interface + Value type: <prop-encoded-array> + Definition: Synchronization source. + 0 - External source + 1 - Internal source + + - qcom,tdm-data-out: + Usage: required for tdm interface + Value type: <prop-encoded-array> + Definition: Data out signal to drive with other masters. + 0 - Disable + 1 - Enable + + - qcom,tdm-invert-sync: + Usage: required for tdm interface + Value type: <prop-encoded-array> + Definition: Invert the sync. + 0 - Normal + 1 - Invert + + - qcom,tdm-data-delay: + Usage: required for tdm interface + Value type: <prop-encoded-array> + Definition: Number of bit clock to delay data + with respect to sync edge. + 0 - 0 bit clock cycle + 1 - 1 bit clock cycle + 2 - 2 bit clock cycle + + - qcom,tdm-data-align: + Usage: required for tdm interface + Value type: <prop-encoded-array> + Definition: Indicate how data is packed + within the slot. For example, 32 slot width in case of + sample bit width is 24. + 0 - MSB + 1 - LSB + += EXAMPLE + +q6afe@4 { + compatible = "qcom,q6afe"; + reg = <APR_SVC_AFE>; + + dais { + compatible = "qcom,q6afe-dais"; + #sound-dai-cells = <1>; + #address-cells = <1>; + #size-cells = <0>; + + hdmi@1 { + reg = <1>; + }; + + tdm@24 { + reg = <24>; + qcom,tdm-sync-mode = <1>: + qcom,tdm-sync-src = <1>; + qcom,tdm-data-out = <0>; + qcom,tdm-invert-sync = <1>; + qcom,tdm-data-delay = <1>; + qcom,tdm-data-align = <0>; + + }; + + tdm@25 { + reg = <25>; + qcom,tdm-sync-mode = <1>: + qcom,tdm-sync-src = <1>; + qcom,tdm-data-out = <0>; + qcom,tdm-invert-sync = <1>; + qcom,tdm-data-delay <1>: + qcom,tdm-data-align = <0>; + }; + + prim-mi2s-rx@16 { + reg = <16>; + qcom,sd-lines = <1 3>; + }; + + prim-mi2s-tx@17 { + reg = <17>; + qcom,sd-lines = <2>; + }; + + sec-mi2s-rx@18 { + reg = <18>; + qcom,sd-lines = <1 4>; + }; + + sec-mi2s-tx@19 { + reg = <19>; + qcom,sd-lines = <2>; + }; + + tert-mi2s-rx@20 { + reg = <20>; + qcom,sd-lines = <2 4>; + }; + + tert-mi2s-tx@21 { + reg = <21>; + qcom,sd-lines = <1>; + }; + + quat-mi2s-rx@22 { + reg = <22>; + qcom,sd-lines = <1>; + }; + + quat-mi2s-tx@23 { + reg = <23>; + qcom,sd-lines = <2>; + }; + }; +}; diff --git a/bindings/sound/qcom,q6asm.txt b/bindings/sound/qcom,q6asm.txt new file mode 100644 index 00000000..f9c7bd8c --- /dev/null +++ b/bindings/sound/qcom,q6asm.txt @@ -0,0 +1,39 @@ +Qualcomm Audio Stream Manager (Q6ASM) binding + +Q6ASM is one of the APR audio service on Q6DSP. +Please refer to qcom,apr.txt for details of the common apr service bindings +used by the apr service device. + +- but must contain the following property: + +- compatible: + Usage: required + Value type: <stringlist> + Definition: must be "qcom,q6asm-v<MAJOR-NUMBER>.<MINOR-NUMBER>". + Or "qcom,q6asm" where the version number can be queried + from DSP. + example "qcom,q6asm-v2.0" + += ASM DAIs (Digial Audio Interface) +"dais" subnode of the ASM node represents dai specific configuration + +- compatible: + Usage: required + Value type: <stringlist> + Definition: must be "qcom,q6asm-dais". + +- #sound-dai-cells + Usage: required + Value type: <u32> + Definition: Must be 1 + += EXAMPLE + +q6asm@7 { + compatible = "qcom,q6asm"; + reg = <APR_SVC_ASM>; + q6asmdai: dais { + compatible = "qcom,q6asm-dais"; + #sound-dai-cells = <1>; + }; +}; diff --git a/bindings/sound/qcom,q6core.txt b/bindings/sound/qcom,q6core.txt new file mode 100644 index 00000000..7f36ff8b --- /dev/null +++ b/bindings/sound/qcom,q6core.txt @@ -0,0 +1,21 @@ +Qualcomm ADSP Core service binding + +Q6CORE is one of the APR audio service on Q6DSP. +Please refer to qcom,apr.txt for details of the common apr service bindings +used by the apr service device. + +- but must contain the following property: + +- compatible: + Usage: required + Value type: <stringlist> + Definition: must be "qcom,q6core-v<MAJOR-NUMBER>.<MINOR-NUMBER>". + Or "qcom,q6core" where the version number can be queried + from DSP. + example "qcom,q6core-v2.0" + += EXAMPLE +q6core@3 { + compatible = "qcom,q6core"; + reg = <APR_SVC_ADSP_CORE>; +}; diff --git a/bindings/sound/qcom,sdm845.txt b/bindings/sound/qcom,sdm845.txt new file mode 100644 index 00000000..408c4837 --- /dev/null +++ b/bindings/sound/qcom,sdm845.txt @@ -0,0 +1,80 @@ +* Qualcomm Technologies Inc. SDM845 ASoC sound card driver + +This binding describes the SDM845 sound card, which uses qdsp for audio. + +- compatible: + Usage: required + Value type: <stringlist> + Definition: must be "qcom,sdm845-sndcard" + +- audio-routing: + Usage: Optional + Value type: <stringlist> + Definition: A list of the connections between audio components. + Each entry is a pair of strings, the first being the + connection's sink, the second being the connection's + source. Valid names could be power supplies, MicBias + of codec and the jacks on the board. + +- model: + Usage: required + Value type: <stringlist> + Definition: The user-visible name of this sound card. + += dailinks +Each subnode of sndcard represents either a dailink, and subnodes of each +dailinks would be cpu/codec/platform dais. + +- link-name: + Usage: required + Value type: <string> + Definition: User friendly name for dai link + += CPU, PLATFORM, CODEC dais subnodes +- cpu: + Usage: required + Value type: <subnode> + Definition: cpu dai sub-node + +- codec: + Usage: required + Value type: <subnode> + Definition: codec dai sub-node + +- platform: + Usage: Optional + Value type: <subnode> + Definition: platform dai sub-node + +- sound-dai: + Usage: required + Value type: <phandle> + Definition: dai phandle/s and port of CPU/CODEC/PLATFORM node. + +Example: + +audio { + compatible = "qcom,sdm845-sndcard"; + model = "sdm845-snd-card"; + pinctrl-names = "default", "sleep"; + pinctrl-0 = <&pri_mi2s_active &pri_mi2s_ws_active>; + pinctrl-1 = <&pri_mi2s_sleep &pri_mi2s_ws_sleep>; + + mm1-dai-link { + link-name = "MultiMedia1"; + cpu { + sound-dai = <&q6asmdai MSM_FRONTEND_DAI_MULTIMEDIA1>; + }; + }; + + pri-mi2s-dai-link { + link-name = "PRI MI2S Playback"; + cpu { + sound-dai = <&q6afedai PRIMARY_MI2S_RX>; + }; + + platform { + sound-dai = <&q6routing>; + }; + }; +}; diff --git a/bindings/sound/qcom,wcd9335.txt b/bindings/sound/qcom,wcd9335.txt new file mode 100644 index 00000000..1d8d49e3 --- /dev/null +++ b/bindings/sound/qcom,wcd9335.txt @@ -0,0 +1,123 @@ +QCOM WCD9335 Codec + +Qualcomm WCD9335 Codec is a standalone Hi-Fi audio codec IC, supports +Qualcomm Technologies, Inc. (QTI) multimedia solutions, including +the MSM8996, MSM8976, and MSM8956 chipsets. It has in-built +Soundwire controller, interrupt mux. It supports both I2S/I2C and +SLIMbus audio interfaces. + +Required properties with SLIMbus Interface: + +- compatible: + Usage: required + Value type: <stringlist> + Definition: For SLIMbus interface it should be "slimMID,PID", + textual representation of Manufacturer ID, Product Code, + shall be in lower case hexadecimal with leading zeroes + suppressed. Refer to slimbus/bus.txt for details. + Should be: + "slim217,1a0" for MSM8996 and APQ8096 SoCs with SLIMbus. + +- reg + Usage: required + Value type: <u32 u32> + Definition: Should be ('Device index', 'Instance ID') + +- interrupts + Usage: required + Value type: <prop-encoded-array> + Definition: Interrupts via WCD INTR1 and INTR2 pins + +- interrupt-names: + Usage: required + Value type: <String array> + Definition: Interrupt names of WCD INTR1 and INTR2 + Should be: "intr1", "intr2" + +- reset-gpio: + Usage: required + Value type: <String Array> + Definition: Reset gpio line + +- qcom,ifd: + Usage: required + Value type: <phandle> + Definition: SLIM interface device + +- clocks: + Usage: required + Value type: <prop-encoded-array> + Definition: See clock-bindings.txt section "consumers". List of + three clock specifiers for mclk, mclk2 and slimbus clock. + +- clock-names: + Usage: required + Value type: <string> + Definition: Must contain "mclk", "mclk2" and "slimbus" strings. + +- vdd-buck-supply: + Usage: required + Value type: <phandle> + Definition: Should contain a reference to the 1.8V buck supply + +- vdd-buck-sido-supply: + Usage: required + Value type: <phandle> + Definition: Should contain a reference to the 1.8V SIDO buck supply + +- vdd-rx-supply: + Usage: required + Value type: <phandle> + Definition: Should contain a reference to the 1.8V rx supply + +- vdd-tx-supply: + Usage: required + Value type: <phandle> + Definition: Should contain a reference to the 1.8V tx supply + +- vdd-vbat-supply: + Usage: Optional + Value type: <phandle> + Definition: Should contain a reference to the vbat supply + +- vdd-micbias-supply: + Usage: required + Value type: <phandle> + Definition: Should contain a reference to the micbias supply + +- vdd-io-supply: + Usage: required + Value type: <phandle> + Definition: Should contain a reference to the 1.8V io supply + +- interrupt-controller: + Usage: required + Definition: Indicating that this is a interrupt controller + +- #interrupt-cells: + Usage: required + Value type: <int> + Definition: should be 1 + +#sound-dai-cells + Usage: required + Value type: <u32> + Definition: Must be 1 + +codec@1{ + compatible = "slim217,1a0"; + reg = <1 0>; + interrupts = <&msmgpio 54 IRQ_TYPE_LEVEL_HIGH>; + interrupt-names = "intr2" + reset-gpio = <&msmgpio 64 0>; + qcom,ifd = <&wc9335_ifd>; + clock-names = "mclk", "native"; + clocks = <&rpmcc RPM_SMD_DIV_CLK1>, + <&rpmcc RPM_SMD_BB_CLK1>; + vdd-buck-supply = <&pm8994_s4>; + vdd-rx-supply = <&pm8994_s4>; + vdd-buck-sido-supply = <&pm8994_s4>; + vdd-tx-supply = <&pm8994_s4>; + vdd-io-supply = <&pm8994_s4>; + #sound-dai-cells = <1>; +} diff --git a/bindings/sound/qcom-audio-dev.txt b/bindings/sound/qcom-audio-dev.txt new file mode 100644 index 00000000..2fb4fca9 --- /dev/null +++ b/bindings/sound/qcom-audio-dev.txt @@ -0,0 +1,1965 @@ +Qualcomm Technologies, Inc. Audio devices for ALSA sound SoC + +* msm-pcm + +Required properties: + + - compatible : "qcom,msm-pcm-dsp" + + - qcom,msm-pcm-dsp-id : device node id + +* msm-pcm-low-latency + +Required properties: + + - compatible : "qcom,msm-pcm-dsp" + + - qcom,msm-pcm-dsp-id : device node id + +Optional properties: + + - qcom,msm-pcm-low-latency : Flag indicating whether + the device node is of type low latency. + + - qcom,latency-level : Flag indicating whether the device node + is of type regular low latency or ultra + low latency. + regular : regular low latency stream + ultra : ultra low latency stream + ull-pp : ultra low latency stream with post-processing capability + +* msm-pcm-dsp-noirq + +Required properties: + + - compatible : "qcom,msm-pcm-dsp-noirq"; + +Optional properties: + + - qcom,msm-pcm-low-latency : Flag indicating whether + the device node is of type low latency. + + - qcom,latency-level : Flag indicating whether the device node + is of type low latency or ultra low latency + ultra : ultra low latency stream + ull-pp : ultra low latency stream with post-processing capability +* msm-pcm-routing + +Required properties: + + - compatible : "qcom,msm-pcm-routing" + +* msm-pcm-lpa + +Required properties: + + - compatible : "qcom,msm-pcm-lpa" + +* msm-compr-dsp + +Required properties: + + - compatible : "qcom,msm-compr-dsp" + +* msm-compress-dsp + +Required properties: + + - compatible : "qcom,msm-compress-dsp" + +Optional properties: + - qcom,adsp-version: + This property can be used to specify the ADSP version/name. + Based on ADSP version, we decide if we have to use older + ADSP APIs or newer. Right now we are adding "MDSP 1.2" for + 8909 purpose. If the ADSP version is anything other than this + we use new ADSP APIs. + +* msm-voip-dsp + +Required properties: + + - compatible : "qcom,msm-voip-dsp" + +* msm-pcm-voice + +Required properties: + + - compatible : "qcom,msm-pcm-voice" + - qcom,destroy-cvd : Flag indicating whether to destroy cvd at + the end of call for low memory targets + +* msm-voice-host-pcm + +Required properties: + + - compatible : "qcom,msm-voice-host-pcm" + +* msm-voice-svc + +Required properties: + + - compatible : "qcom,msm-voice-svc" + +* msm-stub-codec + +Required properties: + + - compatible : "qcom,msm-stub-codec" + +* msm-hdmi-dba-codec-rx + +Required properties: + + - compatible : "qcom,msm-hdmi-dba-codec-rx" + - qcom,dba-bridge-chip: String info to indicate which bridge-chip + is used for HDMI using DBA. + +* msm-dai-fe + +Required properties: + + - compatible : "qcom,msm-dai-fe" + +* msm-pcm-afe + +Required properties: + + - compatible : "qcom,msm-pcm-afe" + +* msm-pcm-dtmf + +Required properties: + + - compatible : "qcom,msm-pcm-dtmf" + - qcom,msm-pcm-dtmf : Enable DTMF driver in Audio. DTMF driver is + used for generation and detection of DTMF tones, when user is in + active voice call. APR commands are sent from DTMF driver to ADSP. + +* msm-dai-stub + +[First Level Nodes] + +Required properties: + + - compatible : "msm-dai-stub" + +[Second Level Nodes] + +Required properties: + + - compatible : "qcom,msm-dai-stub-dev" + - qcom,msm-dai-stub-dev-id : Stub dai port ID value is from 0 to 3. + This enables stub CPU dai in Audio. The stub dai is used when + there is no real backend in Audio. + +* msm-dai-q6-spdif + +Optional properties: + + - compatible : "msm-dai-q6-spdif" + +* msm-dai-q6-hdmi + +Required properties: + - compatible : "msm-dai-q6-hdmi" + - qcom,msm-dai-q6-dev-id : The hdmi multi channel port ID. + It is passed onto the dsp from the apps to form an audio + path to the HDMI device. Currently the only supported value + is 8, which indicates the rx path used for audio playback + on HDMI device. + +* msm-lsm-client + +Required properties: + + - compatible : "qcom,msm-lsm-client" + +* msm-pcm-loopback + +Required properties: + + - compatible : "qcom,msm-pcm-loopback" + +Optional properties: + + - qcom,msm-pcm-loopback-low-latency : Flag indicating whether + the device node is of type low latency. + +* msm-transcode-loopback + +Required properties: + + - compatible : "qcom,msm-transcode-loopback" + +* msm-dai-q6 + +[First Level Nodes] + +Required properties: + + - compatible : "msm-dai-q6" + +Optional properties: + + - qcom,ext-spk-amp-supply : External speaker amplifier power supply. + - qcom,ext-spk-amp-gpio : External speaker amplifier enable signal. + +[Second Level Nodes] + +Required properties: + + - compatible : "qcom,msm-dai-q6-dev" + - qcom,msm-dai-q6-dev-id : The slimbus multi channel port ID + Value is from 16384 to 16397. + BT SCO port ID value from 12288 to 12289. + RT Proxy port ID values from 224 to 225 and 240 to + 241. + FM Rx and TX port ID values from 12292 to 12293. + incall record Rx and TX port ID values from 32771 to 32772. + inCall Music Delivery port ID is 32773. + incall Music 2 Delivery port ID is 32770. + +Optional properties: + + - qcom,msm-dai-q6-slim-dev-id : The Slimbus HW device (instance) ID associated + with Slimbus ports. + 0 - Slimbus HW device ID 0 (first instance) + 1 - Slimbus HW device ID 1 (second instance) + +* msm_dai_cdc_dma + +[First Level Nodes] + +Required properties: + + - compatible : "qcom,msm-dai-cdc-dma" + +[Second Level Nodes] + +Required properties: + + - compatible : "qcom,msm-dai-cdc-dma-dev" + - qcom,msm-dai-cdc-dma-dev-id : WSA codec dma port ID + Value is from 45056 to 45061. + VA codec dma port ID Value is from 45089 to 45091. + RX and TX codec dma port ID values from 45120 + to 45135. + +Optional properties: + +- qcom,msm-dai-is-island-supported: Defines whether this dai supported in + island mode or not. + 0 - Unsupported + 1 - Supported + +* msm-auxpcm + +Required properties: + + - compatible : "qcom,msm-auxpcm-dev" + + - qcom,msm-cpudai-auxpcm-mode: mode information. The first value is + for 8khz mode, the second is for + 16khz + 0 - for PCM + + - qcom,msm-cpudai-auxpcm-sync: sync information. The first value is + for 8khz mode, the second is for + 16khz + + - qcom,msm-cpudai-auxpcm-frame: No.of bytes per frame. The first + value is for 8khz mode, the second + is for 16khz + 5 - 256BPF + 4 - 128BPF + + - qcom,msm-cpudai-auxpcm-quant: Type of quantization. The first + value is for 8khz mode, the second + is for 16khz + 2 - Linear quantization + + - qcom,msm-cpudai-auxpcm-num-slots: Number of slots per mode in the + msm-cpudai-auxpcm-slot-mapping + array. + The first value is for 8khz mode, the + second is for 16khz. Max number of + slots supported by DSP is 4, anything + above 4 will be truncated to 4 when + sent to DSP. + + - qcom,msm-cpudai-auxpcm-slot-mapping: Array of slot numbers for multi + slot scenario. The first array + is for 8khz mode, the second is + for 16khz. The size of the array + is determined by the value in + qcom,msm-cpudai-auxpcm-num-slots + + - qcom,msm-cpudai-auxpcm-data: Data field - 0. The first value is + for 8khz mode, the second is for + 16khz + + - qcom,msm-cpudai-auxpcm-pcm-clk-rate: Clock rate for pcm - 2048000. The + first value is for 8khz mode, the + second is for 16KHz mode. When clock + rate is set to zero, then external + clock is assumed. + + - qcom,msm-auxpcm-interface: name of AUXPCM interface "primary" + indicates primary AUXPCM interface + "secondary" indicates secondary + AUXPCM interface +Optional properties: + +- pinctrl-names: Pinctrl state names for each pin + group configuration. +- pinctrl-x: Defines pinctrl state for each pin + group +- qcom,msm-cpudai-afe-clk-ver: Indicates version of AFE clock + interface to be used for enabling + PCM clock. If not defined, selects + default AFE clock interface. +- qcom,msm-dai-is-island-supported: Defines whether this dai supported in + island mode or not. + 0 - Unsupported + 1 - Supported + +* msm-pcm-hostless + +Required properties: + + - compatible : "qcom,msm-pcm-hostless" + +* msm-audio-apr + +Required properties: + + - compatible : "qcom,msm-audio-apr" + This device is added to represent APR module. + + - qcom,subsys-name: This value provides the subsystem name where codec + is present. It can be "apr_modem" or "apr_adsp". This + property enable apr driver to receive subsystem up/down + notification from modem/adsp. + +* msm-ocmem-audio + +Required properties: + + - compatible : "qcom,msm-ocmem-audio" + + - qcom,msm_bus,name: Client name + + - qcom,msm_bus,num_cases: Total number of use cases + + - qcom,msm_bus,active_only: Context flag for requests in active + or dual (active & sleep) contex + + - qcom,msm_bus,num_paths: Total number of master-slave pairs + + - qcom,msm_bus,vectors: Arrays of unsigned integers + representing: + master-id, slave-id, arbitrated + bandwidth, + instantaneous bandwidth +* wcd9xxx_intc + +Required properties: + + - compatible : "qcom,wcd9xxx-irq" + + - interrupt-controller : Mark this device node as an + interrupt controller + + - #interrupt-cells : Should be 1 + + - interrupt-parent : Parent interrupt controller + + - qcom,gpio-connect Gpio that connects to parent + interrupt controller + +* audio-ext-clk-up + +Required properties: + + - compatible : "qcom,audio-ref-clk" + + - qcom,codec-ext-clk-src: Clock source type like PMIC, LPASS + requested to enable reference + or external clock. + +Optional properties: + + - qcom,codec-lpass-ext-clk-freq: Property used to specify frequency. + + - qcom,codec-lpass-clk-id: Property used to specify LPASS clock + ID value. + + - clock-names: Name of the PMIC clock that needs + to be enabled for audio ref clock. + This clock is set as parent. + + - clocks: phandle reference to the parent + clock. + + - qcom,mclk-clk-reg: Indicate the register address for mclk. + + - qcom,use-pinctrl: Indicates pinctrl required or not for this + clock node. + +* audio_slimslave + +Required properties: + + - compatible : "qcom,audio-slimslave" + + - elemental-addr: slimbus slave enumeration address. + +* msm-cpe-lsm + +Required properties: + + - compatible : "qcom,msm-cpe-lsm" + - qcom,msm-cpe-lsm-id : lsm afe port ID. CPE lsm driver uses + this property to find out the input afe port ID. Currently + only supported values are 1 and 3. + +* wcd_us_euro_gpio + +Required properties: + + - compatible : "qcom,msm-cdc-pinctrl" + +Optional properties: + - qcom,lpi-gpios : This boolean property is added if GPIOs are under + LPI TLMM. + +* msm-dai-slim + +Required properties: + + - compatible : "qcom,msm-dai-slim" + + - elemental-addr: slimbus slave enumeration address. + +* wcd_gpio_ctrl + +Required properties: + + - compatible : "qcom,msm-cdc-pinctrl" + + - qcom,cdc-rst-n-gpio : TLMM GPIO number + + - pinctrl-names: Pinctrl state names for each pin + group configuration. + - pinctrl-x: Defines pinctrl state for each pin + group. +* msm_cdc_pinctrl + +Required properties: + + - compatible : "qcom,msm-cdc-pinctrl" + + - pinctrl-names: Pinctrl state names for each pin + group configuration. + - pinctrl-x: Defines pinctrl state for each pin + group. + +* wcd_dsp_glink + +Required properties: + + - compatible : "qcom,wcd-dsp-glink" + - qcom,wdsp-channels: List of wdsp supported channel names. + +* msm_ext_disp_audio_codec_rx + +Required properties: + + - compatible : "qcom,msm-ext-disp-audio-codec-rx" + +Example: + + qcom,msm-pcm { + compatible = "qcom,msm-pcm-dsp"; + qcom,msm-pcm-dsp-id = <0>; + }; + + qcom,msm-pcm-low-latency { + compatible = "qcom,msm-pcm-dsp"; + qcom,msm-pcm-dsp-id = <1>; + qcom,msm-pcm-low-latency; + }; + + qcom,msm-pcm-loopback-low-latency { + compatible = "qcom,msm-pcm-loopback"; + qcom,msm-pcm-loopback-low-latency; + }; + + qcom,msm-pcm-routing { + compatible = "qcom,msm-pcm-routing"; + }; + + qcom,msm-pcm-lpa { + compatible = "qcom,msm-pcm-lpa"; + }; + + qcom,msm-compr-dsp { + compatible = "qcom,msm-compr-dsp"; + }; + + qcom,msm-compress-dsp { + compatible = "qcom,msm-compress-dsp"; + }; + + qcom,msm-voip-dsp { + compatible = "qcom,msm-voip-dsp"; + }; + + qcom,msm-pcm-voice { + compatible = "qcom,msm-pcm-voice"; + qcom,destroy-cvd; + }; + + qcom,msm-voice-host-pcm { + compatible = "qcom,msm-voice-host-pcm"; + }; + + qcom,msm-stub-codec { + compatible = "qcom,msm-stub-codec"; + }; + + qcom,msm-dai-fe { + compatible = "qcom,msm-dai-fe"; + }; + + qcom,msm-pcm-dtmf { + compatible = "qcom,msm-pcm-dtmf"; + }; + + qcom,msm-dai-stub { + compatible = "qcom,msm-dai-stub"; + }; + + qcom,msm-dai-q6-spdif { + compatible = "qcom,msm-dai-q6-spdif"; + }; + + qcom,msm-dai-q6-hdmi { + compatible = "qcom,msm-dai-q6-hdmi"; + qcom,msm-dai-q6-dev-id = <8>; + }; + + dai_dp: qcom,msm-dai-q6-dp { + compatible = "qcom,msm-dai-q6-hdmi"; + qcom,msm-dai-q6-dev-id = <24608>; + }; + + qcom,msm-dai-q6 { + compatible = "qcom,msm-dai-q6"; + qcom,msm-dai-q6-sb-0-rx { + compatible = "qcom,msm-dai-q6-dev"; + qcom,msm-dai-q6-dev-id = <16384>; + qcom,msm-dai-q6-slim-dev-id = <0>; + }; + + qcom,msm-dai-q6-sb-0-tx { + compatible = "qcom,msm-dai-q6-dev"; + qcom,msm-dai-q6-dev-id = <16385>; + }; + + qcom,msm-dai-q6-sb-1-rx { + compatible = "qcom,msm-dai-q6-dev"; + qcom,msm-dai-q6-dev-id = <16386>; + }; + + qcom,msm-dai-q6-sb-1-tx { + compatible = "qcom,msm-dai-q6-dev"; + qcom,msm-dai-q6-dev-id = <16387>; + }; + + qcom,msm-dai-q6-sb-3-rx { + compatible = "qcom,msm-dai-q6-dev"; + qcom,msm-dai-q6-dev-id = <16390>; + }; + + qcom,msm-dai-q6-sb-3-tx { + compatible = "qcom,msm-dai-q6-dev"; + qcom,msm-dai-q6-dev-id = <16391>; + }; + + qcom,msm-dai-q6-sb-4-rx { + compatible = "qcom,msm-dai-q6-dev"; + qcom,msm-dai-q6-dev-id = <16392>; + }; + + qcom,msm-dai-q6-sb-4-tx { + compatible = "qcom,msm-dai-q6-dev"; + qcom,msm-dai-q6-dev-id = <16393>; + }; + + qcom,msm-dai-q6-sb-5-tx { + compatible = "qcom,msm-dai-q6-dev"; + qcom,msm-dai-q6-dev-id = <16395>; + }; + + qcom,msm-dai-q6-sb-6-rx { + compatible = "qcom,msm-dai-q6-dev"; + qcom,msm-dai-q6-dev-id = <16396>; + }; + + qcom,msm-dai-q6-sb-6-tx { + compatible = "qcom,msm-dai-q6-dev"; + qcom,msm-dai-q6-dev-id = <16397>; + }; + + qcom,msm-dai-q6-bt-sco-rx { + compatible = "qcom,msm-dai-q6-dev"; + qcom,msm-dai-q6-dev-id = <12288>; + }; + + qcom,msm-dai-q6-bt-sco-tx { + compatible = "qcom,msm-dai-q6-dev"; + qcom,msm-dai-q6-dev-id = <12289>; + }; + + qcom,msm-dai-q6-int-fm-rx { + compatible = "qcom,msm-dai-q6-dev"; + qcom,msm-dai-q6-dev-id = <12292>; + }; + + qcom,msm-dai-q6-int-fm-tx { + compatible = "qcom,msm-dai-q6-dev"; + qcom,msm-dai-q6-dev-id = <12293>; + }; + + qcom,msm-dai-q6-be-afe-pcm-rx { + compatible = "qcom,msm-dai-q6-dev"; + qcom,msm-dai-q6-dev-id = <224>; + }; + + qcom,msm-dai-q6-be-afe-pcm-tx { + compatible = "qcom,msm-dai-q6-dev"; + qcom,msm-dai-q6-dev-id = <225>; + }; + + qcom,msm-dai-q6-afe-proxy-rx { + compatible = "qcom,msm-dai-q6-dev"; + qcom,msm-dai-q6-dev-id = <241>; + }; + + qcom,msm-dai-q6-afe-proxy-tx { + compatible = "qcom,msm-dai-q6-dev"; + qcom,msm-dai-q6-dev-id = <240>; + }; + + qcom,msm-dai-q6-incall-record-rx { + compatible = "qcom,msm-dai-q6-dev"; + qcom,msm-dai-q6-dev-id = <32771>; + }; + + qcom,msm-dai-q6-incall-record-tx { + compatible = "qcom,msm-dai-q6-dev"; + qcom,msm-dai-q6-dev-id = <32772>; + }; + + qcom,msm-dai-q6-incall-music-rx { + compatible = "qcom,msm-dai-q6-dev"; + qcom,msm-dai-q6-dev-id = <32773>; + }; + + qcom,msm-dai-q6-incall-music-2-rx { + compatible = "qcom,msm-dai-q6-dev"; + qcom,msm-dai-q6-dev-id = <32770>; + }; + }; + + qcom,msm-pri-auxpcm { + qcom,msm-cpudai-auxpcm-mode = <1>, <1>; + qcom,msm-cpudai-auxpcm-sync = <1>, <1>; + qcom,msm-cpudai-auxpcm-frame = <5>, <4>; + qcom,msm-cpudai-auxpcm-quant = <2>, <2>; + qcom,msm-cpudai-auxpcm-num-slots = <4>, <4>; + qcom,msm-cpudai-auxpcm-slot-mapping = <1 0 0 0>, <1 3 0 0>; + qcom,msm-cpudai-auxpcm-data = <0>, <0>; + qcom,msm-cpudai-auxpcm-pcm-clk-rate = <2048000>, <2048000>; + qcom,msm-auxpcm-interface = "primary"; + compatible = "qcom,msm-auxpcm-dev"; + pinctrl-names = "default", "idle"; + pinctrl-0 = <&pri_aux_pcm_active &pri_aux_pcm_din_active>; + pinctrl-1 = <&pri_aux_pcm_sleep &pri_aux_pcm_din_sleep>; + }; + + qcom,msm-pcm-hostless { + compatible = "qcom,msm-pcm-hostless"; + }; + + audio_apr: qcom,msm-audio-apr { + compatible = "qcom,msm-audio-apr"; + qcom,subsys-name = "apr_adsp"; + q6core { + compatible = "qcom,q6core-audio"; + bolero: bolero-cdc { + compatible = "qcom,bolero-codec"; + }; + }; + }; + + qcom,msm-ocmem-audio { + compatible = "qcom,msm-ocmem-audio"; + qcom,msm_bus,name = "audio-ocmem"; + qcom,msm_bus,num_cases = <2>; + qcom,msm_bus,active_only = <0>; + qcom,msm_bus,num_paths = <1>; + qcom,msm_bus,vectors = + <11 604 0 0>, + <11 604 32505856 325058560>; + }; + + wcd9xxx_intc: wcd9xxx-irq { + compatible = "qcom,wcd9xxx-irq"; + interrupt-controller; + #interrupt-cells = <1>; + interrupt-parent = <&msmgpio>; + interrupts = <72 0>; + interrupt-names = "cdc-int"; + }; + + clock_audio: audio_ext_clk { + compatible = "qcom,audio-ref-clk"; + qcom,codec-ext-clk-src = <2>; + qcom,codec-lpass-ext-clk-freq = <19200000>; + qcom,codec-lpass-clk-id = <1>; + clock-names = "osr_clk"; + clocks = <&clock_rpm clk_div_clk1>; + #clock-cells = <1>; + pinctrl-names = "sleep", "active"; + pinctrl-0 = <&spkr_i2s_clk_sleep>; + pinctrl-1 = <&spkr_i2s_clk_active>; + }; + + audio_slimslave { + compatible = "qcom,audio-slimslave"; + elemental-addr = [ff ff ff ff 17 02]; + }; + + msm_dai_slim { + compatible = "qcom,msm_dai_slim"; + elemental-addr = [ff ff ff fe 17 02]; + }; + + wcd_gpio_ctrl { + compatible = "qcom,msm-cdc-pinctrl"; + qcom,cdc-rst-n-gpio = <&tlmm 64 0>; + pinctrl-names = "aud_active", "aud_sleep"; + pinctrl-0 = <&cdc_reset_active>; + pinctrl-1 = <&cdc_reset_sleep>; + }; + + msm_cdc_pinctrl { + compatible = "qcom,msm-cdc-pinctrl"; + pinctrl-names = "aud_active", "aud_sleep"; + pinctrl-0 = <&cdc_reset_active>; + pinctrl-1 = <&cdc_reset_sleep>; + }; + + wcd_dsp_glink { + compatible = "qcom,wcd-dsp-glink"; + }; + + msm_ext_disp_audio_codec_rx { + compatible = "qcom,msm-ext-disp-audio-codec-rx"; + }; + +* msm-dai-mi2s + +[First Level Nodes] + +Required properties: + + - compatible : "msm-dai-mi2s" + + [Second Level Nodes] + +Required properties: + + - compatible : "qcom,msm-dai-q6-mi2s" + - qcom,msm-dai-q6-mi2s-dev-id: MSM or MDM can use Slimbus or I2S interface to + transfer data to (WCD9XXX) codec. + If slimbus interface is used then "msm-dai-q6" + needs to be filled with correct data for + slimbus interface. + The sections "msm-dai-mi2s" is used by MDM or + MSM to use I2S interface with codec. + This section is used by CPU driver in ASOC MSM + to configure MI2S interface. MSM internally + has multiple MI2S namely Primary, Secondary, + Tertiary and Quaternary MI2S. + They are represented with id 0, 1, 2, 3 + respectively. + The field "qcom,msm-dai-q6-mi2s-dev-id" + represents which of the MI2S block is used. + These MI2S are connected to I2S interface. + + - qcom,msm-mi2s-rx-lines: Each MI2S interface in MSM has one or more SD + lines. These lines are used for data transfer + between codec and MSM. + This element in indicates which output RX lines + are used in the MI2S interface. + + - qcom,msm-mi2s-tx-lines: Each MI2S interface in MSM has one or more SD + lines. These lines are used for data transfer + between codec and MSM. + This element in indicates which input TX lines + are used in the MI2S interface. + +Optional properties: + +- pinctrl-names: Pinctrl state names for each pin group + configuration. +- pinctrl-x: Defines pinctrl state for each pin group +- qcom,msm-dai-is-island-supported: Defines whether this dai supported in + island mode or not. + 0 - Unsupported + 1 - Supported + +Example: + +qcom,msm-dai-mi2s { + compatible = "qcom,msm-dai-mi2s"; + qcom,msm-dai-q6-mi2s-prim { + compatible = "qcom,msm-dai-q6-mi2s"; + qcom,msm-dai-q6-mi2s-dev-id = <0>; + qcom,msm-mi2s-rx-lines = <2>; + qcom,msm-mi2s-tx-lines = <1>; + pinctrl-names = "default", "idle"; + pinctrl-0 = <&tert_mi2s_active &tert_mi2s_sd0_active>; + pinctrl-1 = <&tert_mi2s_sleep &tert_mi2s_sd0_sleep>; + }; +}; + +* msm-dai-spdif + +[First Level Nodes] + +Required properties: + + - compatible : "msm-dai-spdif" + + [Second Level Nodes] + +Required properties: + + - compatible : "qcom,msm-dai-q6-spdif" + - qcom,msm-dai-q6-dev-id: The SPDIF port ID + Value is from 20480 to 20483. + +Example: + +qcom,msm-dai-spdif { + compatible = "qcom,msm-dai-spdif"; + qcom,msm-dai-q6-spdif-pri-rx { + compatible = "qcom,msm-dai-q6-spdif"; + qcom,msm-dai-q6-dev-id = <20480>; + }; +}; + +* msm-adsp-loader + +Required properties: + - compatible : "qcom,adsp-loader" + - qcom,adsp-state: + It is possible that some MSM use PIL to load the ADSP image. While + other MSM may use SBL to load the ADSP image at boot. Audio APR needs + state of ADSP to register and enable APR to be used for sending commands + to ADSP. so adsp-state represents the state of ADSP to ADSP loader. + Value of 0 indicates ADSP loader needs to use PIL and value of 2 means + ADSP image is already loaded by SBL. + +Optional properties: + - qcom,proc-img-to-load: + This property can be used to override default ADSP + loading by PIL. Based on string input, different proc is + loaded. Right now we are adding option "modem" + for 8916 purpose. Default image will be "adsp" which + will load LPASS Q6 for other targets as expected. + "adsp" option need not be explicitly mentioned in + DTSI file, as it is default option. + +Example: + +qcom,msm-adsp-loader { + compatible = "qcom,adsp-loader"; + qcom,adsp-state = <2>; + qcom,proc-img-to-load = "modem"; +}; + +* msm-audio-ion + +Required properties: + - compatible : "qcom,msm-audio-ion" + +Optional properties: + - qcom,smmu-version: + version ID to provide info regarding smmu version + used in chipset. If ARM SMMU HW - use id value as 1, + If QSMMU HW - use id value as 2. + + - qcom,smmu-sid-mask: + Mask for the Stream ID part of SMMU SID. + + - qcom,smmu-enabled: + It is possible that some MSM have SMMU in ADSP. While other MSM use + no SMMU. Audio lib introduce wrapper for ION APIs. The wrapper needs + presence of SMMU in ADSP to handle ION APIs differently. + Presence of this property means ADSP has SMMU in it. + - iommus: + A phandle parsed by smmu driver. Number of entries will vary across + targets. + +Example: + + qcom,msm-audio-ion { + compatible = "qcom,msm-audio-ion; + qcom,smmu-enabled; + }; + +* msm-dai-tdm + +[First Level Nodes] + +Required properties: + + - compatible : "qcom,msm-dai-tdm" + - qcom,msm-cpudai-tdm-group-id: ID of the group device. TDM interface + supports up to 8 groups: + Primary RX: 37120 + Primary TX: 37121 + Secondary RX: 37136 + Secondary TX: 37137 + Tertiary RX: 37152 + Tertiary TX: 37153 + Quaternary RX: 37168 + Quaternary TX: 37169 + + - qcom,msm-cpudai-tdm-group-num-ports: Number of ports in + msm-cpudai-tdm-group-port-id array. + Max number of ports supported by DSP is 8. + + - qcom,msm-cpudai-tdm-group-port-id: Array of TDM port IDs of the group. + The size of the array is determined by + the value in msm-cpudai-tdm-group-num-ports. + Each group supports up to 8 ports: + Primary RX: 36864, 36866, 36868, 36870, + 36872, 36874, 36876, 36878 + Primary TX: 36865, 36867, 36869, 36871, + 36873, 36875, 36877, 36879 + Secondary RX: 36880, 36882, 36884, 36886, + 36888, 36890, 36892, 36894 + Secondary TX: 36881, 36883, 36885, 36887, + 36889, 36891, 36893, 36895 + Tertiary RX: 36896, 36898, 36900, 36902, + 36904, 36906, 36908, 36910 + Tertiary TX: 36897, 36899, 36901, 36903, + 36905, 36907, 36909, 36911 + Quaternary RX: 36912, 36914, 36916, 36918, + 36920, 36922, 36924, 36926 + Quaternary TX: 36913, 36915, 36917, 36919, + 36921, 36923, 36925, 36927 + + - qcom,msm-cpudai-tdm-clk-rate: Clock rate for tdm - 12288000. + When clock rate is set to zero, + then external clock is assumed. + + - qcom,msm-cpudai-tdm-clk-internal: Clock Source. + 0 - EBIT clock from clk tree + 1 - IBIT clock from clk tree + + - qcom,msm-cpudai-tdm-sync-mode: Synchronization setting. + 0 - Short sync bit mode + 1 - Long sync mode + 2 - Short sync slot mode + + - qcom,msm-cpudai-tdm-sync-src: Synchronization source. + 0 - External source + 1 - Internal source + + - qcom,msm-cpudai-tdm-data-out: Data out signal to drive with other masters. + 0 - Disable + 1 - Enable + + - qcom,msm-cpudai-tdm-invert-sync: Invert the sync. + 0 - Normal + 1 - Invert + + - qcom,msm-cpudai-tdm-data-delay: Number of bit clock to delay data + with respect to sync edge. + 0 - 0 bit clock cycle + 1 - 1 bit clock cycle + 2 - 2 bit clock cycle + + [Second Level Nodes] + +Required properties: + + - compatible : "qcom,msm-dai-q6-tdm" + - qcom,msm-dai-q6-mi2s-dev-id: TDM port ID. + + - qcom,msm-cpudai-tdm-data-align: Indicate how data is packed + within the slot. For example, 32 slot width in case of + sample bit width is 24. + 0 - MSB + 1 - LSB + +Optional properties: + + - qcom,msm-cpudai-tdm-header-start-offset: TDM Custom header start offset + in bytes from this sub-frame. The bytes is counted from 0. + 0 is mapped to the 1st byte in or out of + the digital serial data line this sub-frame belong to. + Supported value: 0, 4, 8. + + - qcom,msm-cpudai-tdm-header-width: Header width per frame followed. + 2 bytes for MOST/TDM case. + Supported value: 2. + + - qcom,msm-cpudai-tdm-header-num-frame-repeat: Number of header followed. + Supported value: 8. + + - pinctrl-names: Pinctrl state names for each pin group + configuration. + + - pinctrl-x: Defines pinctrl state for each pin group. + + - qcom,msm-dai-is-island-supported: Defines whether this dai supported in + island mode or not. + 0 - Unsupported + 1 - Supported + +Example: + + qcom,msm-dai-tdm-quat-rx { + compatible = "qcom,msm-dai-tdm"; + qcom,msm-cpudai-tdm-group-id = <37168>; + qcom,msm-cpudai-tdm-group-num-ports = <1>; + qcom,msm-cpudai-tdm-group-port-id = <36912>; + qcom,msm-cpudai-tdm-clk-rate = <12288000>; + qcom,msm-cpudai-tdm-clk-internal = <1>; + qcom,msm-cpudai-tdm-sync-mode = <0>; + qcom,msm-cpudai-tdm-sync-src = <1>; + qcom,msm-cpudai-tdm-data-out = <0>; + qcom,msm-cpudai-tdm-invert-sync = <0>; + qcom,msm-cpudai-tdm-data-delay = <0>; + pinctrl-names = "default", "sleep"; + pinctrl-0 = <&quat_tdm_active &quat_tdm_dout_active>; + pinctrl-1 = <&quat_tdm_sleep &quat_tdm_dout_sleep>; + dai_quat_tdm_rx_0: qcom,msm-dai-q6-tdm-quat-rx-0 { + compatible = "qcom,msm-dai-q6-tdm"; + qcom,msm-cpudai-tdm-dev-id = <36912>; + qcom,msm-cpudai-tdm-data-align = <0>; + qcom,msm-cpudai-tdm-header-start-offset = <0>; + qcom,msm-cpudai-tdm-header-width = <2>; + qcom,msm-cpudai-tdm-header-num-frame-repeat = <8>; + }; + }; + +* MSMSTUB ASoC Machine driver + +Required properties: +- compatible : "qcom,sm8150-asoc-snd-stub" for SM8150 target. +- compatible : "qcom,kona-asoc-snd-stub" for Kona target. +- qcom,model : The user-visible name of this sound card. +- qcom,tasha-mclk-clk-freq : MCLK frequency value for tasha codec +- asoc-platform: This is phandle list containing the references to platform device + nodes that are used as part of the sound card dai-links. +- asoc-platform-names: This property contains list of platform names. The order of + the platform names should match to that of the phandle order + given in "asoc-platform". +- asoc-cpu: This is phandle list containing the references to cpu dai device nodes + that are used as part of the sound card dai-links. +- asoc-cpu-names: This property contains list of cpu dai names. The order of the + cpu dai names should match to that of the phandle order given + in "asoc-cpu". The cpu names are in the form of "%s.%d" form, + where the id (%d) field represents the back-end AFE port id that + this CPU dai is associated with. +- asoc-codec: This is phandle list containing the references to codec dai device + nodes that are used as part of the sound card dai-links. +- asoc-codec-names: This property contains list of codec dai names. The order of the + codec dai names should match to that of the phandle order given + in "asoc-codec". +Optional properties: +- qcom,wsa-max-devs : Maximum number of WSA881x devices present in the target + +Example: + + sound_stub { + compatible = "qcom,sm8150-asoc-snd-stub"; + qcom,model = "sm8150-stub-snd-card"; + + qcom,tasha-mclk-clk-freq = <9600000>; + asoc-platform = <&pcm0>; + asoc-platform-names = "msm-pcm-dsp.0"; + asoc-cpu = <&sb_0_rx>, <&sb_0_tx>; + asoc-cpu-names = "msm-dai-q6-dev.16384", "msm-dai-q6-dev.16385"; + asoc-codec = <&stub_codec>; + asoc-codec-names = "msm-stub-codec.1"; + qcom,wsa-max-devs = <0>; + }; + +* WCD DSP manager driver + +Required properties: +- compatible : "qcom,wcd-dsp-mgr" +- qcom,wdsp-components : This is phandle list containing the references to the + components of the manager driver. Manager driver will + register to component framework with these phandles. +- qcom,img-filename : String property to provide the dsp image file name that is + to be read from file system and downloaded to dsp memory +Optional properties: +- qcom,wdsp-cmpnt-dev-name : Property that manager driver will parse, but defined + in the child's DT entry that is given to manager driver + with phandle. This property will be used by the manager + driver in case the manager driver cannot match child's + of_node pointer to registered phandle. + +Example: + + qcom,wcd-dsp-mgr { + compatible = "qcom,wcd-dsp-mgr"; + qcom,wdsp-components = <&wcd934x_cdc 0>, + <&wcd_spi_0 1>, + <&glink_spi 2>; + qcom,img-filename = "cpe_9340"; + }; + +Example of child node that would have qcom,wdsp-cmpnt-dev-name property + + wcd934x_cdc: tavil_codec { + qcom,wdsp-cmpnt-dev-name = "tavil_codec"; + }; + +* msm-mdf + +Required properties: + - compatible : "qcom,msm-mdf" + +Optional subnodes: + - qcom,msm_mdf_cb : Child nodes representing the compute context banks. + +Subnode Required properties: + - compatible : "qcom,msm-mdf-cb" + - label: Label describing the subsystem this context bank belongs to. + +Subnode Optional properties: + - qcom,smmu-enabled: + It is possible that some MSM subsystems have SMMU, while other MSM + subsystems do not. MDF platform driver needs to handle SMMU APIs + differently according to the availability of SMMU. + Presence of this property means the subsystem has SMMU in it. + - iommus : A list of phandle and IOMMU specifier pairs that describe the + IOMMU master interfaces of the device. + +Example: + qcom,msm-mdf { + compatible = "qcom,msm-mdf"; + + qcom,msm_mdf_cb1 { + compatible = "qcom,msm-mdf-cb"; + label = "adsp"; + qcom,smmu-enabled; + }; + qcom,msm_mdf_cb2 { + compatible = "qcom,msm-mdf-cb"; + label = "dsps"; + }; + qcom,msm_mdf_cb3 { + compatible = "qcom,msm-mdf-cb"; + label = "modem"; + }; + }; + +* msm-mdf-mem + +Required properties: + - compatible : "qcom,msm-mdf-mem-region" + - qcom,msm-mdf-mem-data-size: indicates the size of memory + for MDF purpose + - memory-region : CMA region which is owned by this device. + +Example: + qcom,msm-mdf-mem { + compatible = "qcom,msm-mdf-mem-region"; + memory-region = <&mdf_mem>; + }; + +* SM8150 ASoC Machine driver + +Required properties: +- compatible : "qcom,sm8150-asoc-snd-pahu-aqt" for pahu codec and + "qcom,sm8150-asoc-snd-tavil" for tavil codec. +- qcom,model : The user-visible name of this sound card. +- qcom,pahu-ext-clk-freq : External CLK frequency value for pahu codec +- qcom,audio-routing : A list of the connections between audio components. +- asoc-platform: This is phandle list containing the references to platform device + nodes that are used as part of the sound card dai-links. +- asoc-platform-names: This property contains list of platform names. The order of + the platform names should match to that of the phandle order + given in "asoc-platform". +- asoc-cpu: This is phandle list containing the references to cpu dai device nodes + that are used as part of the sound card dai-links. +- asoc-cpu-names: This property contains list of cpu dai names. The order of the + cpu dai names should match to that of the phandle order given + in "asoc-cpu". The cpu names are in the form of "%s.%d" form, + where the id (%d) field represents the back-end AFE port id that + this CPU dai is associated with. +- asoc-codec: This is phandle list containing the references to codec dai device + nodes that are used as part of the sound card dai-links. +- asoc-codec-names: This property contains list of codec dai names. The order of the + codec dai names should match to that of the phandle order given + in "asoc-codec". +Optional properties: +- clock-names : clock name defined for external clock. +- clocks : external clock defined for codec clock. +- qcom,wsa-max-devs : Maximum number of WSA881x devices present in the target +- qcom,wsa-devs : List of phandles for all possible WSA881x devices supported for the target +- qcom,wsa-aux-dev-prefix : Name prefix with Left/Right configuration for WSA881x device +- qcom,wcn-btfm : Property to specify if WCN BT/FM chip is used for the target + +Example: + + sound-pahu { + compatible = "qcom,sm8150-asoc-snd-pahu-aqt"; + qcom,model = "sm8150-pahu-aqt-snd-card"; + qcom,ext-disp-audio-rx; + qcom,wcn-btfm; + qcom,mi2s-audio-intf; + qcom,auxpcm-audio-intf; + qcom,msm-mi2s-master = <1>, <1>, <1>, <1>; + + reg = <0x1711a000 0x4>, + <0x1711b000 0x4>, + <0x1711c000 0x4>, + <0x1711d000 0x4>; + reg-names = "lpaif_pri_mode_muxsel", + "lpaif_sec_mode_muxsel", + "lpaif_tert_mode_muxsel", + "lpaif_quat_mode_muxsel"; + + qcom,audio-routing = + "MADINPUT", "MCLK", + "AMIC2", "MIC BIAS2", + "AMIC3", "MIC BIAS2", + "AMIC4", "MIC BIAS2", + "AMIC5", "MIC BIAS3", + "MIC BIAS3", "Handset Mic", + "DMIC0", "MIC BIAS1", + "MIC BIAS1", "Digital Mic0", + "DMIC1", "MIC BIAS1", + "MIC BIAS1", "Digital Mic1", + "DMIC2", "MIC BIAS3", + "MIC BIAS3", "Digital Mic2", + "DMIC3", "MIC BIAS3", + "MIC BIAS3", "Digital Mic3", + "DMIC4", "MIC BIAS4", + "MIC BIAS4", "Digital Mic4", + "DMIC5", "MIC BIAS4", + "MIC BIAS4", "Digital Mic5", + "SpkrLeft IN", "SPK1 OUT", + "SpkrRight IN", "SPK2 OUT"; + + qcom,pahu-ext-clk-freq = <19200000>; + asoc-platform = <&pcm0>, <&pcm1>, <&pcm2>, <&voip>, <&voice>, + <&loopback>, <&compress>, <&hostless>, + <&afe>, <&lsm>, <&routing>, <&cpe>, <&compr>, + <&pcm_noirq>; + asoc-platform-names = "msm-pcm-dsp.0", "msm-pcm-dsp.1", + "msm-pcm-dsp.2", "msm-voip-dsp", + "msm-pcm-voice", "msm-pcm-loopback", + "msm-compress-dsp", "msm-pcm-hostless", + "msm-pcm-afe", "msm-lsm-client", + "msm-pcm-routing", "msm-cpe-lsm", + "msm-compr-dsp", "msm-pcm-dsp-noirq"; + asoc-cpu = <&dai_hdmi>, <&dai_dp>, + <&dai_mi2s0>, <&dai_mi2s1>, + <&dai_mi2s2>, <&dai_mi2s3>, + <&dai_pri_auxpcm>, <&dai_sec_auxpcm>, + <&dai_tert_auxpcm>, <&dai_quat_auxpcm>, + <&sb_0_rx>, <&sb_0_tx>, <&sb_1_rx>, <&sb_1_tx>, + <&sb_2_rx>, <&sb_2_tx>, <&sb_3_rx>, <&sb_3_tx>, + <&sb_4_rx>, <&sb_4_tx>, <&sb_5_tx>, + <&afe_pcm_rx>, <&afe_pcm_tx>, <&afe_proxy_rx>, + <&afe_proxy_tx>, <&incall_record_rx>, + <&incall_record_tx>, <&incall_music_rx>, + <&incall_music_2_rx>, <&sb_5_rx>, <&sb_6_rx>, + <&sb_7_rx>, <&sb_7_tx>, <&sb_8_tx>, + <&usb_audio_rx>, <&usb_audio_tx>, + <&dai_pri_tdm_rx_0>, <&dai_pri_tdm_tx_0>, + <&dai_sec_tdm_rx_0>, <&dai_sec_tdm_tx_0>, + <&dai_tert_tdm_rx_0>, <&dai_tert_tdm_tx_0>, + <&dai_quat_tdm_rx_0>, <&dai_quat_tdm_tx_0>; + asoc-cpu-names = "msm-dai-q6-hdmi.8", "msm-dai-q6-dp.24608", + "msm-dai-q6-mi2s.0", "msm-dai-q6-mi2s.1", + "msm-dai-q6-mi2s.2", "msm-dai-q6-mi2s.3", + "msm-dai-q6-auxpcm.1", "msm-dai-q6-auxpcm.2", + "msm-dai-q6-auxpcm.3", "msm-dai-q6-auxpcm.4", + "msm-dai-q6-dev.16384", "msm-dai-q6-dev.16385", + "msm-dai-q6-dev.16386", "msm-dai-q6-dev.16387", + "msm-dai-q6-dev.16388", "msm-dai-q6-dev.16389", + "msm-dai-q6-dev.16390", "msm-dai-q6-dev.16391", + "msm-dai-q6-dev.16392", "msm-dai-q6-dev.16393", + "msm-dai-q6-dev.16395", "msm-dai-q6-dev.224", + "msm-dai-q6-dev.225", "msm-dai-q6-dev.241", + "msm-dai-q6-dev.240", "msm-dai-q6-dev.32771", + "msm-dai-q6-dev.32772", "msm-dai-q6-dev.32773", + "msm-dai-q6-dev.32770", "msm-dai-q6-dev.16394", + "msm-dai-q6-dev.16396", "msm-dai-q6-dev.16398", + "msm-dai-q6-dev.16399", "msm-dai-q6-dev.16401", + "msm-dai-q6-dev.28672", "msm-dai-q6-dev.28673", + "msm-dai-q6-tdm.36864", "msm-dai-q6-tdm.36865", + "msm-dai-q6-tdm.36880", "msm-dai-q6-tdm.36881", + "msm-dai-q6-tdm.36896", "msm-dai-q6-tdm.36897", + "msm-dai-q6-tdm.36912", "msm-dai-q6-tdm.36913"; + asoc-codec = <&stub_codec>, <&ext_disp_audio_codec>; + asoc-codec-names = "msm-stub-codec.1", + "msm-ext-disp-audio-codec-rx"; + qcom,wsa-max-devs = <2>; + qcom,wsa-devs = <&wsa881x_0211>, <&wsa881x_0212>, + <&wsa881x_0213>, <&wsa881x_0214>; + qcom,wsa-aux-dev-prefix = "SpkrLeft", "SpkrRight", + "SpkrLeft", "SpkrRight"; + }; + + +* QCS405 ASoC Machine driver + +Required properties: +- compatible : "qcom,qcs405-asoc-snd". +- qcom,model : The user-visible name of this sound card. +- qcom,audio-routing : A list of the connections between audio components. +- asoc-platform: This is phandle list containing the references to platform device + nodes that are used as part of the sound card dai-links. +- asoc-platform-names: This property contains list of platform names. The order of + the platform names should match to that of the phandle order + given in "asoc-platform". +- asoc-cpu: This is phandle list containing the references to cpu dai device nodes + that are used as part of the sound card dai-links. +- asoc-cpu-names: This property contains list of cpu dai names. The order of the + cpu dai names should match to that of the phandle order given + in "asoc-cpu". The cpu names are in the form of "%s.%d" form, + where the id (%d) field represents the back-end AFE port id that + this CPU dai is associated with. +- asoc-codec: This is phandle list containing the references to codec dai device + nodes that are used as part of the sound card dai-links. +- asoc-codec-names: This property contains list of codec dai names. The order of the + codec dai names should match to that of the phandle order given + in "asoc-codec". +Optional properties: +- clock-names : clock name defined for external clock. +- clocks : external clock defined for codec clock. +- qcom,wsa-max-devs : Maximum number of WSA881x devices present in the target +- qcom,wsa-devs : List of phandles for all possible WSA881x devices supported for the target +- qcom,wsa-aux-dev-prefix : Name prefix with Left/Right configuration for WSA881x device +- qcom,wcn-btfm : Property to specify if WCN BT/FM chip is used for the target +- qcom,wsa_bolero_codec : Property to specify if WSA macro in Bolero codec is used for this target +- qcom,va_bolero_codec : Property to specify if VA macro in Bolero codec is used for this target +- qcom,tasha_codec : Property to specify if Tasha codec is used for this target +- qcom,cdc-dmic-gpios : phandle for Digital mic clk and data gpios. +- qcom,csra-codec : Property to specify if CSRA66x0 is used for this target +- qcom,csra-max-devs : Maximum number of CSRA66x0 devices present in the target +- qcom,csra-devs : List of phandles of all possible CSRA66x0 devices supported for the target +- qcom,csra-aux-dev-prefix : Name prefix in multi-channel configuration for CSRA66x0 device +Example: + + qcs405_snd { + compatible = "qcom,qcs405-asoc-snd"; + qcom,wsa_bolero_codec = <1>; + qcom,va_bolero_codec = <1>; + qcom,tasha_codec = <1>; + qcom,ext-disp-audio-rx = <1>; + qcom,wcn-btfm = <1>; + qcom,mi2s-audio-intf = <1>; + qcom,auxpcm-audio-intf = <1>; + qcom,msm-mi2s-master = <1>, <1>, <1>, <1>; + + qcom,audio-routing = + "MADINPUT", "MCLK", + "AMIC2", "MIC BIAS2", + "AMIC3", "MIC BIAS2", + "AMIC4", "MIC BIAS2", + "AMIC5", "MIC BIAS3", + "MIC BIAS3", "Handset Mic", + "DMIC0", "MIC BIAS1", + "MIC BIAS1", "Digital Mic0", + "DMIC1", "MIC BIAS1", + "MIC BIAS1", "Digital Mic1", + "DMIC2", "MIC BIAS3", + "MIC BIAS3", "Digital Mic2", + "DMIC3", "MIC BIAS3", + "MIC BIAS3", "Digital Mic3", + "DMIC4", "MIC BIAS4", + "MIC BIAS4", "Digital Mic4", + "DMIC5", "MIC BIAS4", + "MIC BIAS4", "Digital Mic5", + "SpkrLeft IN", "SPK1 OUT", + "SpkrRight IN", "SPK2 OUT"; + + asoc-platform = <&pcm0>, <&pcm1>, <&pcm2>, <&voip>, <&voice>, + <&loopback>, <&compress>, <&hostless>, + <&afe>, <&lsm>, <&routing>, <&cpe>, <&compr>, + <&pcm_noirq>; + asoc-platform-names = "msm-pcm-dsp.0", "msm-pcm-dsp.1", + "msm-pcm-dsp.2", "msm-voip-dsp", + "msm-pcm-voice", "msm-pcm-loopback", + "msm-compress-dsp", "msm-pcm-hostless", + "msm-pcm-afe", "msm-lsm-client", + "msm-pcm-routing", "msm-cpe-lsm", + "msm-compr-dsp", "msm-pcm-dsp-noirq"; + asoc-cpu = <&dai_hdmi>, <&dai_dp>, + <&dai_mi2s0>, <&dai_mi2s1>, + <&dai_mi2s2>, <&dai_mi2s3>, + <&dai_pri_auxpcm>, <&dai_sec_auxpcm>, + <&dai_tert_auxpcm>, <&dai_quat_auxpcm>, + <&sb_0_rx>, <&sb_0_tx>, <&sb_1_rx>, <&sb_1_tx>, + <&sb_2_rx>, <&sb_2_tx>, <&sb_3_rx>, <&sb_3_tx>, + <&sb_4_rx>, <&sb_4_tx>, <&sb_5_tx>, + <&afe_pcm_rx>, <&afe_pcm_tx>, <&afe_proxy_rx>, + <&afe_proxy_tx>, <&incall_record_rx>, + <&incall_record_tx>, <&incall_music_rx>, + <&incall_music_2_rx>, <&sb_5_rx>, <&sb_6_rx>, + <&sb_7_rx>, <&sb_7_tx>, <&sb_8_tx>, + <&usb_audio_rx>, <&usb_audio_tx>, + <&dai_pri_tdm_rx_0>, <&dai_pri_tdm_tx_0>, + <&dai_sec_tdm_rx_0>, <&dai_sec_tdm_tx_0>, + <&dai_tert_tdm_rx_0>, <&dai_tert_tdm_tx_0>, + <&dai_quat_tdm_rx_0>, <&dai_quat_tdm_tx_0>, + <&wsa_cdc_dma_0_rx>, <&wsa_cdc_dma_0_tx>, + <&wsa_cdc_dma_1_rx>, <&wsa_cdc_dma_1_tx>, + <&wsa_cdc_dma_2_tx>, <&va_cdc_dma_0_tx>, + <&va_cdc_dma_1_tx>; + asoc-cpu-names = "msm-dai-q6-hdmi.8", "msm-dai-q6-dp.24608", + "msm-dai-q6-mi2s.0", "msm-dai-q6-mi2s.1", + "msm-dai-q6-mi2s.2", "msm-dai-q6-mi2s.3", + "msm-dai-q6-auxpcm.1", "msm-dai-q6-auxpcm.2", + "msm-dai-q6-auxpcm.3", "msm-dai-q6-auxpcm.4", + "msm-dai-q6-dev.16384", "msm-dai-q6-dev.16385", + "msm-dai-q6-dev.16386", "msm-dai-q6-dev.16387", + "msm-dai-q6-dev.16388", "msm-dai-q6-dev.16389", + "msm-dai-q6-dev.16390", "msm-dai-q6-dev.16391", + "msm-dai-q6-dev.16392", "msm-dai-q6-dev.16393", + "msm-dai-q6-dev.16395", "msm-dai-q6-dev.224", + "msm-dai-q6-dev.225", "msm-dai-q6-dev.241", + "msm-dai-q6-dev.240", "msm-dai-q6-dev.32771", + "msm-dai-q6-dev.32772", "msm-dai-q6-dev.32773", + "msm-dai-q6-dev.32770", "msm-dai-q6-dev.16394", + "msm-dai-q6-dev.16396", "msm-dai-q6-dev.16398", + "msm-dai-q6-dev.16399", "msm-dai-q6-dev.16401", + "msm-dai-q6-dev.28672", "msm-dai-q6-dev.28673", + "msm-dai-q6-tdm.36864", "msm-dai-q6-tdm.36865", + "msm-dai-q6-tdm.36880", "msm-dai-q6-tdm.36881", + "msm-dai-q6-tdm.36896", "msm-dai-q6-tdm.36897", + "msm-dai-q6-tdm.36912", "msm-dai-q6-tdm.36913", + "msm-dai-q6-cdc-dma-dev.45056", + "msm-dai-q6-cdc-dma-dev.45057", + "msm-dai-q6-cdc-dma-dev.45058", + "msm-dai-q6-cdc-dma-dev.45059", + "msm-dai-q6-cdc-dma-dev.45061", + "msm-dai-q6-cdc-dma-dev.45089", + "msm-dai-q6-cdc-dma-dev.45091"; + asoc-codec = <&stub_codec>, <&ext_disp_audio_codec>, + <&bolero>;; + asoc-codec-names = "msm-stub-codec.1", + "msm-ext-disp-audio-codec-rx", + "bolero_codec"; + qcom,wsa-max-devs = <2>; + qcom,wsa-devs = <&wsa881x_0211>, <&wsa881x_0212>, + <&wsa881x_0213>, <&wsa881x_0214>; + qcom,wsa-aux-dev-prefix = "SpkrLeft", "SpkrRight", + "SpkrLeft", "SpkrRight"; + qcom,cdc-dmic-gpios = <&cdc_dmic12_gpios>, <&cdc_dmic34_gpios>, + <&cdc_dmic56_gpios>, <&cdc_dmic78_gpios>; + }; + +* SM6150 ASoC Machine driver + +Required properties: +- compatible : "qcom,sm6150-asoc-snd". +- qcom,model : The user-visible name of this sound card. +- qcom,audio-routing : A list of the connections between audio components. +- asoc-platform: This is phandle list containing the references to platform device + nodes that are used as part of the sound card dai-links. +- asoc-platform-names: This property contains list of platform names. The order of + the platform names should match to that of the phandle order + given in "asoc-platform". +- asoc-cpu: This is phandle list containing the references to cpu dai device nodes + that are used as part of the sound card dai-links. +- asoc-cpu-names: This property contains list of cpu dai names. The order of the + cpu dai names should match to that of the phandle order given + in "asoc-cpu". The cpu names are in the form of "%s.%d" form, + where the id (%d) field represents the back-end AFE port id that + this CPU dai is associated with. +- asoc-codec: This is phandle list containing the references to codec dai device + nodes that are used as part of the sound card dai-links. +- asoc-codec-names: This property contains list of codec dai names. The order of the + codec dai names should match to that of the phandle order given + in "asoc-codec". +- qcom,codec-aux-devs: This is phandle list containing the references to Auxilary + codec devices. + +Optional properties: +- qcom,msm-mi2s-master: This property is used to inform machine driver + if MSM is the clock master of mi2s. 1 means master and 0 means slave. The + first entry is primary mi2s; the second entry is secondary mi2s, and so on. +- qcom,msm-mbhc-hphl-swh: This property is used to distinguish headset HPHL + switch type on target typically the switch type will be normally open or + normally close, value for this property 0 for normally close and 1 for + normally open. +- qcom,msm-mbhc-gnd-swh: This property is used to distinguish headset GND + switch type on target typically the switch type will be normally open or + normally close, value for this property 0 for normally close and 1 for + normally open. +- qcom,wsa-max-devs : Maximum number of WSA881x devices present in the target +- qcom,wsa-devs : List of phandles for all possible WSA881x devices supported for the target +- qcom,wsa-aux-dev-prefix : Name prefix with Left/Right configuration for WSA881x device +- qcom,ext-disp-audio-rx: Property to specify if Audio over Display port is supported for the target +- qcom,wcn-btfm : Property to specify if WCN BT/FM chip is used for the target +- qcom,mi2s-audio-intf: Property to specify if MI2S interface is used for the target +- qcom,auxpcm-audio-intf: Property to specify if Aux PCM interface is used for the target +- qcom,tavil_codec : Property to specify if Tavil codec is used for this target +- qcom,cdc-dmic-gpios : phandle for Digital mic clk and data gpios. +- qcom,msm_audio_ssr_devs: List the snd event framework clients + +Example: + sm6150_snd: sound { + status = "okay"; + compatible = "qcom,sm6150-asoc-snd"; + qcom,ext-disp-audio-rx = <1>; + qcom,wcn-btfm = <1>; + qcom,mi2s-audio-intf = <1>; + qcom,auxpcm-audio-intf = <1>; + + asoc-platform = <&pcm0>, <&pcm1>, <&pcm2>, <&voip>, <&voice>, + <&loopback>, <&compress>, <&hostless>, + <&afe>, <&lsm>, <&routing>, <&compr>, + <&pcm_noirq>; + asoc-platform-names = "msm-pcm-dsp.0", "msm-pcm-dsp.1", + "msm-pcm-dsp.2", "msm-voip-dsp", + "msm-pcm-voice", "msm-pcm-loopback", + "msm-compress-dsp", "msm-pcm-hostless", + "msm-pcm-afe", "msm-lsm-client", + "msm-pcm-routing", "msm-compr-dsp", + "msm-pcm-dsp-noirq"; + asoc-cpu = <&dai_dp>, + <&dai_mi2s0>, <&dai_mi2s1>, + <&dai_mi2s2>, <&dai_mi2s3>, + <&dai_mi2s4>, <&dai_pri_auxpcm>, + <&dai_sec_auxpcm>, <&dai_tert_auxpcm>, + <&dai_quat_auxpcm>, <&dai_quin_auxpcm>, + <&afe_pcm_rx>, <&afe_pcm_tx>, <&afe_proxy_rx>, + <&afe_proxy_tx>, <&incall_record_rx>, + <&incall_record_tx>, <&incall_music_rx>, + <&incall_music_2_rx>, + <&sb_7_rx>, <&sb_7_tx>, <&sb_8_tx>, <&sb_8_rx>, + <&usb_audio_rx>, <&usb_audio_tx>, + <&dai_pri_tdm_rx_0>, <&dai_pri_tdm_tx_0>, + <&dai_sec_tdm_rx_0>, <&dai_sec_tdm_tx_0>, + <&dai_tert_tdm_rx_0>, <&dai_tert_tdm_tx_0>, + <&dai_quat_tdm_rx_0>, <&dai_quat_tdm_tx_0>, + <&dai_quin_tdm_rx_0>, <&dai_quin_tdm_tx_0>, + <&wsa_cdc_dma_0_rx>, <&wsa_cdc_dma_0_tx>, + <&wsa_cdc_dma_1_rx>, <&wsa_cdc_dma_1_tx>, + <&wsa_cdc_dma_2_tx>, + <&va_cdc_dma_0_tx>, <&va_cdc_dma_1_tx>, + <&rx_cdc_dma_0_rx>, <&tx_cdc_dma_0_tx>, + <&rx_cdc_dma_1_rx>, <&tx_cdc_dma_1_tx>, + <&rx_cdc_dma_2_rx>, <&tx_cdc_dma_2_tx>, + <&rx_cdc_dma_3_rx>, <&tx_cdc_dma_3_tx>, + <&rx_cdc_dma_4_rx>, <&tx_cdc_dma_4_tx>, + <&rx_cdc_dma_5_rx>, <&tx_cdc_dma_5_tx>, + <&tx_cdc_dma_6_tx>, <&tx_cdc_dma_7_tx>; + asoc-cpu-names = "msm-dai-q6-dp.24608", + "msm-dai-q6-mi2s.0", "msm-dai-q6-mi2s.1", + "msm-dai-q6-mi2s.2", "msm-dai-q6-mi2s.3", + "msm-dai-q6-mi2s.4", "msm-dai-q6-auxpcm.1", + "msm-dai-q6-auxpcm.2", "msm-dai-q6-auxpcm.3", + "msm-dai-q6-auxpcm.4", "msm-dai-q6-auxpcm.5", + "msm-dai-q6-dev.224", + "msm-dai-q6-dev.225", "msm-dai-q6-dev.241", + "msm-dai-q6-dev.240", "msm-dai-q6-dev.32771", + "msm-dai-q6-dev.32772", "msm-dai-q6-dev.32773", + "msm-dai-q6-dev.32770", "msm-dai-q6-dev.16398", + "msm-dai-q6-dev.16399", "msm-dai-q6-dev.16401", + "msm-dai-q6-dev.16400", + "msm-dai-q6-dev.28672", "msm-dai-q6-dev.28673", + "msm-dai-q6-tdm.36864", "msm-dai-q6-tdm.36865", + "msm-dai-q6-tdm.36880", "msm-dai-q6-tdm.36881", + "msm-dai-q6-tdm.36896", "msm-dai-q6-tdm.36897", + "msm-dai-q6-tdm.36912", "msm-dai-q6-tdm.36913", + "msm-dai-q6-tdm.36928", "msm-dai-q6-tdm.36929", + "msm-dai-cdc-dma-dev.45056", + "msm-dai-cdc-dma-dev.45057", + "msm-dai-cdc-dma-dev.45058", + "msm-dai-cdc-dma-dev.45059", + "msm-dai-cdc-dma-dev.45061", + "msm-dai-cdc-dma-dev.45089", + "msm-dai-cdc-dma-dev.45091", + "msm-dai-cdc-dma-dev.45120", + "msm-dai-cdc-dma-dev.45121", + "msm-dai-cdc-dma-dev.45122", + "msm-dai-cdc-dma-dev.45123", + "msm-dai-cdc-dma-dev.45124", + "msm-dai-cdc-dma-dev.45125", + "msm-dai-cdc-dma-dev.45126", + "msm-dai-cdc-dma-dev.45127", + "msm-dai-cdc-dma-dev.45128", + "msm-dai-cdc-dma-dev.45129", + "msm-dai-cdc-dma-dev.45130", + "msm-dai-cdc-dma-dev.45131", + "msm-dai-cdc-dma-dev.45133", + "msm-dai-cdc-dma-dev.45135"; + qcom,msm-mi2s-master = <1>, <1>, <1>, <1>, <1>; + qcom,msm-mbhc-hphl-swh = <1>; + qcom,msm-mbhc-gnd-swh = <1>; + qcom,cdc-dmic-gpios = <&cdc_dmic12_gpios>, <&cdc_dmic34_gpios>; + asoc-codec = <&stub_codec>, <&bolero>, + <&ext_disp_audio_codec>; + asoc-codec-names = "msm-stub-codec.1", "bolero-codec", + "msm-ext-disp-audio-codec-rx"; + qcom,wsa-max-devs = <2>; + qcom,wsa-devs = <&wsa881x_0211>, <&wsa881x_0212>, + <&wsa881x_0213>, <&wsa881x_0214>; + qcom,wsa-aux-dev-prefix = "SpkrLeft", "SpkrRight", + "SpkrLeft", "SpkrRight"; + qcom,codec-aux-devs = <&wcd937x_codec>; + qcom,msm_audio_ssr_devs = <&audio_apr>, <&q6core>; + }; +}; + +* MSMSTUB ASoC Machine driver + +Required properties: +- compatible : "qcom,sm6150-asoc-snd-stub" for SM6150 target. +- qcom,model : The user-visible name of this sound card. +- asoc-platform: This is phandle list containing the references to platform device + nodes that are used as part of the sound card dai-links. +- asoc-platform-names: This property contains list of platform names. The order of + the platform names should match to that of the phandle order + given in "asoc-platform". +- asoc-cpu: This is phandle list containing the references to cpu dai device nodes + that are used as part of the sound card dai-links. +- asoc-cpu-names: This property contains list of cpu dai names. The order of the + cpu dai names should match to that of the phandle order given + in "asoc-cpu". The cpu names are in the form of "%s.%d" form, + where the id (%d) field represents the back-end AFE port id that + this CPU dai is associated with. +- asoc-codec: This is phandle list containing the references to codec dai device + nodes that are used as part of the sound card dai-links. +- asoc-codec-names: This property contains list of codec dai names. The order of the + codec dai names should match to that of the phandle order given + in "asoc-codec". +Optional properties: +- qcom,wsa-max-devs : Maximum number of WSA881x devices present in the target + +Example: + + sound_stub { + compatible = "qcom,sm6150-asoc-snd-stub"; + qcom,model = "sm6150-stub-snd-card"; + + asoc-platform = <&pcm0>; + asoc-platform-names = "msm-pcm-dsp.0"; + asoc-cpu = <&sb_0_rx>, <&sb_0_tx>; + asoc-cpu-names = "msm-dai-q6-dev.16384", "msm-dai-q6-dev.16385"; + asoc-codec = <&stub_codec>; + asoc-codec-names = "msm-stub-codec.1"; + qcom,wsa-max-devs = <0>; + }; + +* SA8155 ASoC Machine driver + +Required properties: +- compatible : "qcom,sa8155-asoc-snd-auto" for auto adp codec and + "qcom,sa8155-asoc-snd-auto-custom" for auto custom codec. +- qcom,model : The user-visible name of this sound card. +- asoc-platform: This is phandle list containing the references to platform device + nodes that are used as part of the sound card dai-links. +- asoc-platform-names: This property contains list of platform names. The order of + the platform names should match to that of the phandle order + given in "asoc-platform". +- asoc-cpu: This is phandle list containing the references to cpu dai device nodes + that are used as part of the sound card dai-links. +- asoc-cpu-names: This property contains list of cpu dai names. The order of the + cpu dai names should match to that of the phandle order given + in "asoc-cpu". The cpu names are in the form of "%s.%d" form, + where the id (%d) field represents the back-end AFE port id that + this CPU dai is associated with. +- asoc-codec: This is phandle list containing the references to codec dai device + nodes that are used as part of the sound card dai-links. +- asoc-codec-names: This property contains list of codec dai names. The order of the + codec dai names should match to that of the phandle order given + in "asoc-codec". +Optional properties: +- qcom,mi2s-audio-intf : Property to specify if MI2S interface is used for the target +- qcom,auxpcm-audio-intf : Property to specify if AUX PCM interface is used for the target +- qcom,msm-mi2s-master : List of master/slave configuration for MI2S interfaces + +Example: + + sound-adp-star { + compatible = "qcom,sa8155-asoc-snd-adp-star"; + qcom,model = "sa8155-adp-star-snd-card"; + qcom,mi2s-audio-intf; + qcom,auxpcm-audio-intf; + qcom,msm-mi2s-master = <1>, <1>, <1>, <1>, <1>; + + asoc-platform = <&pcm0>, <&pcm1>, <&pcm2>, <&voip>, <&voice>, + <&loopback>, <&compress>, <&hostless>, + <&afe>, <&lsm>, <&routing>, <&compr>, + <&pcm_noirq>, <&loopback1>, <&pcm_dtmf>; + asoc-platform-names = "msm-pcm-dsp.0", "msm-pcm-dsp.1", + "msm-pcm-dsp.2", "msm-voip-dsp", + "msm-pcm-voice", "msm-pcm-loopback", + "msm-compress-dsp", "msm-pcm-hostless", + "msm-pcm-afe", "msm-lsm-client", + "msm-pcm-routing", "msm-compr-dsp", + "msm-pcm-dsp-noirq", "msm-pcm-loopback.1", + "msm-pcm-dtmf"; + asoc-cpu = <&dai_hdmi>, <&dai_dp>, + <&dai_mi2s0>, <&dai_mi2s1>, + <&dai_mi2s2>, <&dai_mi2s3>, + <&dai_mi2s4>, <&dai_pri_auxpcm>, + <&dai_sec_auxpcm>, <&dai_tert_auxpcm>, + <&dai_quat_auxpcm>, <&dai_quin_auxpcm>, + <&afe_pcm_rx>, <&afe_pcm_tx>, <&afe_proxy_rx>, + <&afe_proxy_tx>, <&incall_record_rx>, + <&incall_record_tx>, <&incall_music_rx>, + <&incall_music_2_rx>, + <&usb_audio_rx>, <&usb_audio_tx>, + <&dai_pri_tdm_rx_0>, <&dai_pri_tdm_rx_1>, + <&dai_pri_tdm_rx_2>, <&dai_pri_tdm_rx_3>, + <&dai_pri_tdm_tx_0>, <&dai_pri_tdm_tx_1>, + <&dai_pri_tdm_tx_2>, <&dai_pri_tdm_tx_3>, + <&dai_sec_tdm_rx_0>, <&dai_sec_tdm_rx_1>, + <&dai_sec_tdm_rx_2>, <&dai_sec_tdm_rx_3>, + <&dai_sec_tdm_tx_0>, <&dai_sec_tdm_tx_1>, + <&dai_sec_tdm_tx_2>, <&dai_sec_tdm_tx_3>, + <&dai_tert_tdm_rx_0>, <&dai_tert_tdm_rx_1>, + <&dai_tert_tdm_rx_2>, <&dai_tert_tdm_rx_3>, + <&dai_tert_tdm_rx_4>, <&dai_tert_tdm_tx_0>, + <&dai_tert_tdm_tx_1>, <&dai_tert_tdm_tx_2>, + <&dai_tert_tdm_tx_3>, <&dai_quat_tdm_rx_0>, + <&dai_quat_tdm_rx_1>, <&dai_quat_tdm_rx_2>, + <&dai_quat_tdm_rx_3>, <&dai_quat_tdm_tx_0>, + <&dai_quat_tdm_tx_1>, <&dai_quat_tdm_tx_2>, + <&dai_quat_tdm_tx_3>, <&dai_quin_tdm_rx_0>, + <&dai_quin_tdm_rx_1>, <&dai_quin_tdm_rx_2>, + <&dai_quin_tdm_rx_3>, <&dai_quin_tdm_tx_0>, + <&dai_quin_tdm_tx_1>, <&dai_quin_tdm_tx_2>, + <&dai_quin_tdm_tx_3>; + asoc-cpu-names = "msm-dai-q6-hdmi.8", "msm-dai-q6-dp.24608", + "msm-dai-q6-mi2s.0", "msm-dai-q6-mi2s.1", + "msm-dai-q6-mi2s.2", "msm-dai-q6-mi2s.3", + "msm-dai-q6-mi2s.4", "msm-dai-q6-auxpcm.1", + "msm-dai-q6-auxpcm.2", "msm-dai-q6-auxpcm.3", + "msm-dai-q6-auxpcm.4", "msm-dai-q6-auxpcm.5", + "msm-dai-q6-dev.224", "msm-dai-q6-dev.225", + "msm-dai-q6-dev.241", "msm-dai-q6-dev.240", + "msm-dai-q6-dev.32771", "msm-dai-q6-dev.32772", + "msm-dai-q6-dev.32773", "msm-dai-q6-dev.32770", + "msm-dai-q6-dev.28672", "msm-dai-q6-dev.28673", + "msm-dai-q6-tdm.36864", "msm-dai-q6-tdm.36866", + "msm-dai-q6-tdm.36868", "msm-dai-q6-tdm.36870", + "msm-dai-q6-tdm.36865", "msm-dai-q6-tdm.36867", + "msm-dai-q6-tdm.36869", "msm-dai-q6-tdm.36871", + "msm-dai-q6-tdm.36880", "msm-dai-q6-tdm.36882", + "msm-dai-q6-tdm.36884", "msm-dai-q6-tdm.36886", + "msm-dai-q6-tdm.36881", "msm-dai-q6-tdm.36883", + "msm-dai-q6-tdm.36885", "msm-dai-q6-tdm.36887", + "msm-dai-q6-tdm.36896", "msm-dai-q6-tdm.36898", + "msm-dai-q6-tdm.36900", "msm-dai-q6-tdm.36902", + "msm-dai-q6-tdm.36904", "msm-dai-q6-tdm.36897", + "msm-dai-q6-tdm.36899", "msm-dai-q6-tdm.36901", + "msm-dai-q6-tdm.36903", "msm-dai-q6-tdm.36912", + "msm-dai-q6-tdm.36914", "msm-dai-q6-tdm.36916", + "msm-dai-q6-tdm.36918", "msm-dai-q6-tdm.36913", + "msm-dai-q6-tdm.36915", "msm-dai-q6-tdm.36917", + "msm-dai-q6-tdm.36919", "msm-dai-q6-tdm.36928", + "msm-dai-q6-tdm.36930", "msm-dai-q6-tdm.36932", + "msm-dai-q6-tdm.36934", "msm-dai-q6-tdm.36929", + "msm-dai-q6-tdm.36931", "msm-dai-q6-tdm.36933", + "msm-dai-q6-tdm.36935"; + asoc-codec = <&stub_codec>; + asoc-codec-names = "msm-stub-codec.1"; + }; + +* KONA ASoC Machine driver + +Required properties: +- compatible : "qcom,kona-asoc-snd". +- qcom,model : The user-visible name of this sound card. +- qcom,audio-routing : A list of the connections between audio components. +- asoc-platform: This is phandle list containing the references to platform device + nodes that are used as part of the sound card dai-links. +- asoc-platform-names: This property contains list of platform names. The order of + the platform names should match to that of the phandle order + given in "asoc-platform". +- asoc-cpu: This is phandle list containing the references to cpu dai device nodes + that are used as part of the sound card dai-links. +- asoc-cpu-names: This property contains list of cpu dai names. The order of the + cpu dai names should match to that of the phandle order given + in "asoc-cpu". The cpu names are in the form of "%s.%d" form, + where the id (%d) field represents the back-end AFE port id that + this CPU dai is associated with. +- asoc-codec: This is phandle list containing the references to codec dai device + nodes that are used as part of the sound card dai-links. +- asoc-codec-names: This property contains list of codec dai names. The order of the + codec dai names should match to that of the phandle order given + in "asoc-codec". +- qcom,codec-aux-devs: This is phandle list containing the references to Auxilary + codec devices. + +Optional properties: +- qcom,msm-mi2s-master: This property is used to inform machine driver + if MSM is the clock master of mi2s. 1 means master and 0 means slave. The + first entry is primary mi2s; the second entry is secondary mi2s, and so on. +- qcom,msm-mbhc-hphl-swh: This property is used to distinguish headset HPHL + switch type on target typically the switch type will be normally open or + normally close, value for this property 0 for normally close and 1 for + normally open. +- qcom,msm-mbhc-gnd-swh: This property is used to distinguish headset GND + switch type on target typically the switch type will be normally open or + normally close, value for this property 0 for normally close and 1 for + normally open. +- qcom,wsa-max-devs : Maximum number of WSA881x devices present in the target +- qcom,wsa-devs : List of phandles for all possible WSA881x devices supported for the target +- qcom,wsa-aux-dev-prefix : Name prefix with Left/Right configuration for WSA881x device +- qcom,ext-disp-audio-rx: Property to specify if Audio over Display port is supported for the target +- qcom,wcn-btfm : Property to specify if WCN BT/FM chip is used for the target +- qcom,mi2s-audio-intf: Property to specify if MI2S interface is used for the target +- qcom,auxpcm-audio-intf: Property to specify if Aux PCM interface is used for the target +- qcom,cdc-dmic-gpios : phandle for Digital mic clk and data gpios. +- qcom,msm_audio_ssr_devs: List the snd event framework clients +- qcom,afe-rxtx-lb: AFE RX to TX loopback. + +Example: + kona_snd: sound { + status = "okay"; + compatible = "qcom,kona-asoc-snd"; + qcom,ext-disp-audio-rx = <1>; + qcom,wcn-btfm = <1>; + qcom,mi2s-audio-intf = <1>; + qcom,auxpcm-audio-intf = <1>; + qcom,afe-rxtx-lb = <1>; + + asoc-platform = <&pcm0>, <&pcm1>, <&pcm2>, <&voip>, <&voice>, + <&loopback>, <&compress>, <&hostless>, + <&afe>, <&lsm>, <&routing>, <&compr>, + <&pcm_noirq>; + asoc-platform-names = "msm-pcm-dsp.0", "msm-pcm-dsp.1", + "msm-pcm-dsp.2", "msm-voip-dsp", + "msm-pcm-voice", "msm-pcm-loopback", + "msm-compress-dsp", "msm-pcm-hostless", + "msm-pcm-afe", "msm-lsm-client", + "msm-pcm-routing", "msm-compr-dsp", + "msm-pcm-dsp-noirq"; + asoc-cpu = <&dai_dp>, + <&dai_mi2s0>, <&dai_mi2s1>, + <&dai_mi2s2>, <&dai_mi2s3>, + <&dai_mi2s4>, <&dai_pri_auxpcm>, + <&dai_sec_auxpcm>, <&dai_tert_auxpcm>, + <&dai_quat_auxpcm>, <&dai_quin_auxpcm>, + <&afe_pcm_rx>, <&afe_pcm_tx>, <&afe_proxy_rx>, + <&afe_proxy_tx>, <&incall_record_rx>, + <&incall_record_tx>, <&incall_music_rx>, + <&incall_music_2_rx>, + <&usb_audio_rx>, <&usb_audio_tx>, + <&dai_pri_tdm_rx_0>, <&dai_pri_tdm_tx_0>, + <&dai_sec_tdm_rx_0>, <&dai_sec_tdm_tx_0>, + <&dai_tert_tdm_rx_0>, <&dai_tert_tdm_tx_0>, + <&dai_quat_tdm_rx_0>, <&dai_quat_tdm_tx_0>, + <&dai_quin_tdm_rx_0>, <&dai_quin_tdm_tx_0>, + <&wsa_cdc_dma_0_rx>, <&wsa_cdc_dma_0_tx>, + <&wsa_cdc_dma_1_rx>, <&wsa_cdc_dma_1_tx>, + <&wsa_cdc_dma_2_tx>, + <&va_cdc_dma_0_tx>, <&va_cdc_dma_1_tx>, + <&rx_cdc_dma_0_rx>, <&tx_cdc_dma_0_tx>, + <&rx_cdc_dma_1_rx>, <&tx_cdc_dma_1_tx>, + <&rx_cdc_dma_2_rx>, <&tx_cdc_dma_2_tx>, + <&rx_cdc_dma_3_rx>, <&tx_cdc_dma_3_tx>, + <&rx_cdc_dma_4_rx>, <&tx_cdc_dma_4_tx>, + <&rx_cdc_dma_5_rx>, <&tx_cdc_dma_5_tx>, + <&tx_cdc_dma_6_tx>, <&tx_cdc_dma_7_tx>; + asoc-cpu-names = "msm-dai-q6-dp.24608", + "msm-dai-q6-mi2s.0", "msm-dai-q6-mi2s.1", + "msm-dai-q6-mi2s.2", "msm-dai-q6-mi2s.3", + "msm-dai-q6-mi2s.4", "msm-dai-q6-auxpcm.1", + "msm-dai-q6-auxpcm.2", "msm-dai-q6-auxpcm.3", + "msm-dai-q6-auxpcm.4", "msm-dai-q6-auxpcm.5", + "msm-dai-q6-dev.224", + "msm-dai-q6-dev.225", "msm-dai-q6-dev.241", + "msm-dai-q6-dev.240", "msm-dai-q6-dev.32771", + "msm-dai-q6-dev.32772", "msm-dai-q6-dev.32773", + "msm-dai-q6-dev.32770", "msm-dai-q6-dev.16398", + "msm-dai-q6-dev.16399", "msm-dai-q6-dev.16401", + "msm-dai-q6-dev.16400", + "msm-dai-q6-dev.28672", "msm-dai-q6-dev.28673", + "msm-dai-q6-tdm.36864", "msm-dai-q6-tdm.36865", + "msm-dai-q6-tdm.36880", "msm-dai-q6-tdm.36881", + "msm-dai-q6-tdm.36896", "msm-dai-q6-tdm.36897", + "msm-dai-q6-tdm.36912", "msm-dai-q6-tdm.36913", + "msm-dai-q6-tdm.36928", "msm-dai-q6-tdm.36929", + "msm-dai-cdc-dma-dev.45056", + "msm-dai-cdc-dma-dev.45057", + "msm-dai-cdc-dma-dev.45058", + "msm-dai-cdc-dma-dev.45059", + "msm-dai-cdc-dma-dev.45061", + "msm-dai-cdc-dma-dev.45089", + "msm-dai-cdc-dma-dev.45091", + "msm-dai-cdc-dma-dev.45120", + "msm-dai-cdc-dma-dev.45121", + "msm-dai-cdc-dma-dev.45122", + "msm-dai-cdc-dma-dev.45123", + "msm-dai-cdc-dma-dev.45124", + "msm-dai-cdc-dma-dev.45125", + "msm-dai-cdc-dma-dev.45126", + "msm-dai-cdc-dma-dev.45127", + "msm-dai-cdc-dma-dev.45128", + "msm-dai-cdc-dma-dev.45129", + "msm-dai-cdc-dma-dev.45130", + "msm-dai-cdc-dma-dev.45131", + "msm-dai-cdc-dma-dev.45133", + "msm-dai-cdc-dma-dev.45135"; + qcom,msm-mi2s-master = <1>, <1>, <1>, <1>, <1>; + qcom,msm-mbhc-hphl-swh = <1>; + qcom,msm-mbhc-gnd-swh = <1>; + qcom,cdc-dmic-gpios = <&cdc_dmic12_gpios>, <&cdc_dmic34_gpios>; + asoc-codec = <&stub_codec>, <&bolero>, + <&ext_disp_audio_codec>; + asoc-codec-names = "msm-stub-codec.1", "bolero-codec", + "msm-ext-disp-audio-codec-rx"; + qcom,wsa-max-devs = <2>; + qcom,wsa-devs = <&wsa881x_0211>, <&wsa881x_0212>, + <&wsa881x_0213>, <&wsa881x_0214>; + qcom,wsa-aux-dev-prefix = "SpkrLeft", "SpkrRight", + "SpkrLeft", "SpkrRight"; + qcom,codec-aux-devs = <&wcd937x_codec>; + qcom,msm_audio_ssr_devs = <&audio_apr>, <&q6core>; + }; + +* voice-mhi-audio + +Required properties: + - compatible : "qcom,voice-mhi-audio" + - memory-region : CMA region owned by this device + +Optional properties: + - voice_mhi_voting : Property that defines whether voting is needed or not for this device + +Example: + + qcom,voice-mhi-audio { + compatible = "qcom,voice-mhi-audio"; + memory-region = <&mailbox_mem>; + voice_mhi_voting; + }; + +}; diff --git a/bindings/sound/qcom-usb-audio-qmi-dev.txt b/bindings/sound/qcom-usb-audio-qmi-dev.txt new file mode 100644 index 00000000..9d3fb78f --- /dev/null +++ b/bindings/sound/qcom-usb-audio-qmi-dev.txt @@ -0,0 +1,26 @@ +QTI USB Audio QMI Device + +USB Audio QMI device is used to attach to remote processor IOMMU and +map USB Audio driver specific memory to iova to share with remote +processor. + +Required Properties: + +- compatible : "qcom,usb-audio-qmi-dev" + +- iommus : A list of phandle and IOMMU specifier pairs that describe the + IOMMU master interfaces of the device. + +- qcom,usb-audio-stream-id : Stream id is prepended to iova before passing + iova to remote processor. This allows remote processor to access iova. + +- qcom,usb-audio-intr-num : Interrupter number for external sub system + destination. + +Example: + usb_audio_qmi_dev { + compatible = "qcom,usb-audio-qmi-dev"; + iommus = <&lpass_q6_smmu 12>; + qcom,usb-audio-stream-id = <12>; + qcom,usb-audio-intr-num = <1>; + }; diff --git a/bindings/sound/renesas,fsi.txt b/bindings/sound/renesas,fsi.txt new file mode 100644 index 00000000..0cf0f819 --- /dev/null +++ b/bindings/sound/renesas,fsi.txt @@ -0,0 +1,31 @@ +Renesas FSI + +Required properties: +- compatible : "renesas,fsi2-<soctype>", + "renesas,sh_fsi2" or "renesas,sh_fsi" as + fallback. + Examples with soctypes are: + - "renesas,fsi2-r8a7740" (R-Mobile A1) + - "renesas,fsi2-sh73a0" (SH-Mobile AG5) +- reg : Should contain the register physical address and length +- interrupts : Should contain FSI interrupt + +- fsia,spdif-connection : FSI is connected by S/PDIF +- 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@ec230000 { + 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/bindings/sound/renesas,rsnd.txt b/bindings/sound/renesas,rsnd.txt new file mode 100644 index 00000000..9e764270 --- /dev/null +++ b/bindings/sound/renesas,rsnd.txt @@ -0,0 +1,684 @@ +Renesas R-Car sound + +============================================= +* Modules +============================================= + +Renesas R-Car and RZ/G sound is constructed from below modules +(for Gen2 or later) + + SCU : Sampling Rate Converter Unit + - SRC : Sampling Rate Converter + - CMD + - CTU : Channel Transfer Unit + - MIX : Mixer + - DVC : Digital Volume and Mute Function + SSIU : Serial Sound Interface Unit + SSI : Serial Sound Interface + +See detail of each module's channels, connection, limitation on datasheet + +============================================= +* Multi channel +============================================= + +Multi channel is supported by Multi-SSI, or TDM-SSI. + + Multi-SSI : 6ch case, you can use stereo x 3 SSI + TDM-SSI : 6ch case, you can use TDM + +============================================= +* Enable/Disable each modules +============================================= + +See datasheet to check SRC/CTU/MIX/DVC connect-limitation. +DT controls enabling/disabling module. +${LINUX}/arch/arm/boot/dts/r8a7790-lager.dts can be good example. +This is example of + +Playback: [MEM] -> [SRC2] -> [DVC0] -> [SSIU0/SSI0] -> [codec] +Capture: [MEM] <- [DVC1] <- [SRC3] <- [SSIU1/SSI1] <- [codec] + + &rcar_sound { + ... + rcar_sound,dai { + dai0 { + playback = <&ssi0 &src2 &dvc0>; + capture = <&ssi1 &src3 &dvc1>; + }; + }; + }; + +You can use below. +${LINUX}/arch/arm/boot/dts/r8a7790.dts can be good example. + + &src0 &ctu00 &mix0 &dvc0 &ssi0 + &src1 &ctu01 &mix1 &dvc1 &ssi1 + &src2 &ctu02 &ssi2 + &src3 &ctu03 &ssi3 + &src4 &ssi4 + &src5 &ctu10 &ssi5 + &src6 &ctu11 &ssi6 + &src7 &ctu12 &ssi7 + &src8 &ctu13 &ssi8 + &src9 &ssi9 + +============================================= +* SRC (Sampling Rate Converter) +============================================= + + [xx]Hz [yy]Hz + ------> [SRC] ------> + +SRC can convert [xx]Hz to [yy]Hz. Then, it has below 2 modes + + Asynchronous mode: input data / output data are based on different clocks. + you can use this mode on Playback / Capture + Synchronous mode: input data / output data are based on same clocks. + This mode will be used if system doesn't have its input clock, + for example digital TV case. + you can use this mode on Playback + +------------------ +** Asynchronous mode +------------------ + +You need to use "simple-scu-audio-card" sound card for it. +example) + + sound { + compatible = "simple-scu-audio-card"; + ... + /* + * SRC Asynchronous mode setting + * Playback: + * All input data will be converted to 48kHz + * Capture: + * Inputed 48kHz data will be converted to + * system specified Hz + */ + simple-audio-card,convert-rate = <48000>; + ... + simple-audio-card,cpu { + sound-dai = <&rcar_sound>; + }; + simple-audio-card,codec { + ... + }; + }; + +------------------ +** Synchronous mode +------------------ + + > amixer set "SRC Out Rate" on + > aplay xxxx.wav + > amixer set "SRC Out Rate" 48000 + > amixer set "SRC Out Rate" 44100 + +============================================= +* CTU (Channel Transfer Unit) +============================================= + + [xx]ch [yy]ch + ------> [CTU] --------> + +CTU can convert [xx]ch to [yy]ch, or exchange outputed channel. +CTU conversion needs matrix settings. +For more detail information, see below + + Renesas R-Car datasheet + - Sampling Rate Converter Unit (SCU) + - SCU Operation + - CMD Block + - Functional Blocks in CMD + + Renesas R-Car datasheet + - Sampling Rate Converter Unit (SCU) + - Register Description + - CTUn Scale Value exx Register (CTUn_SVxxR) + + ${LINUX}/sound/soc/sh/rcar/ctu.c + - comment of header + +You need to use "simple-scu-audio-card" sound card for it. +example) + + sound { + compatible = "simple-scu-audio-card"; + ... + /* + * CTU setting + * All input data will be converted to 2ch + * as output data + */ + simple-audio-card,convert-channels = <2>; + ... + simple-audio-card,cpu { + sound-dai = <&rcar_sound>; + }; + simple-audio-card,codec { + ... + }; + }; + +Ex) Exchange output channel + Input -> Output + 1ch -> 0ch + 0ch -> 1ch + + example of using matrix + output 0ch = (input 0ch x 0) + (input 1ch x 1) + output 1ch = (input 0ch x 1) + (input 1ch x 0) + + amixer set "CTU Reset" on + amixer set "CTU Pass" 9,10 + amixer set "CTU SV0" 0,4194304 + amixer set "CTU SV1" 4194304,0 + + example of changing connection + amixer set "CTU Reset" on + amixer set "CTU Pass" 2,1 + +============================================= +* MIX (Mixer) +============================================= + +MIX merges 2 sounds path. You can see 2 sound interface on system, +and these sounds will be merged by MIX. + + aplay -D plughw:0,0 xxxx.wav & + aplay -D plughw:0,1 yyyy.wav + +You need to use "simple-scu-audio-card" sound card for it. +Ex) + [MEM] -> [SRC1] -> [CTU02] -+-> [MIX0] -> [DVC0] -> [SSI0] + | + [MEM] -> [SRC2] -> [CTU03] -+ + + sound { + #address-cells = <1>; + #size-cells = <0>; + + compatible = "simple-scu-audio-card"; + ... + simple-audio-card,cpu@0 { + reg = <0>; + sound-dai = <&rcar_sound 0>; + }; + simple-audio-card,cpu@1 { + reg = <1>; + sound-dai = <&rcar_sound 1>; + }; + simple-audio-card,codec { + ... + }; + }; + + &rcar_sound { + ... + rcar_sound,dai { + dai0 { + playback = <&src1 &ctu02 &mix0 &dvc0 &ssi0>; + }; + dai1 { + playback = <&src2 &ctu03 &mix0 &dvc0 &ssi0>; + }; + }; + }; + +============================================= +* DVC (Digital Volume and Mute Function) +============================================= + +DVC controls Playback/Capture volume. + +Playback Volume + amixer set "DVC Out" 100% + +Capture Volume + amixer set "DVC In" 100% + +Playback Mute + amixer set "DVC Out Mute" on + +Capture Mute + amixer set "DVC In Mute" on + +Volume Ramp + amixer set "DVC Out Ramp Up Rate" "0.125 dB/64 steps" + amixer set "DVC Out Ramp Down Rate" "0.125 dB/512 steps" + amixer set "DVC Out Ramp" on + aplay xxx.wav & + amixer set "DVC Out" 80% // Volume Down + amixer set "DVC Out" 100% // Volume Up + +============================================= +* SSIU (Serial Sound Interface Unit) +============================================= + +There is no DT settings for SSIU, because SSIU will be automatically +selected via SSI. +SSIU can avoid some under/over run error, because it has some buffer. +But you can't use it if SSI was PIO mode. +In DMA mode, you can select not to use SSIU by using "no-busif" on DT. + + &ssi0 { + no-busif; + }; + +============================================= +* SSI (Serial Sound Interface) +============================================= + +** PIO mode + +You can use PIO mode which is for connection check by using. +Note: The system will drop non-SSI modules in PIO mode +even though if DT is selecting other modules. + + &ssi0 { + pio-transfer + }; + +** DMA mode without SSIU + +You can use DMA without SSIU. +Note: under/over run, or noise are likely to occur + + &ssi0 { + no-busif; + }; + +** PIN sharing + +Each SSI can share WS pin. It is based on platform. +This is example if SSI1 want to share WS pin with SSI0 + + &ssi1 { + shared-pin; + }; + +** Multi-SSI + +You can use Multi-SSI. +This is example of SSI0/SSI1/SSI2 (= for 6ch) + + &rcar_sound { + ... + rcar_sound,dai { + dai0 { + playback = <&ssi0 &ssi1 &ssi2 &src0 &dvc0>; + }; + }; + }; + +** TDM-SSI + +You can use TDM with SSI. +This is example of TDM 6ch. +Driver can automatically switches TDM <-> stereo mode in this case. + + rsnd_tdm: sound { + compatible = "simple-audio-card"; + ... + simple-audio-card,cpu { + /* system can use TDM 6ch */ + dai-tdm-slot-num = <6>; + sound-dai = <&rcar_sound>; + }; + simple-audio-card,codec { + ... + }; + }; + + +============================================= +Required properties: +============================================= + +- compatible : "renesas,rcar_sound-<soctype>", fallbacks + "renesas,rcar_sound-gen1" if generation1, and + "renesas,rcar_sound-gen2" if generation2 (or RZ/G1) + "renesas,rcar_sound-gen3" if generation3 + Examples with soctypes are: + - "renesas,rcar_sound-r8a7743" (RZ/G1M) + - "renesas,rcar_sound-r8a7745" (RZ/G1E) + - "renesas,rcar_sound-r8a7778" (R-Car M1A) + - "renesas,rcar_sound-r8a7779" (R-Car H1) + - "renesas,rcar_sound-r8a7790" (R-Car H2) + - "renesas,rcar_sound-r8a7791" (R-Car M2-W) + - "renesas,rcar_sound-r8a7793" (R-Car M2-N) + - "renesas,rcar_sound-r8a7794" (R-Car E2) + - "renesas,rcar_sound-r8a7795" (R-Car H3) + - "renesas,rcar_sound-r8a7796" (R-Car M3-W) + - "renesas,rcar_sound-r8a77965" (R-Car M3-N) +- 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,ctu : Should contain CTU feature. + The number of CTU subnode should be same as HW. + see below for detail. +- rcar_sound,mix : Should contain MIX feature. + The number of MIX subnode should be same as HW. + see below for detail. +- rcar_sound,dvc : Should contain DVC feature. + The number of DVC 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. +- #sound-dai-cells : it must be 0 if your system is using single DAI + it must be 1 if your system is using multi DAI +- clocks : References to SSI/SRC/MIX/CTU/DVC/AUDIO_CLK clocks. +- clock-names : List of necessary clock names. + "ssi-all", "ssi.X", "src.X", "mix.X", "ctu.X", + "dvc.X", "clk_a", "clk_b", "clk_c", "clk_i" + +Optional properties: +- #clock-cells : it must be 0 if your system has audio_clkout + it must be 1 if your system has audio_clkout0/1/2/3 +- clock-frequency : for all audio_clkout0/1/2/3 +- clkout-lr-asynchronous : boolean property. it indicates that audio_clkoutn + is asynchronizes with lr-clock. +- resets : References to SSI resets. +- reset-names : List of valid reset names. + "ssi-all", "ssi.X" + +SSI subnode properties: +- interrupts : Should contain SSI interrupt for PIO transfer +- shared-pin : if shared clock pin +- pio-transfer : use PIO transfer mode +- no-busif : BUSIF is not ussed when [mem -> SSI] via DMA case +- dma : Should contain Audio DMAC entry +- dma-names : SSI case "rx" (=playback), "tx" (=capture) + SSIU case "rxu" (=playback), "txu" (=capture) + +SRC subnode properties: +- dma : Should contain Audio DMAC entry +- dma-names : "rx" (=playback), "tx" (=capture) + +DVC subnode properties: +- dma : Should contain Audio DMAC entry +- dma-names : "tx" (=playback/capture) + +DAI subnode properties: +- playback : list of playback modules +- capture : list of capture modules + + +============================================= +Example: +============================================= + +rcar_sound: sound@ec500000 { + #sound-dai-cells = <1>; + compatible = "renesas,rcar_sound-r8a7791", "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 */ + <0 0xec740000 0 0x200>; /* Audio DMAC peri peri*/ + reg-names = "scu", "adg", "ssiu", "ssi", "audmapp"; + + clocks = <&mstp10_clks R8A7790_CLK_SSI_ALL>, + <&mstp10_clks R8A7790_CLK_SSI9>, <&mstp10_clks R8A7790_CLK_SSI8>, + <&mstp10_clks R8A7790_CLK_SSI7>, <&mstp10_clks R8A7790_CLK_SSI6>, + <&mstp10_clks R8A7790_CLK_SSI5>, <&mstp10_clks R8A7790_CLK_SSI4>, + <&mstp10_clks R8A7790_CLK_SSI3>, <&mstp10_clks R8A7790_CLK_SSI2>, + <&mstp10_clks R8A7790_CLK_SSI1>, <&mstp10_clks R8A7790_CLK_SSI0>, + <&mstp10_clks R8A7790_CLK_SCU_SRC9>, <&mstp10_clks R8A7790_CLK_SCU_SRC8>, + <&mstp10_clks R8A7790_CLK_SCU_SRC7>, <&mstp10_clks R8A7790_CLK_SCU_SRC6>, + <&mstp10_clks R8A7790_CLK_SCU_SRC5>, <&mstp10_clks R8A7790_CLK_SCU_SRC4>, + <&mstp10_clks R8A7790_CLK_SCU_SRC3>, <&mstp10_clks R8A7790_CLK_SCU_SRC2>, + <&mstp10_clks R8A7790_CLK_SCU_SRC1>, <&mstp10_clks R8A7790_CLK_SCU_SRC0>, + <&mstp10_clks R8A7790_CLK_SCU_DVC0>, <&mstp10_clks R8A7790_CLK_SCU_DVC1>, + <&audio_clk_a>, <&audio_clk_b>, <&audio_clk_c>, <&m2_clk>; + clock-names = "ssi-all", + "ssi.9", "ssi.8", "ssi.7", "ssi.6", "ssi.5", + "ssi.4", "ssi.3", "ssi.2", "ssi.1", "ssi.0", + "src.9", "src.8", "src.7", "src.6", "src.5", + "src.4", "src.3", "src.2", "src.1", "src.0", + "dvc.0", "dvc.1", + "clk_a", "clk_b", "clk_c", "clk_i"; + + rcar_sound,dvc { + dvc0: dvc-0 { + dmas = <&audma0 0xbc>; + dma-names = "tx"; + }; + dvc1: dvc-1 { + dmas = <&audma0 0xbe>; + dma-names = "tx"; + }; + }; + + rcar_sound,mix { + mix0: mix-0 { }; + mix1: mix-1 { }; + }; + + rcar_sound,ctu { + ctu00: ctu-0 { }; + ctu01: ctu-1 { }; + ctu02: ctu-2 { }; + ctu03: ctu-3 { }; + ctu10: ctu-4 { }; + ctu11: ctu-5 { }; + ctu12: ctu-6 { }; + ctu13: ctu-7 { }; + }; + + rcar_sound,src { + src0: src-0 { + interrupts = <0 352 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x85>, <&audma1 0x9a>; + dma-names = "rx", "tx"; + }; + src1: src-1 { + interrupts = <0 353 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x87>, <&audma1 0x9c>; + dma-names = "rx", "tx"; + }; + src2: src-2 { + interrupts = <0 354 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x89>, <&audma1 0x9e>; + dma-names = "rx", "tx"; + }; + src3: src-3 { + interrupts = <0 355 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x8b>, <&audma1 0xa0>; + dma-names = "rx", "tx"; + }; + src4: src-4 { + interrupts = <0 356 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x8d>, <&audma1 0xb0>; + dma-names = "rx", "tx"; + }; + src5: src-5 { + interrupts = <0 357 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x8f>, <&audma1 0xb2>; + dma-names = "rx", "tx"; + }; + src6: src-6 { + interrupts = <0 358 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x91>, <&audma1 0xb4>; + dma-names = "rx", "tx"; + }; + src7: src-7 { + interrupts = <0 359 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x93>, <&audma1 0xb6>; + dma-names = "rx", "tx"; + }; + src8: src-8 { + interrupts = <0 360 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x95>, <&audma1 0xb8>; + dma-names = "rx", "tx"; + }; + src9: src-9 { + interrupts = <0 361 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x97>, <&audma1 0xba>; + dma-names = "rx", "tx"; + }; + }; + + rcar_sound,ssi { + ssi0: ssi-0 { + interrupts = <0 370 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x01>, <&audma1 0x02>, <&audma0 0x15>, <&audma1 0x16>; + dma-names = "rx", "tx", "rxu", "txu"; + }; + ssi1: ssi-1 { + interrupts = <0 371 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x03>, <&audma1 0x04>, <&audma0 0x49>, <&audma1 0x4a>; + dma-names = "rx", "tx", "rxu", "txu"; + }; + ssi2: ssi-2 { + interrupts = <0 372 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x05>, <&audma1 0x06>, <&audma0 0x63>, <&audma1 0x64>; + dma-names = "rx", "tx", "rxu", "txu"; + }; + ssi3: ssi-3 { + interrupts = <0 373 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x07>, <&audma1 0x08>, <&audma0 0x6f>, <&audma1 0x70>; + dma-names = "rx", "tx", "rxu", "txu"; + }; + ssi4: ssi-4 { + interrupts = <0 374 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x09>, <&audma1 0x0a>, <&audma0 0x71>, <&audma1 0x72>; + dma-names = "rx", "tx", "rxu", "txu"; + }; + ssi5: ssi-5 { + interrupts = <0 375 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x0b>, <&audma1 0x0c>, <&audma0 0x73>, <&audma1 0x74>; + dma-names = "rx", "tx", "rxu", "txu"; + }; + ssi6: ssi-6 { + interrupts = <0 376 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x0d>, <&audma1 0x0e>, <&audma0 0x75>, <&audma1 0x76>; + dma-names = "rx", "tx", "rxu", "txu"; + }; + ssi7: ssi-7 { + interrupts = <0 377 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x0f>, <&audma1 0x10>, <&audma0 0x79>, <&audma1 0x7a>; + dma-names = "rx", "tx", "rxu", "txu"; + }; + ssi8: ssi-8 { + interrupts = <0 378 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x11>, <&audma1 0x12>, <&audma0 0x7b>, <&audma1 0x7c>; + dma-names = "rx", "tx", "rxu", "txu"; + }; + ssi9: ssi-9 { + interrupts = <0 379 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x13>, <&audma1 0x14>, <&audma0 0x7d>, <&audma1 0x7e>; + dma-names = "rx", "tx", "rxu", "txu"; + }; + }; + + rcar_sound,dai { + dai0 { + playback = <&ssi5 &src5>; + capture = <&ssi6>; + }; + dai1 { + playback = <&ssi3>; + }; + dai2 { + capture = <&ssi4>; + }; + dai3 { + playback = <&ssi7>; + }; + dai4 { + capture = <&ssi8>; + }; + }; +}; + +============================================= +Example: simple sound card +============================================= + + rsnd_ak4643: sound { + compatible = "simple-audio-card"; + + simple-audio-card,format = "left_j"; + simple-audio-card,bitclock-master = <&sndcodec>; + simple-audio-card,frame-master = <&sndcodec>; + + sndcpu: simple-audio-card,cpu { + sound-dai = <&rcar_sound>; + }; + + sndcodec: simple-audio-card,codec { + sound-dai = <&ak4643>; + clocks = <&audio_clock>; + }; + }; + +&rcar_sound { + pinctrl-0 = <&sound_pins &sound_clk_pins>; + pinctrl-names = "default"; + + /* Single DAI */ + #sound-dai-cells = <0>; + + + rcar_sound,dai { + dai0 { + playback = <&ssi0 &src2 &dvc0>; + capture = <&ssi1 &src3 &dvc1>; + }; + }; +}; + +&ssi1 { + shared-pin; +}; + +============================================= +Example: simple sound card for TDM +============================================= + + rsnd_tdm: sound { + compatible = "simple-audio-card"; + + simple-audio-card,format = "left_j"; + simple-audio-card,bitclock-master = <&sndcodec>; + simple-audio-card,frame-master = <&sndcodec>; + + sndcpu: simple-audio-card,cpu { + sound-dai = <&rcar_sound>; + dai-tdm-slot-num = <6>; + }; + + sndcodec: simple-audio-card,codec { + sound-dai = <&xxx>; + }; + }; + +============================================= +Example: simple sound card for Multi channel +============================================= + +&rcar_sound { + pinctrl-0 = <&sound_pins &sound_clk_pins>; + pinctrl-names = "default"; + + /* Single DAI */ + #sound-dai-cells = <0>; + + + rcar_sound,dai { + dai0 { + playback = <&ssi0 &ssi1 &ssi2 &src0 &dvc0>; + }; + }; +}; diff --git a/bindings/sound/rockchip,pdm.txt b/bindings/sound/rockchip,pdm.txt new file mode 100644 index 00000000..47f164fb --- /dev/null +++ b/bindings/sound/rockchip,pdm.txt @@ -0,0 +1,41 @@ +* Rockchip PDM controller + +Required properties: + +- compatible: "rockchip,pdm" +- reg: physical base address of the controller and length of memory mapped + region. +- dmas: DMA specifiers for rx dma. See the DMA client binding, + Documentation/devicetree/bindings/dma/dma.txt +- dma-names: should include "rx". +- clocks: a list of phandle + clock-specifer pairs, one for each entry in clock-names. +- clock-names: should contain following: + - "pdm_hclk": clock for PDM BUS + - "pdm_clk" : clock for PDM controller +- pinctrl-names: Must contain a "default" entry. +- pinctrl-N: One property must exist for each entry in + pinctrl-names. See ../pinctrl/pinctrl-bindings.txt + for details of the property values. + +Example for rk3328 PDM controller: + +pdm: pdm@ff040000 { + compatible = "rockchip,pdm"; + reg = <0x0 0xff040000 0x0 0x1000>; + clocks = <&clk_pdm>, <&clk_gates28 0>; + clock-names = "pdm_clk", "pdm_hclk"; + dmas = <&pdma 16>; + #dma-cells = <1>; + dma-names = "rx"; + pinctrl-names = "default", "sleep"; + pinctrl-0 = <&pdmm0_clk + &pdmm0_sdi0 + &pdmm0_sdi1 + &pdmm0_sdi2 + &pdmm0_sdi3>; + pinctrl-1 = <&pdmm0_clk_sleep + &pdmm0_sdi0_sleep + &pdmm0_sdi1_sleep + &pdmm0_sdi2_sleep + &pdmm0_sdi3_sleep>; +}; diff --git a/bindings/sound/rockchip,rk3288-hdmi-analog.txt b/bindings/sound/rockchip,rk3288-hdmi-analog.txt new file mode 100644 index 00000000..e5430d1d --- /dev/null +++ b/bindings/sound/rockchip,rk3288-hdmi-analog.txt @@ -0,0 +1,36 @@ +ROCKCHIP RK3288 with HDMI and analog audio + +Required properties: +- compatible: "rockchip,rk3288-hdmi-analog" +- rockchip,model: The user-visible name of this sound complex +- rockchip,i2s-controller: The phandle of the Rockchip I2S controller that's + connected to the CODEC +- rockchip,audio-codec: The phandle of the analog audio codec. +- rockchip,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. For this driver the first string should always be + "Analog". + +Optionnal properties: +- rockchip,hp-en-gpios = The phandle of the GPIO that power up/down the + headphone (when the analog output is an headphone). +- rockchip,hp-det-gpios = The phandle of the GPIO that detects the headphone + (when the analog output is an headphone). +- pinctrl-names, pinctrl-0: Please refer to pinctrl-bindings.txt + +Example: + +sound { + compatible = "rockchip,rk3288-hdmi-analog"; + rockchip,model = "Analog audio output"; + rockchip,i2s-controller = <&i2s>; + rockchip,audio-codec = <&es8388>; + rockchip,routing = "Analog", "LOUT2", + "Analog", "ROUT2"; + rockchip,hp-en-gpios = <&gpio8 0 GPIO_ACTIVE_HIGH>; + rockchip,hp-det-gpios = <&gpio7 7 GPIO_ACTIVE_HIGH>; + pinctrl-names = "default"; + pinctrl-0 = <&headphone>; +}; + diff --git a/bindings/sound/rockchip,rk3399-gru-sound.txt b/bindings/sound/rockchip,rk3399-gru-sound.txt new file mode 100644 index 00000000..72d3cf4c --- /dev/null +++ b/bindings/sound/rockchip,rk3399-gru-sound.txt @@ -0,0 +1,22 @@ +ROCKCHIP with MAX98357A/RT5514/DA7219 codecs on GRU boards + +Required properties: +- compatible: "rockchip,rk3399-gru-sound" +- rockchip,cpu: The phandle of the Rockchip I2S controller that's + connected to the codecs +- rockchip,codec: The phandle of the audio codecs + +Optional properties: +- dmic-wakeup-delay-ms : specify delay time (ms) for DMIC ready. + If this option is specified, which means it's required dmic need + delay for DMIC to ready so that rt5514 can avoid recording before + DMIC send valid data + +Example: + +sound { + compatible = "rockchip,rk3399-gru-sound"; + rockchip,cpu = <&i2s0>; + rockchip,codec = <&max98357a &rt5514 &da7219>; + dmic-wakeup-delay-ms = <20>; +}; diff --git a/bindings/sound/rockchip-i2s.txt b/bindings/sound/rockchip-i2s.txt new file mode 100644 index 00000000..54aefab7 --- /dev/null +++ b/bindings/sound/rockchip-i2s.txt @@ -0,0 +1,49 @@ +* Rockchip I2S controller + +The I2S bus (Inter-IC sound bus) is a serial link for digital +audio data transfer between devices in the system. + +Required properties: + +- compatible: should be one of the following: + - "rockchip,rk3066-i2s": for rk3066 + - "rockchip,px30-i2s", "rockchip,rk3066-i2s": for px30 + - "rockchip,rk3036-i2s", "rockchip,rk3066-i2s": for rk3036 + - "rockchip,rk3188-i2s", "rockchip,rk3066-i2s": for rk3188 + - "rockchip,rk3228-i2s", "rockchip,rk3066-i2s": for rk3228 + - "rockchip,rk3288-i2s", "rockchip,rk3066-i2s": for rk3288 + - "rockchip,rk3328-i2s", "rockchip,rk3066-i2s": for rk3328 + - "rockchip,rk3366-i2s", "rockchip,rk3066-i2s": for rk3366 + - "rockchip,rk3368-i2s", "rockchip,rk3066-i2s": for rk3368 + - "rockchip,rk3399-i2s", "rockchip,rk3066-i2s": for rk3399 +- reg: physical base address of the controller and length of memory mapped + region. +- interrupts: should contain the I2S interrupt. +- dmas: DMA specifiers for tx and rx dma. See the DMA client binding, + Documentation/devicetree/bindings/dma/dma.txt +- dma-names: should include "tx" and "rx". +- clocks: a list of phandle + clock-specifer pairs, one for each entry in clock-names. +- clock-names: should contain the following: + - "i2s_hclk": clock for I2S BUS + - "i2s_clk" : clock for I2S controller +- rockchip,playback-channels: max playback channels, if not set, 8 channels default. +- rockchip,capture-channels: max capture channels, if not set, 2 channels default. + +Required properties for controller which support multi channels +playback/capture: + +- rockchip,grf: the phandle of the syscon node for GRF register. + +Example for rk3288 I2S controller: + +i2s@ff890000 { + compatible = "rockchip,rk3288-i2s", "rockchip,rk3066-i2s"; + reg = <0xff890000 0x10000>; + interrupts = <GIC_SPI 85 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&pdma1 0>, <&pdma1 1>; + dma-names = "tx", "rx"; + clock-names = "i2s_hclk", "i2s_clk"; + clocks = <&cru HCLK_I2S0>, <&cru SCLK_I2S0>; + rockchip,playback-channels = <8>; + rockchip,capture-channels = <2>; +}; diff --git a/bindings/sound/rockchip-max98090.txt b/bindings/sound/rockchip-max98090.txt new file mode 100644 index 00000000..a805aa99 --- /dev/null +++ b/bindings/sound/rockchip-max98090.txt @@ -0,0 +1,19 @@ +ROCKCHIP with MAX98090 CODEC + +Required properties: +- compatible: "rockchip,rockchip-audio-max98090" +- rockchip,model: The user-visible name of this sound complex +- rockchip,i2s-controller: The phandle of the Rockchip I2S controller that's + connected to the CODEC +- rockchip,audio-codec: The phandle of the MAX98090 audio codec +- rockchip,headset-codec: The phandle of Ext chip for jack detection + +Example: + +sound { + compatible = "rockchip,rockchip-audio-max98090"; + rockchip,model = "ROCKCHIP-I2S"; + rockchip,i2s-controller = <&i2s>; + rockchip,audio-codec = <&max98090>; + rockchip,headset-codec = <&headsetcodec>; +}; diff --git a/bindings/sound/rockchip-rt5645.txt b/bindings/sound/rockchip-rt5645.txt new file mode 100644 index 00000000..411a62b3 --- /dev/null +++ b/bindings/sound/rockchip-rt5645.txt @@ -0,0 +1,17 @@ +ROCKCHIP with RT5645/RT5650 CODECS + +Required properties: +- compatible: "rockchip,rockchip-audio-rt5645" +- rockchip,model: The user-visible name of this sound complex +- rockchip,i2s-controller: The phandle of the Rockchip I2S controller that's + connected to the CODEC +- rockchip,audio-codec: The phandle of the RT5645/RT5650 audio codec + +Example: + +sound { + compatible = "rockchip,rockchip-audio-rt5645"; + rockchip,model = "ROCKCHIP-I2S"; + rockchip,i2s-controller = <&i2s>; + rockchip,audio-codec = <&rt5645>; +}; diff --git a/bindings/sound/rockchip-spdif.txt b/bindings/sound/rockchip-spdif.txt new file mode 100644 index 00000000..ec20c127 --- /dev/null +++ b/bindings/sound/rockchip-spdif.txt @@ -0,0 +1,45 @@ +* Rockchip SPDIF transceiver + +The 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: should be one of the following: + - "rockchip,rk3066-spdif" + - "rockchip,rk3188-spdif" + - "rockchip,rk3228-spdif" + - "rockchip,rk3288-spdif" + - "rockchip,rk3328-spdif" + - "rockchip,rk3366-spdif" + - "rockchip,rk3368-spdif" + - "rockchip,rk3399-spdif" +- reg: physical base address of the controller and length of memory mapped + region. +- interrupts: should contain the SPDIF interrupt. +- dmas: DMA specifiers for tx dma. See the DMA client binding, + Documentation/devicetree/bindings/dma/dma.txt +- dma-names: should be "tx" +- clocks: a list of phandle + clock-specifier pairs, one for each entry + in clock-names. +- clock-names: should contain following: + - "hclk": clock for SPDIF controller + - "mclk" : clock for SPDIF bus + +Required properties on RK3288: + - rockchip,grf: the phandle of the syscon node for the general register + file (GRF) + +Example for the rk3188 SPDIF controller: + +spdif: spdif@1011e000 { + compatible = "rockchip,rk3188-spdif", "rockchip,rk3066-spdif"; + reg = <0x1011e000 0x2000>; + interrupts = <GIC_SPI 32 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&dmac1_s 8>; + dma-names = "tx"; + clock-names = "hclk", "mclk"; + clocks = <&cru HCLK_SPDIF>, <&cru SCLK_SPDIF>; + #sound-dai-cells = <0>; +}; diff --git a/bindings/sound/rohm,bd28623.txt b/bindings/sound/rohm,bd28623.txt new file mode 100644 index 00000000..d84557c2 --- /dev/null +++ b/bindings/sound/rohm,bd28623.txt @@ -0,0 +1,29 @@ +ROHM BD28623MUV Class D speaker amplifier for digital input + +This codec does not have any control buses such as I2C, it detect format and +rate of I2S signal automatically. It has two signals that can be connected +to GPIOs: reset and mute. + +Required properties: +- compatible : should be "rohm,bd28623" +- #sound-dai-cells: should be 0. +- VCCA-supply : regulator phandle for the VCCA supply +- VCCP1-supply : regulator phandle for the VCCP1 supply +- VCCP2-supply : regulator phandle for the VCCP2 supply + +Optional properties: +- reset-gpios : GPIO specifier for the active low reset line +- mute-gpios : GPIO specifier for the active low mute line + +Example: + + codec { + compatible = "rohm,bd28623"; + #sound-dai-cells = <0>; + + VCCA-supply = <&vcc_reg>; + VCCP1-supply = <&vcc_reg>; + VCCP2-supply = <&vcc_reg>; + reset-gpios = <&gpio 0 GPIO_ACTIVE_LOW>; + mute-gpios = <&gpio 1 GPIO_ACTIVE_LOW>; + }; diff --git a/bindings/sound/rt274.txt b/bindings/sound/rt274.txt new file mode 100644 index 00000000..791a1bd7 --- /dev/null +++ b/bindings/sound/rt274.txt @@ -0,0 +1,33 @@ +RT274 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : "realtek,rt274". + +- reg : The I2C address of the device. + +Optional properties: + +- interrupts : The CODEC's interrupt output. + + +Pins on the device (for linking into audio routes) for RT274: + + * DMIC1 Pin + * DMIC2 Pin + * MIC + * LINE1 + * LINE2 + * HPO Pin + * SPDIF + * LINE3 + +Example: + +rt274: codec@1c { + compatible = "realtek,rt274"; + reg = <0x1c>; + interrupts = <7 IRQ_TYPE_EDGE_FALLING>; +}; diff --git a/bindings/sound/rt5514.txt b/bindings/sound/rt5514.txt new file mode 100644 index 00000000..d2cc171f --- /dev/null +++ b/bindings/sound/rt5514.txt @@ -0,0 +1,37 @@ +RT5514 audio CODEC + +This device supports both I2C and SPI. + +Required properties: + +- compatible : "realtek,rt5514". + +- reg : the I2C address of the device for I2C, the chip select + number for SPI. + +Optional properties: + +- clocks: The phandle of the master clock to the CODEC +- clock-names: Should be "mclk" + +- interrupts: The interrupt number to the cpu. The interrupt specifier format + depends on the interrupt controller. + +- realtek,dmic-init-delay-ms + Set the DMIC initial delay (ms) to wait it ready for I2C. + +Pins on the device (for linking into audio routes) for I2C: + + * DMIC1L + * DMIC1R + * DMIC2L + * DMIC2R + * AMICL + * AMICR + +Example: + +rt5514: codec@57 { + compatible = "realtek,rt5514"; + reg = <0x57>; +}; diff --git a/bindings/sound/rt5616.txt b/bindings/sound/rt5616.txt new file mode 100644 index 00000000..540a4bf2 --- /dev/null +++ b/bindings/sound/rt5616.txt @@ -0,0 +1,32 @@ +RT5616 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : "realtek,rt5616". + +- reg : The I2C address of the device. + +Optional properties: + +- clocks: The phandle of the master clock to the CODEC. + +- clock-names: Should be "mclk". + +Pins on the device (for linking into audio routes) for RT5616: + + * IN1P + * IN2P + * IN2N + * LOUTL + * LOUTR + * HPOL + * HPOR + +Example: + +rt5616: codec@1b { + compatible = "realtek,rt5616"; + reg = <0x1b>; +}; diff --git a/bindings/sound/rt5631.txt b/bindings/sound/rt5631.txt new file mode 100644 index 00000000..92b986ca --- /dev/null +++ b/bindings/sound/rt5631.txt @@ -0,0 +1,48 @@ +ALC5631/RT5631 audio CODEC + +This device supports I2C only. + +Required properties: + + - compatible : "realtek,alc5631" or "realtek,rt5631" + + - reg : the I2C address of the device. + +Pins on the device (for linking into audio routes): + + * SPK_OUT_R_P + * SPK_OUT_R_N + * SPK_OUT_L_P + * SPK_OUT_L_N + * HP_OUT_L + * HP_OUT_R + * AUX_OUT2_LP + * AUX_OUT2_RN + * AUX_OUT1_LP + * AUX_OUT1_RN + * AUX_IN_L_JD + * AUX_IN_R_JD + * MONO_IN_P + * MONO_IN_N + * MIC1_P + * MIC1_N + * MIC2_P + * MIC2_N + * MONO_OUT_P + * MONO_OUT_N + * MICBIAS1 + * MICBIAS2 + +Example: + +alc5631: alc5631@1a { + compatible = "realtek,alc5631"; + reg = <0x1a>; +}; + +or + +rt5631: rt5631@1a { + compatible = "realtek,rt5631"; + reg = <0x1a>; +}; diff --git a/bindings/sound/rt5640.txt b/bindings/sound/rt5640.txt new file mode 100644 index 00000000..e40e4893 --- /dev/null +++ b/bindings/sound/rt5640.txt @@ -0,0 +1,94 @@ +RT5640/RT5639 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : One of "realtek,rt5640" or "realtek,rt5639". + +- reg : The I2C address of the device. + +- interrupts : The CODEC's interrupt output. + +Optional properties: + +- clocks: The phandle of the master clock to the CODEC +- clock-names: Should be "mclk" + +- realtek,in1-differential +- realtek,in2-differential +- realtek,in3-differential + Boolean. Indicate MIC1/2/3 input are differential, rather than single-ended. + +- realtek,ldo1-en-gpios : The GPIO that controls the CODEC's LDO1_EN pin. + +- realtek,dmic1-data-pin + 0: dmic1 is not used + 1: using IN1P pin as dmic1 data pin + 2: using GPIO3 pin as dmic1 data pin + +- realtek,dmic2-data-pin + 0: dmic2 is not used + 1: using IN1N pin as dmic2 data pin + 2: using GPIO4 pin as dmic2 data pin + +- realtek,jack-detect-source + u32. Valid values: + 0: jack-detect is not used + 1: Use GPIO1 for jack-detect + 2: Use JD1_IN4P for jack-detect + 3: Use JD2_IN4N for jack-detect + 4: Use GPIO2 for jack-detect + 5: Use GPIO3 for jack-detect + 6: Use GPIO4 for jack-detect + +- realtek,jack-detect-not-inverted + bool. Normal jack-detect switches give an inverted signal, set this bool + in the rare case you've a jack-detect switch which is not inverted. + +- realtek,over-current-threshold-microamp + u32, micbias over-current detection threshold in µA, valid values are + 600, 1500 and 2000µA. + +- realtek,over-current-scale-factor + u32, micbias over-current detection scale-factor, valid values are: + 0: Scale current by 0.5 + 1: Scale current by 0.75 + 2: Scale current by 1.0 + 3: Scale current by 1.5 + +Pins on the device (for linking into audio routes) for RT5639/RT5640: + + * DMIC1 + * DMIC2 + * MICBIAS1 + * IN1P + * IN1N + * IN2P + * IN2N + * IN3P + * IN3N + * HPOL + * HPOR + * LOUTL + * LOUTR + * SPOLP + * SPOLN + * SPORP + * SPORN + +Additional pins on the device for RT5640: + + * MONOP + * MONON + +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/bindings/sound/rt5645.txt b/bindings/sound/rt5645.txt new file mode 100644 index 00000000..a03f9a87 --- /dev/null +++ b/bindings/sound/rt5645.txt @@ -0,0 +1,72 @@ +RT5650/RT5645 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : One of "realtek,rt5645" or "realtek,rt5650". + +- reg : The I2C address of the device. + +- interrupts : The CODEC's interrupt output. + +Optional properties: + +- hp-detect-gpios: + a GPIO spec for the external headphone detect pin. If jd-mode = 0, + we will get the JD status by getting the value of hp-detect-gpios. + +- realtek,in2-differential + Boolean. Indicate MIC2 input are differential, rather than single-ended. + +- realtek,dmic1-data-pin + 0: dmic1 is not used + 1: using IN2P pin as dmic1 data pin + 2: using GPIO6 pin as dmic1 data pin + 3: using GPIO10 pin as dmic1 data pin + 4: using GPIO12 pin as dmic1 data pin + +- realtek,dmic2-data-pin + 0: dmic2 is not used + 1: using IN2N pin as dmic2 data pin + 2: using GPIO5 pin as dmic2 data pin + 3: using GPIO11 pin as dmic2 data pin + +-- realtek,jd-mode : The JD mode of rt5645/rt5650 + 0 : rt5645/rt5650 JD function is not used + 1 : Mode-0 (VDD=3.3V), two port jack detection + 2 : Mode-1 (VDD=3.3V), one port jack detection + 3 : Mode-2 (VDD=1.8V), one port jack detection + +Pins on the device (for linking into audio routes) for RT5645/RT5650: + + * DMIC L1 + * DMIC R1 + * DMIC L2 + * DMIC R2 + * IN1P + * IN1N + * IN2P + * IN2N + * Haptic Generator + * HPOL + * HPOR + * LOUTL + * LOUTR + * PDM1L + * PDM1R + * SPOL + * SPOR + +Example: + +codec: rt5650@1a { + compatible = "realtek,rt5650"; + reg = <0x1a>; + hp-detect-gpios = <&gpio 19 0>; + interrupt-parent = <&gpio>; + interrupts = <7 IRQ_TYPE_EDGE_FALLING>; + realtek,dmic-en = "true"; + realtek,en-jd-func = "true"; + realtek,jd-mode = <3>; +}; diff --git a/bindings/sound/rt5651.txt b/bindings/sound/rt5651.txt new file mode 100644 index 00000000..a41199a5 --- /dev/null +++ b/bindings/sound/rt5651.txt @@ -0,0 +1,58 @@ +RT5651 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : "realtek,rt5651". + +- reg : The I2C address of the device. + +Optional properties: + +- realtek,in2-differential + Boolean. Indicate MIC2 input are differential, rather than single-ended. + +- realtek,dmic-en + Boolean. true if dmic is used. + +- realtek,jack-detect-source + u32. Valid values: + 1: Use JD1_1 pin for jack-detect + 2: Use JD1_2 pin for jack-detect + 3: Use JD2 pin for jack-detect + +- realtek,over-current-threshold-microamp + u32, micbias over-current detection threshold in µA, valid values are + 600, 1500 and 2000µA. + +- realtek,over-current-scale-factor + u32, micbias over-current detection scale-factor, valid values are: + 0: Scale current by 0.5 + 1: Scale current by 0.75 + 2: Scale current by 1.0 + 3: Scale current by 1.5 + +Pins on the device (for linking into audio routes) for RT5651: + + * DMIC L1 + * DMIC R1 + * IN1P + * IN2P + * IN2N + * IN3P + * HPOL + * HPOR + * LOUTL + * LOUTR + * PDML + * PDMR + +Example: + +rt5651: codec@1a { + compatible = "realtek,rt5651"; + reg = <0x1a>; + realtek,dmic-en = "true"; + realtek,in2-diff = "false"; +}; diff --git a/bindings/sound/rt5659.txt b/bindings/sound/rt5659.txt new file mode 100644 index 00000000..1766e054 --- /dev/null +++ b/bindings/sound/rt5659.txt @@ -0,0 +1,78 @@ +RT5659/RT5658 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : One of "realtek,rt5659" or "realtek,rt5658". + +- reg : The I2C address of the device. + +- interrupts : The CODEC's interrupt output. + +Optional properties: + +- clocks: The phandle of the master clock to the CODEC +- clock-names: Should be "mclk" + +- realtek,in1-differential +- realtek,in3-differential +- realtek,in4-differential + Boolean. Indicate MIC1/3/4 input are differential, rather than single-ended. + +- realtek,dmic1-data-pin + 0: dmic1 is not used + 1: using IN2N pin as dmic1 data pin + 2: using GPIO5 pin as dmic1 data pin + 3: using GPIO9 pin as dmic1 data pin + 4: using GPIO11 pin as dmic1 data pin + +- realtek,dmic2-data-pin + 0: dmic2 is not used + 1: using IN2P pin as dmic2 data pin + 2: using GPIO6 pin as dmic2 data pin + 3: using GPIO10 pin as dmic2 data pin + 4: using GPIO12 pin as dmic2 data pin + +- realtek,jd-src + 0: No JD is used + 1: using JD3 as JD source + +- realtek,ldo1-en-gpios : The GPIO that controls the CODEC's LDO1_EN pin. +- realtek,reset-gpios : The GPIO that controls the CODEC's RESET pin. + +Pins on the device (for linking into audio routes) for RT5659/RT5658: + + * DMIC L1 + * DMIC R1 + * DMIC L2 + * DMIC R2 + * IN1P + * IN1N + * IN2P + * IN2N + * IN3P + * IN3N + * IN4P + * IN4N + * HPOL + * HPOR + * SPOL + * SPOR + * LOUTL + * LOUTR + * MONOOUT + * PDML + * PDMR + * SPDIF + +Example: + +rt5659 { + compatible = "realtek,rt5659"; + reg = <0x1b>; + 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/bindings/sound/rt5660.txt b/bindings/sound/rt5660.txt new file mode 100644 index 00000000..30be5f92 --- /dev/null +++ b/bindings/sound/rt5660.txt @@ -0,0 +1,47 @@ +RT5660 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : "realtek,rt5660". + +- reg : The I2C address of the device. + +Optional properties: + +- clocks: The phandle of the master clock to the CODEC +- clock-names: Should be "mclk" + +- realtek,in1-differential +- realtek,in3-differential + Boolean. Indicate MIC1/3 input are differential, rather than single-ended. + +- realtek,poweroff-in-suspend + Boolean. If the codec will be powered off in suspend, the resume should be + added delay time for waiting codec power ready. + +- realtek,dmic1-data-pin + 0: dmic1 is not used + 1: using GPIO2 pin as dmic1 data pin + 2: using IN1P pin as dmic1 data pin + +Pins on the device (for linking into audio routes) for RT5660: + + * DMIC L1 + * DMIC R1 + * IN1P + * IN1N + * IN2P + * IN3P + * IN3N + * SPO + * LOUTL + * LOUTR + +Example: + +rt5660 { + compatible = "realtek,rt5660"; + reg = <0x1c>; +}; diff --git a/bindings/sound/rt5663.txt b/bindings/sound/rt5663.txt new file mode 100644 index 00000000..23386446 --- /dev/null +++ b/bindings/sound/rt5663.txt @@ -0,0 +1,54 @@ +RT5663 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : "realtek,rt5663". + +- reg : The I2C address of the device. + +- interrupts : The CODEC's interrupt output. + +Optional properties: + +- "realtek,dc_offset_l_manual" +- "realtek,dc_offset_r_manual" +- "realtek,dc_offset_l_manual_mic" +- "realtek,dc_offset_r_manual_mic" + Based on the different PCB layout, add the manual offset value to + compensate the DC offset for each L and R channel, and they are different + between headphone and headset. +- "realtek,impedance_sensing_num" + The matrix row number of the impedance sensing table. + If the value is 0, it means the impedance sensing is not supported. +- "realtek,impedance_sensing_table" + The matrix rows of the impedance sensing table are consisted by impedance + minimum, impedance maximun, volume, DC offset w/o and w/ mic of each L and + R channel accordingly. Example is shown as following. + < 0 300 7 0xffd160 0xffd1c0 0xff8a10 0xff8ab0 + 301 65535 4 0xffe470 0xffe470 0xffb8e0 0xffb8e0> + The first and second column are defined for the impedance range. If the + detected impedance value is in the range, then the volume value of the + third column will be set to codec. In our codec design, each volume value + should compensate different DC offset to avoid the pop sound, and it is + also different between headphone and headset. In the example, the + "realtek,impedance_sensing_num" is 2. It means that there are 2 ranges of + impedance in the impedance sensing function. + +Pins on the device (for linking into audio routes) for RT5663: + + * IN1P + * IN1N + * IN2P + * IN2N + * HPOL + * HPOR + +Example: + +rt5663: codec@12 { + compatible = "realtek,rt5663"; + reg = <0x12>; + interrupts = <7 IRQ_TYPE_EDGE_FALLING>; +}; diff --git a/bindings/sound/rt5665.txt b/bindings/sound/rt5665.txt new file mode 100644 index 00000000..8df17050 --- /dev/null +++ b/bindings/sound/rt5665.txt @@ -0,0 +1,68 @@ +RT5665/RT5666 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : One of "realtek,rt5665", "realtek,rt5666". + +- reg : The I2C address of the device. + +- interrupts : The CODEC's interrupt output. + +Optional properties: + +- realtek,in1-differential +- realtek,in2-differential +- realtek,in3-differential +- realtek,in4-differential + Boolean. Indicate MIC1/2/3/4 input are differential, rather than single-ended. + +- realtek,dmic1-data-pin + 0: dmic1 is not used + 1: using GPIO4 pin as dmic1 data pin + 2: using IN2N pin as dmic2 data pin + +- realtek,dmic2-data-pin + 0: dmic2 is not used + 1: using GPIO5 pin as dmic2 data pin + 2: using IN2P pin as dmic2 data pin + +- realtek,jd-src + 0: No JD is used + 1: using JD1 as JD source + +- realtek,ldo1-en-gpios : The GPIO that controls the CODEC's LDO1_EN pin. + +Pins on the device (for linking into audio routes) for RT5659/RT5658: + + * DMIC L1 + * DMIC R1 + * DMIC L2 + * DMIC R2 + * IN1P + * IN1N + * IN2P + * IN2N + * IN3P + * IN3N + * IN4P + * IN4N + * HPOL + * HPOR + * LOUTL + * LOUTR + * MONOOUT + * PDML + * PDMR + +Example: + +rt5659 { + compatible = "realtek,rt5665"; + reg = <0x1b>; + 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/bindings/sound/rt5668.txt b/bindings/sound/rt5668.txt new file mode 100644 index 00000000..c88b96e7 --- /dev/null +++ b/bindings/sound/rt5668.txt @@ -0,0 +1,50 @@ +RT5668B audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : "realtek,rt5668b" + +- reg : The I2C address of the device. + +Optional properties: + +- interrupts : The CODEC's interrupt output. + +- realtek,dmic1-data-pin + 0: dmic1 is not used + 1: using GPIO2 pin as dmic1 data pin + 2: using GPIO5 pin as dmic1 data pin + +- realtek,dmic1-clk-pin + 0: using GPIO1 pin as dmic1 clock pin + 1: using GPIO3 pin as dmic1 clock pin + +- realtek,jd-src + 0: No JD is used + 1: using JD1 as JD source + +- realtek,ldo1-en-gpios : The GPIO that controls the CODEC's LDO1_EN pin. + +Pins on the device (for linking into audio routes) for RT5668B: + + * DMIC L1 + * DMIC R1 + * IN1P + * HPOL + * HPOR + +Example: + +rt5668 { + compatible = "realtek,rt5668b"; + reg = <0x1a>; + interrupt-parent = <&gpio>; + interrupts = <TEGRA_GPIO(U, 6) GPIO_ACTIVE_HIGH>; + realtek,ldo1-en-gpios = + <&gpio TEGRA_GPIO(R, 2) GPIO_ACTIVE_HIGH>; + realtek,dmic1-data-pin = <1>; + realtek,dmic1-clk-pin = <1>; + realtek,jd-src = <1>; +}; diff --git a/bindings/sound/rt5677.txt b/bindings/sound/rt5677.txt new file mode 100644 index 00000000..1b3c13d2 --- /dev/null +++ b/bindings/sound/rt5677.txt @@ -0,0 +1,78 @@ +RT5677 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : "realtek,rt5677". + +- reg : The I2C address of the device. + +- interrupts : The CODEC's interrupt output. + +- 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: + +- realtek,pow-ldo2-gpio : The GPIO that controls the CODEC's POW_LDO2 pin. +- realtek,reset-gpio : The GPIO that controls the CODEC's RESET pin. Active low. + +- realtek,in1-differential +- realtek,in2-differential +- realtek,lout1-differential +- realtek,lout2-differential +- realtek,lout3-differential + Boolean. Indicate MIC1/2 input and LOUT1/2/3 outputs are differential, + rather than single-ended. + +- realtek,gpio-config + Array of six 8bit elements that configures GPIO. + 0 - floating (reset value) + 1 - pull down + 2 - pull up + +- realtek,jd1-gpio + Configures GPIO Mic Jack detection 1. + Select 0 ~ 3 as OFF, GPIO1, GPIO2 and GPIO3 respectively. + +- realtek,jd2-gpio +- realtek,jd3-gpio + Configures GPIO Mic Jack detection 2 and 3. + Select 0 ~ 3 as OFF, GPIO4, GPIO5 and GPIO6 respectively. + +Pins on the device (for linking into audio routes): + + * IN1P + * IN1N + * IN2P + * IN2N + * MICBIAS1 + * DMIC1 + * DMIC2 + * DMIC3 + * DMIC4 + * LOUT1 + * LOUT2 + * LOUT3 + +Example: + +rt5677 { + compatible = "realtek,rt5677"; + reg = <0x2c>; + interrupt-parent = <&gpio>; + interrupts = <TEGRA_GPIO(W, 3) GPIO_ACTIVE_HIGH>; + + gpio-controller; + #gpio-cells = <2>; + + realtek,pow-ldo2-gpio = + <&gpio TEGRA_GPIO(V, 3) GPIO_ACTIVE_HIGH>; + realtek,reset-gpio = <&gpio TEGRA_GPIO(BB, 3) GPIO_ACTIVE_LOW>; + realtek,in1-differential = "true"; + realtek,gpio-config = /bits/ 8 <0 0 0 0 0 2>; /* pull up GPIO6 */ + realtek,jd2-gpio = <3>; /* Enables Jack detection for GPIO6 */ +}; diff --git a/bindings/sound/rt5682.txt b/bindings/sound/rt5682.txt new file mode 100644 index 00000000..312e9a12 --- /dev/null +++ b/bindings/sound/rt5682.txt @@ -0,0 +1,50 @@ +RT5682 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : "realtek,rt5682" or "realtek,rt5682i" + +- reg : The I2C address of the device. + +Optional properties: + +- interrupts : The CODEC's interrupt output. + +- realtek,dmic1-data-pin + 0: dmic1 is not used + 1: using GPIO2 pin as dmic1 data pin + 2: using GPIO5 pin as dmic1 data pin + +- realtek,dmic1-clk-pin + 0: using GPIO1 pin as dmic1 clock pin + 1: using GPIO3 pin as dmic1 clock pin + +- realtek,jd-src + 0: No JD is used + 1: using JD1 as JD source + +- realtek,ldo1-en-gpios : The GPIO that controls the CODEC's LDO1_EN pin. + +Pins on the device (for linking into audio routes) for RT5682: + + * DMIC L1 + * DMIC R1 + * IN1P + * HPOL + * HPOR + +Example: + +rt5682 { + compatible = "realtek,rt5682i"; + reg = <0x1a>; + interrupt-parent = <&gpio>; + interrupts = <TEGRA_GPIO(U, 6) GPIO_ACTIVE_HIGH>; + realtek,ldo1-en-gpios = + <&gpio TEGRA_GPIO(R, 2) GPIO_ACTIVE_HIGH>; + realtek,dmic1-data-pin = <1>; + realtek,dmic1-clk-pin = <1>; + realtek,jd-src = <1>; +}; diff --git a/bindings/sound/samsung,odroid.txt b/bindings/sound/samsung,odroid.txt new file mode 100644 index 00000000..e9da2200 --- /dev/null +++ b/bindings/sound/samsung,odroid.txt @@ -0,0 +1,54 @@ +Samsung Exynos Odroid XU3/XU4 audio complex with MAX98090 codec + +Required properties: + + - compatible - "hardkernel,odroid-xu3-audio" - for Odroid XU3 board, + "hardkernel,odroid-xu4-audio" - for Odroid XU4 board (deprecated), + "samsung,odroid-xu3-audio" - for Odroid XU3 board (deprecated), + "samsung,odroid-xu4-audio" - for Odroid XU4 board (deprecated) + - model - the user-visible name of this sound complex + - clocks - should contain entries matching clock names in the clock-names + property + - samsung,audio-widgets - this property specifies off-codec audio elements + like headphones or speakers, for details see widgets.txt + - samsung,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 + + For Odroid X2: + "Headphone Jack", "Mic Jack", "DMIC" + + For Odroid U3, XU3: + "Headphone Jack", "Speakers" + + For Odroid XU4: + no entries + +Required sub-nodes: + + - 'cpu' subnode with a 'sound-dai' property containing the phandle of the I2S + controller + - 'codec' subnode with a 'sound-dai' property containing list of phandles + to the CODEC nodes, first entry must be corresponding to the MAX98090 + CODEC and the second entry must be the phandle of the HDMI IP block node + +Example: + +sound { + compatible = "hardkernel,odroid-xu3-audio"; + model = "Odroid-XU3"; + samsung,audio-routing = + "Headphone Jack", "HPL", + "Headphone Jack", "HPR", + "IN1", "Mic Jack", + "Mic Jack", "MICBIAS"; + + cpu { + sound-dai = <&i2s0 0>; + }; + codec { + sound-dai = <&hdmi>, <&max98090>; + }; +}; diff --git a/bindings/sound/samsung,smdk-wm8994.txt b/bindings/sound/samsung,smdk-wm8994.txt new file mode 100644 index 00000000..4686646f --- /dev/null +++ b/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/bindings/sound/samsung,tm2-audio.txt b/bindings/sound/samsung,tm2-audio.txt new file mode 100644 index 00000000..f5ccc12d --- /dev/null +++ b/bindings/sound/samsung,tm2-audio.txt @@ -0,0 +1,42 @@ +Samsung Exynos5433 TM2(E) audio complex with WM5110 codec + +Required properties: + + - compatible : "samsung,tm2-audio" + - model : the user-visible name of this sound complex + - audio-codec : the first entry should be phandle of the wm5110 audio + codec node, as described in ../mfd/arizona.txt; + the second entry should be phandle of the HDMI + transmitter node + - i2s-controller : the list of phandle and argument tuples pointing to + I2S controllers, the first entry should be I2S0 and + the second one I2S1 + - audio-amplifier : the phandle of the MAX98504 amplifier + - samsung,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 + WM5110's and MAX98504's pins and the jacks on the + board: HP, SPK, Main Mic, Sub Mic, Third Mic, + Headset Mic + - mic-bias-gpios : GPIO pin that enables the Main Mic bias regulator + + +Example: + +sound { + compatible = "samsung,tm2-audio"; + audio-codec = <&wm5110>, <&hdmi>; + i2s-controller = <&i2s0 0>, <&i2s1 0>; + audio-amplifier = <&max98504>; + mic-bias-gpios = <&gpr3 2 0>; + model = "wm5110"; + samsung,audio-routing = + "HP", "HPOUT1L", + "HP", "HPOUT1R", + "SPK", "SPKOUT", + "SPKOUT", "HPOUT2L", + "SPKOUT", "HPOUT2R", + "Main Mic", "MICBIAS2", + "IN1R", "Main Mic"; +}; diff --git a/bindings/sound/samsung-i2s.txt b/bindings/sound/samsung-i2s.txt new file mode 100644 index 00000000..a88cb00f --- /dev/null +++ b/bindings/sound/samsung-i2s.txt @@ -0,0 +1,84 @@ +* 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(5.1) I2S for + playback, stereo channel capture, secondary fifo using internal + or external dma, s/w reset control, internal mux for root clk src + and 7.1 channel TDM support for playback. TDM (Time division multiplexing) + is to allow transfer of multiple channel audio data on single data line. + - samsung,exynos7-i2s: with all the available features of exynos5 i2s, + exynos7 I2S has 7.1 channel TDM support for capture, secondary fifo + with only external dma and more no.of root clk sampling frequencies. + - samsung,exynos7-i2s1: I2S1 on previous samsung platforms supports + stereo channels. exynos7 i2s1 upgraded to 5.1 multichannel with + slightly modified bit offsets. + +- 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 clocks 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. +- #clock-cells: should be 1, this property must be present if the I2S device + is a clock provider in terms of the common clock bindings, described in + ../clock/clock-bindings.txt. +- clock-output-names (deprecated): from the common clock bindings, names of + the CDCLK I2S output clocks, suggested values are "i2s_cdclk0", "i2s_cdclk1", + "i2s_cdclk3" for the I2S0, I2S1, I2S2 devices respectively. + +There are following clocks available at the I2S device nodes: + CLK_I2S_CDCLK - the CDCLK (CODECLKO) gate clock, + CLK_I2S_RCLK_PSR - the RCLK prescaler divider clock (corresponding to the + IISPSR register), + CLK_I2S_RCLK_SRC - the RCLKSRC mux clock (corresponding to RCLKSRC bit in + IISMOD register). + +Refer to the SoC datasheet for availability of the above clocks. +The CLK_I2S_RCLK_PSR and CLK_I2S_RCLK_SRC clocks are usually only available +in the IIS Multi Audio Interface. + +Note: Old DTs may not have the #clock-cells property and then not use the I2S +node as a clock supplier. + +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". +- #sound-dai-cells: should be 1. + + +Example: + +i2s0: i2s@3830000 { + 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"; + #clock-cells = <1>; + samsung,idma-addr = <0x03000000>; + pinctrl-names = "default"; + pinctrl-0 = <&i2s0_bus>; + #sound-dai-cells = <1>; +}; diff --git a/bindings/sound/sgtl5000.txt b/bindings/sound/sgtl5000.txt new file mode 100644 index 00000000..9c58f724 --- /dev/null +++ b/bindings/sound/sgtl5000.txt @@ -0,0 +1,51 @@ +* Freescale SGTL5000 Stereo Codec + +Required properties: +- compatible : "fsl,sgtl5000". + +- reg : the I2C address of the device + +- #sound-dai-cells: must be equal to 0 + +- clocks : the clock provider of SYS_MCLK + +- VDDA-supply : the regulator provider of VDDA + +- VDDIO-supply: the regulator provider of VDDIO + +Optional properties: + +- VDDD-supply : the regulator provider of VDDD + +- micbias-resistor-k-ohms : the bias resistor to be used in kOhms + The resistor can take values of 2k, 4k or 8k. + If set to 0 it will be off. + If this node is not mentioned or if the value is unknown, then + micbias resistor is set to 4K. + +- micbias-voltage-m-volts : the bias voltage to be used in mVolts + The voltage can take values from 1.25V to 3V by 250mV steps + If this node is not mentioned or the value is unknown, then + the value is set to 1.25V. + +- lrclk-strength: the LRCLK pad strength. Possible values are: +0, 1, 2 and 3 as per the table below: + +VDDIO 1.8V 2.5V 3.3V +0 = Disable +1 = 1.66 mA 2.87 mA 4.02 mA +2 = 3.33 mA 5.74 mA 8.03 mA +3 = 4.99 mA 8.61 mA 12.05 mA + +Example: + +sgtl5000: codec@a { + compatible = "fsl,sgtl5000"; + reg = <0x0a>; + #sound-dai-cells = <0>; + clocks = <&clks 150>; + micbias-resistor-k-ohms = <2>; + micbias-voltage-m-volts = <2250>; + VDDA-supply = <®_3p3v>; + VDDIO-supply = <®_3p3v>; +}; diff --git a/bindings/sound/simple-amplifier.txt b/bindings/sound/simple-amplifier.txt new file mode 100644 index 00000000..8647edae --- /dev/null +++ b/bindings/sound/simple-amplifier.txt @@ -0,0 +1,12 @@ +Simple Amplifier Audio Driver + +Required properties: +- compatible : "dioo,dio2125" or "simple-audio-amplifier" +- enable-gpios : the gpio connected to the enable pin of the simple amplifier + +Example: + +amp: analog-amplifier { + compatible = "simple-audio-amplifier"; + enable-gpios = <&gpio GPIOH_3 0>; +}; diff --git a/bindings/sound/simple-card.txt b/bindings/sound/simple-card.txt new file mode 100644 index 00000000..a4c72d09 --- /dev/null +++ b/bindings/sound/simple-card.txt @@ -0,0 +1,212 @@ +Simple-Card: + +Simple-Card specifies audio DAI connections 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,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. +- simple-audio-card,mclk-fs : Multiplication factor between stream rate and codec + mclk. When defined, mclk-fs property defined in + dai-link sub nodes are ignored. +- simple-audio-card,hp-det-gpio : Reference to GPIO that signals when + headphones are attached. +- simple-audio-card,mic-det-gpio : Reference to GPIO that signals when + a microphone is attached. +- simple-audio-card,aux-devs : List of phandles pointing to auxiliary devices, such + as amplifiers, to be added to the sound card. + +Optional subnodes: + +- simple-audio-card,dai-link : Container for dai-link level + properties and the CPU and CODEC + sub-nodes. This container may be + omitted when the card has only one + DAI link. See the examples and the + section below. + +Dai-link subnode properties and subnodes: + +If dai-link subnode is omitted and the subnode properties are directly +under "sound"-node the subnode property and subnode names have to be +prefixed with "simple-audio-card,"-prefix. + +Required dai-link subnodes: + +- cpu : CPU sub-node +- codec : CODEC sub-node + +Optional dai-link subnode properties: + +- format : CPU/CODEC common audio format. + "i2s", "right_j", "left_j" , "dsp_a" + "dsp_b", "ac97", "pdm", "msb", "lsb" +- frame-master : Indicates dai-link frame master. + phandle to a cpu or codec subnode. +- bitclock-master : Indicates dai-link bit clock master. + phandle to a cpu or codec subnode. +- bitclock-inversion : bool property. Add this if the + dai-link uses bit clock inversion. +- frame-inversion : bool property. Add this if the + dai-link uses frame clock inversion. +- mclk-fs : Multiplication factor between stream + rate and codec mclk, applied only for + the dai-link. + +For backward compatibility the frame-master and bitclock-master +properties can be used as booleans in codec subnode to indicate if the +codec is the dai-link frame or bit clock master. In this case there +should be no dai-link node, the same properties should not be present +at sound-node level, and the bitclock-inversion and frame-inversion +properties should also be placed in the codec node if needed. + +Required CPU/CODEC subnodes properties: + +- sound-dai : phandle and port of CPU/CODEC + +Optional CPU/CODEC subnodes properties: + +- dai-tdm-slot-num : Please refer to tdm-slot.txt. +- dai-tdm-slot-width : Please refer to tdm-slot.txt. +- 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) + If a clock is specified, it is + enabled with clk_prepare_enable() + in dai startup() and disabled with + clk_disable_unprepare() in dai + shutdown(). + If a clock is specified and a + multiplication factor is given with + mclk-fs, the clock will be set to the + calculated mclk frequency when the + stream starts. +- system-clock-direction-out : specifies clock direction as 'out' on + initialization. It is useful for some aCPUs with + fixed clocks. + +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,bitclock-master = <&dailink0_master>; + simple-audio-card,frame-master = <&dailink0_master>; + 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"; + + simple-audio-card,cpu { + sound-dai = <&sh_fsi2 0>; + }; + + dailink0_master: simple-audio-card,codec { + sound-dai = <&ak4648>; + 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,dai-link@0 { /* I2S - HDMI */ + reg = <0>; + format = "i2s"; + cpu { + sound-dai = <&audio1 0>; + }; + codec { + sound-dai = <&tda998x 0>; + }; + }; + + simple-audio-card,dai-link@1 { /* S/PDIF - HDMI */ + reg = <1>; + cpu { + sound-dai = <&audio1 1>; + }; + codec { + sound-dai = <&tda998x 1>; + }; + }; + + simple-audio-card,dai-link@2 { /* S/PDIF - S/PDIF */ + reg = <2>; + cpu { + sound-dai = <&audio1 1>; + }; + codec { + sound-dai = <&spdif_codec>; + }; + }; +}; + +Example 3 - route audio from IMX6 SSI2 through TLV320DAC3100 codec +through TPA6130A2 amplifier to headphones: + +&i2c0 { + codec: tlv320dac3100@18 { + compatible = "ti,tlv320dac3100"; + ... + } + + amp: tpa6130a2@60 { + compatible = "ti,tpa6130a2"; + ... + } +} + +sound { + compatible = "simple-audio-card"; + ... + simple-audio-card,widgets = + "Headphone", "Headphone Jack"; + simple-audio-card,routing = + "Headphone Jack", "HPLEFT", + "Headphone Jack", "HPRIGHT", + "LEFTIN", "HPL", + "RIGHTIN", "HPR"; + simple-audio-card,aux-devs = <&>; + simple-audio-card,cpu { + sound-dai = <&ssi2>; + }; + simple-audio-card,codec { + sound-dai = <&codec>; + clocks = ... + }; +}; diff --git a/bindings/sound/simple-scu-card.txt b/bindings/sound/simple-scu-card.txt new file mode 100644 index 00000000..32f8dbce --- /dev/null +++ b/bindings/sound/simple-scu-card.txt @@ -0,0 +1,94 @@ +ASoC Simple SCU Sound Card + +Simple SCU Sound Card is "Simple Sound Card" + "ALSA DPCM". +For example, you can use this driver if you want to exchange sampling rate convert, +Mixing, etc... + +Required properties: + +- compatible : "simple-scu-audio-card" + "renesas,rsrc-card" +Optional properties: + +- simple-audio-card,name : see simple-audio-card.txt +- simple-audio-card,cpu : see simple-audio-card.txt +- simple-audio-card,codec : see simple-audio-card.txt + +Optional subnode properties: + +- simple-audio-card,format : see simple-audio-card.txt +- simple-audio-card,frame-master : see simple-audio-card.txt +- simple-audio-card,bitclock-master : see simple-audio-card.txt +- simple-audio-card,bitclock-inversion : see simple-audio-card.txt +- simple-audio-card,frame-inversion : see simple-audio-card.txt +- simple-audio-card,convert-rate : platform specified sampling rate convert +- simple-audio-card,convert-channels : platform specified converted channel size (2 - 8 ch) +- simple-audio-card,prefix : see routing +- 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. Valid names for sources. + use audio-prefix if some components is using same sink/sources naming. + it can be used if compatible was "renesas,rsrc-card"; + +Required CPU/CODEC subnodes properties: + +- sound-dai : see simple-audio-card.txt + +Optional CPU/CODEC subnodes properties: + +- clocks / system-clock-frequency : see simple-audio-card.txt + +Example 1. Sampling Rate Conversion + +sound { + compatible = "simple-scu-audio-card"; + + simple-audio-card,name = "rsnd-ak4643"; + simple-audio-card,format = "left_j"; + simple-audio-card,bitclock-master = <&sndcodec>; + simple-audio-card,frame-master = <&sndcodec>; + + simple-audio-card,convert-rate = <48000>; + + simple-audio-card,prefix = "ak4642"; + simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback", + "DAI0 Capture", "ak4642 Capture"; + + sndcpu: simple-audio-card,cpu { + sound-dai = <&rcar_sound>; + }; + + sndcodec: simple-audio-card,codec { + sound-dai = <&ak4643>; + system-clock-frequency = <11289600>; + }; +}; + +Example 2. 2 CPU 1 Codec (Mixing) + +sound { + compatible = "simple-scu-audio-card"; + + simple-audio-card,name = "rsnd-ak4643"; + simple-audio-card,format = "left_j"; + simple-audio-card,bitclock-master = <&dpcmcpu>; + simple-audio-card,frame-master = <&dpcmcpu>; + + simple-audio-card,prefix = "ak4642"; + simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback", + "ak4642 Playback", "DAI1 Playback"; + + dpcmcpu: cpu@0 { + sound-dai = <&rcar_sound 0>; + }; + + cpu@1 { + sound-dai = <&rcar_sound 1>; + }; + + codec { + sound-dai = <&ak4643>; + clocks = <&audio_clock>; + }; +}; diff --git a/bindings/sound/sirf-audio-codec.txt b/bindings/sound/sirf-audio-codec.txt new file mode 100644 index 00000000..062f5ec3 --- /dev/null +++ b/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/bindings/sound/sirf-audio-port.txt b/bindings/sound/sirf-audio-port.txt new file mode 100644 index 00000000..1f66de3c --- /dev/null +++ b/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/bindings/sound/sirf-audio.txt b/bindings/sound/sirf-audio.txt new file mode 100644 index 00000000..c88882ca --- /dev/null +++ b/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/bindings/sound/sirf-usp.txt b/bindings/sound/sirf-usp.txt new file mode 100644 index 00000000..02f85b32 --- /dev/null +++ b/bindings/sound/sirf-usp.txt @@ -0,0 +1,27 @@ +* SiRF SoC USP module + +Required properties: +- compatible: "sirf,prima2-usp-pcm" +- 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"). + +- clocks: USP controller clock source +- pinctrl-names: Must contain a "default" entry. +- pinctrl-NNN: One property must exist for each entry in pinctrl-names. + +Example: +usp0: usp@b0080000 { + compatible = "sirf,prima2-usp-pcm"; + reg = <0xb0080000 0x10000>; + clocks = <&clks 28>; + dmas = <&dmac1 1>, <&dmac1 2>; + dma-names = "rx", "tx"; + pinctrl-names = "default"; + pinctrl-0 = <&usp0_only_utfs_pins_a>; +}; + diff --git a/bindings/sound/snow.txt b/bindings/sound/snow.txt new file mode 100644 index 00000000..80fd9a87 --- /dev/null +++ b/bindings/sound/snow.txt @@ -0,0 +1,31 @@ +Audio Binding for Snow boards + +Required properties: +- compatible : Can be one of the following, + "google,snow-audio-max98090" or + "google,snow-audio-max98091" or + "google,snow-audio-max98095" +- samsung,i2s-controller (deprecated): The phandle of the Samsung I2S controller +- samsung,audio-codec (deprecated): The phandle of the audio codec + +Required sub-nodes: + + - 'cpu' subnode with a 'sound-dai' property containing the phandle of the I2S + controller + - 'codec' subnode with a 'sound-dai' property containing list of phandles + to the CODEC nodes, first entry must be the phandle of the MAX98090, + MAX98091 or MAX98095 CODEC (exact device type is indicated by the compatible + string) and the second entry must be the phandle of the HDMI IP block node + +Optional: +- samsung,model: The name of the sound-card + +Example: + +sound { + compatible = "google,snow-audio-max98095"; + + samsung,model = "Snow-I2S-MAX98095"; + samsung,i2s-controller = <&i2s0>; + samsung,audio-codec = <&max98095>; +}; diff --git a/bindings/sound/soc-ac97link.txt b/bindings/sound/soc-ac97link.txt new file mode 100644 index 00000000..80152a87 --- /dev/null +++ b/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/bindings/sound/spdif-receiver.txt b/bindings/sound/spdif-receiver.txt new file mode 100644 index 00000000..80f807bf --- /dev/null +++ b/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/bindings/sound/spdif-transmitter.txt b/bindings/sound/spdif-transmitter.txt new file mode 100644 index 00000000..55a85841 --- /dev/null +++ b/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/bindings/sound/ssm2518.txt b/bindings/sound/ssm2518.txt new file mode 100644 index 00000000..59381a77 --- /dev/null +++ b/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/bindings/sound/ssm4567.txt b/bindings/sound/ssm4567.txt new file mode 100644 index 00000000..ec3d9e70 --- /dev/null +++ b/bindings/sound/ssm4567.txt @@ -0,0 +1,15 @@ +Analog Devices SSM4567 audio amplifier + +This device supports I2C only. + +Required properties: + - compatible : Must be "adi,ssm4567" + - reg : the I2C address of the device. This will either be 0x34 (LR_SEL/ADDR connected to AGND), + 0x35 (LR_SEL/ADDR connected to IOVDD) or 0x36 (LR_SEL/ADDR open). + +Example: + + ssm4567: ssm4567@34 { + compatible = "adi,ssm4567"; + reg = <0x34>; + }; diff --git a/bindings/sound/st,sta32x.txt b/bindings/sound/st,sta32x.txt new file mode 100644 index 00000000..255de3ae --- /dev/null +++ b/bindings/sound/st,sta32x.txt @@ -0,0 +1,92 @@ +STA32X audio CODEC + +The driver for this device only supports I2C. + +Required properties: + + - compatible: "st,sta32x" + - reg: the I2C address of the device for I2C + - reset-gpios: a GPIO spec for the reset pin. If specified, it will be + deasserted before communication to the codec starts. + + - power-down-gpios: a GPIO spec for the power down pin. If specified, + it will be deasserted before communication to the codec + starts. + + - Vdda-supply: regulator spec, providing 3.3V + - Vdd3-supply: regulator spec, providing 3.3V + - Vcc-supply: regulator spec, providing 5V - 26V + +Optional properties: + + - st,output-conf: number, Selects the output configuration: + 0: 2-channel (full-bridge) power, 2-channel data-out + 1: 2 (half-bridge). 1 (full-bridge) on-board power + 2: 2 Channel (Full-Bridge) Power, 1 Channel FFX + 3: 1 Channel Mono-Parallel + If parameter is missing, mode 0 will be enabled. + This property has to be specified as '/bits/ 8' value. + + - st,ch1-output-mapping: Channel 1 output mapping + - st,ch2-output-mapping: Channel 2 output mapping + - st,ch3-output-mapping: Channel 3 output mapping + 0: Channel 1 + 1: Channel 2 + 2: Channel 3 + If parameter is missing, channel 1 is chosen. + This properties have to be specified as '/bits/ 8' values. + + - st,thermal-warning-recover: + If present, thermal warning recovery is enabled. + + - st,thermal-warning-adjustment: + If present, thermal warning adjustment is enabled. + + - st,fault-detect-recovery: + If present, then fault recovery will be enabled. + + - st,drop-compensation-ns: number + Only required for "st,ffx-power-output-mode" == + "variable-drop-compensation". + Specifies the drop compensation in nanoseconds. + The value must be in the range of 0..300, and only + multiples of 20 are allowed. Default is 140ns. + + - st,max-power-use-mpcc: + If present, then MPCC bits are used for MPC coefficients, + otherwise standard MPC coefficients are used. + + - st,max-power-corr: + If present, power bridge correction for THD reduction near maximum + power output is enabled. + + - st,am-reduction-mode: + If present, FFX mode runs in AM reduction mode, otherwise normal + FFX mode is used. + + - st,odd-pwm-speed-mode: + If present, PWM speed mode run on odd speed mode (341.3 kHz) on all + channels. If not present, normal PWM spped mode (384 kHz) will be used. + + - st,invalid-input-detect-mute: + If present, automatic invalid input detect mute is enabled. + +Example: + +codec: sta32x@38 { + compatible = "st,sta32x"; + reg = <0x1c>; + reset-gpios = <&gpio1 19 0>; + power-down-gpios = <&gpio1 16 0>; + st,output-conf = /bits/ 8 <0x3>; // set output to 2-channel + // (full-bridge) power, + // 2-channel data-out + st,ch1-output-mapping = /bits/ 8 <0>; // set channel 1 output ch 1 + st,ch2-output-mapping = /bits/ 8 <0>; // set channel 2 output ch 1 + st,ch3-output-mapping = /bits/ 8 <0>; // set channel 3 output ch 1 + st,max-power-correction; // enables power bridge + // correction for THD reduction + // near maximum power output + st,invalid-input-detect-mute; // mute if no valid digital + // audio signal is provided. +}; diff --git a/bindings/sound/st,sta350.txt b/bindings/sound/st,sta350.txt new file mode 100644 index 00000000..307398ef --- /dev/null +++ b/bindings/sound/st,sta350.txt @@ -0,0 +1,131 @@ +STA350 audio CODEC + +The driver for this device only supports I2C. + +Required properties: + + - compatible: "st,sta350" + - reg: the I2C address of the device for I2C + - reset-gpios: a GPIO spec for the reset pin. If specified, it will be + deasserted before communication to the codec starts. + + - power-down-gpios: a GPIO spec for the power down pin. If specified, + it will be deasserted before communication to the codec + starts. + + - vdd-dig-supply: regulator spec, providing 3.3V + - vdd-pll-supply: regulator spec, providing 3.3V + - vcc-supply: regulator spec, providing 5V - 26V + +Optional properties: + + - st,output-conf: number, Selects the output configuration: + 0: 2-channel (full-bridge) power, 2-channel data-out + 1: 2 (half-bridge). 1 (full-bridge) on-board power + 2: 2 Channel (Full-Bridge) Power, 1 Channel FFX + 3: 1 Channel Mono-Parallel + If parameter is missing, mode 0 will be enabled. + This property has to be specified as '/bits/ 8' value. + + - st,ch1-output-mapping: Channel 1 output mapping + - st,ch2-output-mapping: Channel 2 output mapping + - st,ch3-output-mapping: Channel 3 output mapping + 0: Channel 1 + 1: Channel 2 + 2: Channel 3 + If parameter is missing, channel 1 is chosen. + This properties have to be specified as '/bits/ 8' values. + + - st,thermal-warning-recover: + If present, thermal warning recovery is enabled. + + - st,thermal-warning-adjustment: + If present, thermal warning adjustment is enabled. + + - st,fault-detect-recovery: + If present, then fault recovery will be enabled. + + - st,ffx-power-output-mode: string + The FFX power output mode selects how the FFX output timing is + configured. Must be one of these values: + - "drop-compensation" + - "tapered-compensation" + - "full-power-mode" + - "variable-drop-compensation" (default) + + - st,drop-compensation-ns: number + Only required for "st,ffx-power-output-mode" == + "variable-drop-compensation". + Specifies the drop compensation in nanoseconds. + The value must be in the range of 0..300, and only + multiples of 20 are allowed. Default is 140ns. + + - st,overcurrent-warning-adjustment: + If present, overcurrent warning adjustment is enabled. + + - st,max-power-use-mpcc: + If present, then MPCC bits are used for MPC coefficients, + otherwise standard MPC coefficients are used. + + - st,max-power-corr: + If present, power bridge correction for THD reduction near maximum + power output is enabled. + + - st,am-reduction-mode: + If present, FFX mode runs in AM reduction mode, otherwise normal + FFX mode is used. + + - st,odd-pwm-speed-mode: + If present, PWM speed mode run on odd speed mode (341.3 kHz) on all + channels. If not present, normal PWM spped mode (384 kHz) will be used. + + - st,distortion-compensation: + If present, distortion compensation variable uses DCC coefficient. + If not present, preset DC coefficient is used. + + - st,invalid-input-detect-mute: + If present, automatic invalid input detect mute is enabled. + + - st,activate-mute-output: + If present, a mute output will be activated in ase the volume will + reach a value lower than -76 dBFS. + + - st,bridge-immediate-off: + If present, the bridge will be switched off immediately after the + power-down-gpio goes low. Otherwise, the bridge will wait for 13 + million clock cycles to pass before shutting down. + + - st,noise-shape-dc-cut: + If present, the noise-shaping technique on the DC cutoff filter are + enabled. + + - st,powerdown-master-volume: + If present, the power-down pin and I2C power-down functions will + act on the master volume. Otherwise, the functions will act on the + mute commands. + + - st,powerdown-delay-divider: + If present, the bridge power-down time will be divided by the provided + value. If not specified, a divider of 1 will be used. Allowed values + are 1, 2, 4, 8, 16, 32, 64 and 128. + This property has to be specified as '/bits/ 8' value. + +Example: + +codec: sta350@38 { + compatible = "st,sta350"; + reg = <0x1c>; + reset-gpios = <&gpio1 19 0>; + power-down-gpios = <&gpio1 16 0>; + st,output-conf = /bits/ 8 <0x3>; // set output to 2-channel + // (full-bridge) power, + // 2-channel data-out + st,ch1-output-mapping = /bits/ 8 <0>; // set channel 1 output ch 1 + st,ch2-output-mapping = /bits/ 8 <0>; // set channel 2 output ch 1 + st,ch3-output-mapping = /bits/ 8 <0>; // set channel 3 output ch 1 + st,max-power-correction; // enables power bridge + // correction for THD reduction + // near maximum power output + st,invalid-input-detect-mute; // mute if no valid digital + // audio signal is provided. +}; diff --git a/bindings/sound/st,sti-asoc-card.txt b/bindings/sound/st,sti-asoc-card.txt new file mode 100644 index 00000000..4d51f3f5 --- /dev/null +++ b/bindings/sound/st,sti-asoc-card.txt @@ -0,0 +1,164 @@ +STMicroelectronics sti ASoC cards + +The sti ASoC Sound Card can be used, for all sti SoCs using internal sti-sas +codec or external codecs. + +sti sound drivers allows to expose sti SoC audio interface through the +generic ASoC simple card. For details about sound card declaration please refer to +Documentation/devicetree/bindings/sound/simple-card.txt. + +1) sti-uniperiph-dai: audio dai device. +--------------------------------------- + +Required properties: + - compatible: "st,stih407-uni-player-hdmi", "st,stih407-uni-player-pcm-out", + "st,stih407-uni-player-dac", "st,stih407-uni-player-spdif", + "st,stih407-uni-reader-pcm_in", "st,stih407-uni-reader-hdmi", + + - st,syscfg: phandle to boot-device system configuration registers + + - clock-names: name of the clocks listed in clocks property in the same order + + - reg: CPU DAI IP Base address and size entries, listed in same + order than the CPU_DAI properties. + + - reg-names: names of the mapped memory regions listed in regs property in + the same order. + + - interrupts: CPU_DAI interrupt line, listed in the same order than the + CPU_DAI properties. + + - dma: CPU_DAI DMA controller phandle and DMA request line, listed in the same + order than the CPU_DAI properties. + + - dma-names: identifier string for each DMA request line in the dmas property. + "tx" for "st,sti-uni-player" compatibility + "rx" for "st,sti-uni-reader" compatibility + +Required properties ("st,sti-uni-player" compatibility only): + - clocks: CPU_DAI IP clock source, listed in the same order than the + CPU_DAI properties. + +Optional properties: + - pinctrl-0: defined for CPU_DAI@1 and CPU_DAI@4 to describe I2S PIOs for + external codecs connection. + + - pinctrl-names: should contain only one value - "default". + + - st,tdm-mode: to declare to set TDM mode for unireader and uniplayer IPs. + Only compartible with IPs in charge of the external I2S/TDM bus. + Should be declared depending on associated codec. + +Example: + + sti_uni_player1: sti-uni-player@8d81000 { + compatible = "st,stih407-uni-player-hdmi"; + #sound-dai-cells = <0>; + st,syscfg = <&syscfg_core>; + clocks = <&clk_s_d0_flexgen CLK_PCM_1>; + reg = <0x8D81000 0x158>; + interrupts = <GIC_SPI 85 IRQ_TYPE_NONE>; + dmas = <&fdma0 3 0 1>; + dma-names = "tx"; + st,tdm-mode = <1>; + }; + + sti_uni_player2: sti-uni-player@8d82000 { + compatible = "st,stih407-uni-player-pcm-out"; + #sound-dai-cells = <0>; + st,syscfg = <&syscfg_core>; + clocks = <&clk_s_d0_flexgen CLK_PCM_2>; + reg = <0x8D82000 0x158>; + interrupts = <GIC_SPI 86 IRQ_TYPE_NONE>; + dmas = <&fdma0 4 0 1>; + dma-names = "tx"; + }; + + sti_uni_player3: sti-uni-player@8d85000 { + compatible = "st,stih407-uni-player-spdif"; + #sound-dai-cells = <0>; + st,syscfg = <&syscfg_core>; + clocks = <&clk_s_d0_flexgen CLK_SPDIFF>; + reg = <0x8D85000 0x158>; + interrupts = <GIC_SPI 89 IRQ_TYPE_NONE>; + dmas = <&fdma0 7 0 1>; + dma-names = "tx"; + }; + + sti_uni_reader1: sti-uni-reader@8d84000 { + compatible = "st,stih407-uni-reader-hdmi"; + #sound-dai-cells = <0>; + st,syscfg = <&syscfg_core>; + reg = <0x8D84000 0x158>; + interrupts = <GIC_SPI 88 IRQ_TYPE_NONE>; + dmas = <&fdma0 6 0 1>; + dma-names = "rx"; + }; + +2) sti-sas-codec: internal audio codec IPs driver +------------------------------------------------- + +Required properties: + - compatible: "st,sti<chip>-sas-codec" . + Should be chip "st,stih416-sas-codec" or "st,stih407-sas-codec" + + - st,syscfg: phandle to boot-device system configuration registers. + + - pinctrl-0: SPDIF PIO description. + + - pinctrl-names: should contain only one value - "default". + +Example: + sti_sas_codec: sti-sas-codec { + compatible = "st,stih407-sas-codec"; + #sound-dai-cells = <1>; + st,reg_audio = <&syscfg_core>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_spdif_out >; + }; + +Example of audio card declaration: + sound { + compatible = "simple-audio-card"; + simple-audio-card,name = "sti audio card"; + + simple-audio-card,dai-link@0 { + /* DAC */ + format = "i2s"; + dai-tdm-slot-width = <32>; + cpu { + sound-dai = <&sti_uni_player2>; + }; + + codec { + sound-dai = <&sti_sasg_codec 1>; + }; + }; + simple-audio-card,dai-link@1 { + /* SPDIF */ + format = "left_j"; + cpu { + sound-dai = <&sti_uni_player3>; + }; + + codec { + sound-dai = <&sti_sasg_codec 0>; + }; + }; + simple-audio-card,dai-link@2 { + /* TDM playback */ + format = "left_j"; + frame-inversion = <1>; + cpu { + sound-dai = <&sti_uni_player1>; + dai-tdm-slot-num = <16>; + dai-tdm-slot-width = <16>; + dai-tdm-slot-tx-mask = + <1 1 1 1 0 0 0 0 0 0 1 1 0 0 1 1>; + }; + + codec { + sound-dai = <&sti_sasg_codec 3>; + }; + }; + }; diff --git a/bindings/sound/st,stm32-adfsdm.txt b/bindings/sound/st,stm32-adfsdm.txt new file mode 100644 index 00000000..864f5b00 --- /dev/null +++ b/bindings/sound/st,stm32-adfsdm.txt @@ -0,0 +1,63 @@ +STMicroelectronics Audio Digital Filter Sigma Delta modulators(DFSDM) + +The DFSDM allows PDM microphones capture through SPI interface. The Audio +interface is seems as a sub block of the DFSDM device. +For details on DFSDM bindings refer to ../iio/adc/st,stm32-dfsdm-adc.txt + +Required properties: + - compatible: "st,stm32h7-dfsdm-dai". + + - #sound-dai-cells : Must be equal to 0 + + - io-channels : phandle to iio dfsdm instance node. + +Example of a sound card using audio DFSDM node. + + sound_card { + compatible = "audio-graph-card"; + + dais = <&cpu_port>; + }; + + dfsdm: dfsdm@40017000 { + compatible = "st,stm32h7-dfsdm"; + reg = <0x40017000 0x400>; + clocks = <&rcc DFSDM1_CK>; + clock-names = "dfsdm"; + #interrupt-cells = <1>; + #address-cells = <1>; + #size-cells = <0>; + + dfsdm_adc0: filter@0 { + compatible = "st,stm32-dfsdm-dmic"; + reg = <0>; + interrupts = <110>; + dmas = <&dmamux1 101 0x400 0x00>; + dma-names = "rx"; + st,adc-channels = <1>; + st,adc-channel-names = "dmic0"; + st,adc-channel-types = "SPI_R"; + st,adc-channel-clk-src = "CLKOUT"; + st,filter-order = <5>; + + dfsdm_dai0: dfsdm-dai { + compatible = "st,stm32h7-dfsdm-dai"; + #sound-dai-cells = <0>; + io-channels = <&dfsdm_adc0 0>; + cpu_port: port { + dfsdm_endpoint: endpoint { + remote-endpoint = <&dmic0_endpoint>; + }; + }; + }; + }; + + dmic0: dmic@0 { + compatible = "dmic-codec"; + #sound-dai-cells = <0>; + port { + dmic0_endpoint: endpoint { + remote-endpoint = <&dfsdm_endpoint>; + }; + }; + }; diff --git a/bindings/sound/st,stm32-i2s.txt b/bindings/sound/st,stm32-i2s.txt new file mode 100644 index 00000000..58c34130 --- /dev/null +++ b/bindings/sound/st,stm32-i2s.txt @@ -0,0 +1,62 @@ +STMicroelectronics STM32 SPI/I2S Controller + +The SPI/I2S block supports I2S/PCM protocols when configured on I2S mode. +Only some SPI instances support I2S. + +Required properties: + - compatible: Must be "st,stm32h7-i2s" + - reg: Offset and length of the device's register set. + - interrupts: Must contain the interrupt line id. + - clocks: Must contain phandle and clock specifier pairs for each entry + in clock-names. + - clock-names: Must contain "i2sclk", "pclk", "x8k" and "x11k". + "i2sclk": clock which feeds the internal clock generator + "pclk": clock which feeds the peripheral bus interface + "x8k": I2S parent clock for sampling rates multiple of 8kHz. + "x11k": I2S parent clock for sampling rates multiple of 11.025kHz. + - dmas: DMA specifiers for tx and rx dma. + See Documentation/devicetree/bindings/dma/stm32-dma.txt. + - dma-names: Identifier for each DMA request line. Must be "tx" and "rx". + - pinctrl-names: should contain only value "default" + - pinctrl-0: see Documentation/devicetree/bindings/pinctrl/st,stm32-pinctrl.txt + +Optional properties: + - resets: Reference to a reset controller asserting the reset controller + +The device node should contain one 'port' child node with one child 'endpoint' +node, according to the bindings defined in Documentation/devicetree/bindings/ +graph.txt. + +Example: +sound_card { + compatible = "audio-graph-card"; + dais = <&i2s2_port>; +}; + +i2s2: audio-controller@40003800 { + compatible = "st,stm32h7-i2s"; + reg = <0x40003800 0x400>; + interrupts = <36>; + clocks = <&rcc PCLK1>, <&rcc SPI2_CK>, <&rcc PLL1_Q>, <&rcc PLL2_P>; + clock-names = "pclk", "i2sclk", "x8k", "x11k"; + dmas = <&dmamux2 2 39 0x400 0x1>, + <&dmamux2 3 40 0x400 0x1>; + dma-names = "rx", "tx"; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_i2s2>; + + i2s2_port: port@0 { + cpu_endpoint: endpoint { + remote-endpoint = <&codec_endpoint>; + format = "i2s"; + }; + }; +}; + +audio-codec { + codec_port: port@0 { + codec_endpoint: endpoint { + remote-endpoint = <&cpu_endpoint>; + }; + }; +}; diff --git a/bindings/sound/st,stm32-sai.txt b/bindings/sound/st,stm32-sai.txt new file mode 100644 index 00000000..3a3fc506 --- /dev/null +++ b/bindings/sound/st,stm32-sai.txt @@ -0,0 +1,100 @@ +STMicroelectronics STM32 Serial Audio Interface (SAI). + +The SAI interface (Serial Audio Interface) offers a wide set of audio protocols +as I2S standards, LSB or MSB-justified, PCM/DSP, TDM, and AC'97. +The SAI contains two independent audio sub-blocks. Each sub-block has +its own clock generator and I/O lines controller. + +Required properties: + - compatible: Should be "st,stm32f4-sai" or "st,stm32h7-sai" + - reg: Base address and size of SAI common register set. + - clocks: Must contain phandle and clock specifier pairs for each entry + in clock-names. + - clock-names: Must contain "pclk" "x8k" and "x11k" + "pclk": Clock which feeds the peripheral bus interface. + Mandatory for "st,stm32h7-sai" compatible. + Not used for "st,stm32f4-sai" compatible. + "x8k": SAI parent clock for sampling rates multiple of 8kHz. + "x11k": SAI parent clock for sampling rates multiple of 11.025kHz. + - interrupts: cpu DAI interrupt line shared by SAI sub-blocks + +Optional properties: + - resets: Reference to a reset controller asserting the SAI + +SAI subnodes: +Two subnodes corresponding to SAI sub-block instances A et B can be defined. +Subnode can be omitted for unsused sub-block. + +SAI subnodes required properties: + - compatible: Should be "st,stm32-sai-sub-a" or "st,stm32-sai-sub-b" + for SAI sub-block A or B respectively. + - reg: Base address and size of SAI sub-block register set. + - clocks: Must contain one phandle and clock specifier pair + for sai_ck which feeds the internal clock generator. + - clock-names: Must contain "sai_ck". + - dmas: see Documentation/devicetree/bindings/dma/stm32-dma.txt + - dma-names: identifier string for each DMA request line + "tx": if sai sub-block is configured as playback DAI + "rx": if sai sub-block is configured as capture DAI + - pinctrl-names: should contain only value "default" + - pinctrl-0: see Documentation/devicetree/bindings/pinctrl/st,stm32-pinctrl.txt + +SAI subnodes Optional properties: + - st,sync: specify synchronization mode. + By default SAI sub-block is in asynchronous mode. + This property sets SAI sub-block as slave of another SAI sub-block. + Must contain the phandle and index of the sai sub-block providing + the synchronization. + - st,iec60958: support S/PDIF IEC6958 protocol for playback + IEC60958 protocol is not available for capture. + By default, custom protocol is assumed, meaning that protocol is + configured according to protocol defined in related DAI link node, + such as i2s, left justified, right justified, dsp and pdm protocols. + Note: ac97 protocol is not supported by SAI driver + +The device node should contain one 'port' child node with one child 'endpoint' +node, according to the bindings defined in Documentation/devicetree/bindings/ +graph.txt. + +Example: +sound_card { + compatible = "audio-graph-card"; + dais = <&sai1b_port>; +}; + +sai1: sai1@40015800 { + compatible = "st,stm32h7-sai"; + #address-cells = <1>; + #size-cells = <1>; + ranges = <0 0x40015800 0x400>; + reg = <0x40015800 0x4>; + clocks = <&rcc SAI1_CK>, <&rcc PLL1_Q>, <&rcc PLL2_P>; + clock-names = "pclk", "x8k", "x11k"; + interrupts = <87>; + + sai1a: audio-controller@40015804 { + compatible = "st,stm32-sai-sub-a"; + reg = <0x4 0x1C>; + clocks = <&rcc SAI1_CK>; + clock-names = "sai_ck"; + dmas = <&dmamux1 1 87 0x400 0x0>; + dma-names = "tx"; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_sai1a>; + + sai1b_port: port { + cpu_endpoint: endpoint { + remote-endpoint = <&codec_endpoint>; + format = "i2s"; + }; + }; + }; +}; + +audio-codec { + codec_port: port { + codec_endpoint: endpoint { + remote-endpoint = <&cpu_endpoint>; + }; + }; +}; diff --git a/bindings/sound/st,stm32-spdifrx.txt b/bindings/sound/st,stm32-spdifrx.txt new file mode 100644 index 00000000..33826f24 --- /dev/null +++ b/bindings/sound/st,stm32-spdifrx.txt @@ -0,0 +1,56 @@ +STMicroelectronics STM32 S/PDIF receiver (SPDIFRX). + +The SPDIFRX peripheral, is designed to receive an S/PDIF flow compliant with +IEC-60958 and IEC-61937. + +Required properties: + - compatible: should be "st,stm32h7-spdifrx" + - reg: cpu DAI IP base address and size + - clocks: must contain an entry for kclk (used as S/PDIF signal reference) + - clock-names: must contain "kclk" + - interrupts: cpu DAI interrupt line + - dmas: DMA specifiers for audio data DMA and iec control flow DMA + See STM32 DMA bindings, Documentation/devicetree/bindings/dma/stm32-dma.txt + - dma-names: two dmas have to be defined, "rx" and "rx-ctrl" + +Optional properties: + - resets: Reference to a reset controller asserting the SPDIFRX + +The device node should contain one 'port' child node with one child 'endpoint' +node, according to the bindings defined in Documentation/devicetree/bindings/ +graph.txt. + +Example: +spdifrx: spdifrx@40004000 { + compatible = "st,stm32h7-spdifrx"; + reg = <0x40004000 0x400>; + clocks = <&rcc SPDIFRX_CK>; + clock-names = "kclk"; + interrupts = <97>; + dmas = <&dmamux1 2 93 0x400 0x0>, + <&dmamux1 3 94 0x400 0x0>; + dma-names = "rx", "rx-ctrl"; + pinctrl-0 = <&spdifrx_pins>; + pinctrl-names = "default"; + + spdifrx_port: port { + cpu_endpoint: endpoint { + remote-endpoint = <&codec_endpoint>; + }; + }; +}; + +spdif_in: spdif-in { + compatible = "linux,spdif-dir"; + + codec_port: port { + codec_endpoint: endpoint { + remote-endpoint = <&cpu_endpoint>; + }; + }; +}; + +soundcard { + compatible = "audio-graph-card"; + dais = <&spdifrx_port>; +}; diff --git a/bindings/sound/storm.txt b/bindings/sound/storm.txt new file mode 100644 index 00000000..062a4c18 --- /dev/null +++ b/bindings/sound/storm.txt @@ -0,0 +1,23 @@ +* Sound complex for Storm boards + +Models a soundcard for Storm boards with the Qualcomm Technologies IPQ806x SOC +connected to a MAX98357A DAC via I2S. + +Required properties: + +- compatible : "google,storm-audio" +- cpu : Phandle of the CPU DAI +- codec : Phandle of the codec DAI + +Optional properties: + +- qcom,model : The user-visible name of this sound card. + +Example: + +sound { + compatible = "google,storm-audio"; + qcom,model = "ipq806x-storm"; + cpu = <&lpass_cpu>; + codec = <&max98357a>; +}; diff --git a/bindings/sound/sun4i-codec.txt b/bindings/sound/sun4i-codec.txt new file mode 100644 index 00000000..66579bbd --- /dev/null +++ b/bindings/sound/sun4i-codec.txt @@ -0,0 +1,94 @@ +* Allwinner A10 Codec + +Required properties: +- compatible: must be one of the following compatibles: + - "allwinner,sun4i-a10-codec" + - "allwinner,sun6i-a31-codec" + - "allwinner,sun7i-a20-codec" + - "allwinner,sun8i-a23-codec" + - "allwinner,sun8i-h3-codec" + - "allwinner,sun8i-v3s-codec" +- reg: must contain the registers location and length +- interrupts: must contain the codec interrupt +- dmas: DMA channels for tx and rx dma. See the DMA client binding, + Documentation/devicetree/bindings/dma/dma.txt +- dma-names: should include "tx" and "rx". +- clocks: a list of phandle + clock-specifer pairs, one for each entry + in clock-names. +- clock-names: should contain the following: + - "apb": the parent APB clock for this controller + - "codec": the parent module clock + +Optional properties: +- allwinner,pa-gpios: gpio to enable external amplifier + +Required properties for the following compatibles: + - "allwinner,sun6i-a31-codec" + - "allwinner,sun8i-a23-codec" + - "allwinner,sun8i-h3-codec" + - "allwinner,sun8i-v3s-codec" +- resets: phandle to the reset control for this device +- allwinner,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 include: + + Audio pins on the SoC: + "HP" + "HPCOM" + "LINEIN" (not on sun8i-v3s) + "LINEOUT" (not on sun8i-a23 or sun8i-v3s) + "MIC1" + "MIC2" (not on sun8i-v3s) + "MIC3" (sun6i-a31 only) + + Microphone biases from the SoC: + "HBIAS" + "MBIAS" (not on sun8i-v3s) + + Board connectors: + "Headphone" + "Headset Mic" + "Line In" + "Line Out" + "Mic" + "Speaker" + +Required properties for the following compatibles: + - "allwinner,sun8i-a23-codec" + - "allwinner,sun8i-h3-codec" + - "allwinner,sun8i-v3s-codec" +- allwinner,codec-analog-controls: A phandle to the codec analog controls + block in the PRCM. + +Example: +codec: codec@1c22c00 { + #sound-dai-cells = <0>; + compatible = "allwinner,sun7i-a20-codec"; + reg = <0x01c22c00 0x40>; + interrupts = <0 30 4>; + clocks = <&apb0_gates 0>, <&codec_clk>; + clock-names = "apb", "codec"; + dmas = <&dma 0 19>, <&dma 0 19>; + dma-names = "rx", "tx"; +}; + +codec: codec@1c22c00 { + #sound-dai-cells = <0>; + compatible = "allwinner,sun6i-a31-codec"; + reg = <0x01c22c00 0x98>; + interrupts = <GIC_SPI 29 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&ccu CLK_APB1_CODEC>, <&ccu CLK_CODEC>; + clock-names = "apb", "codec"; + resets = <&ccu RST_APB1_CODEC>; + dmas = <&dma 15>, <&dma 15>; + dma-names = "rx", "tx"; + allwinner,audio-routing = + "Headphone", "HP", + "Speaker", "LINEOUT", + "LINEIN", "Line In", + "MIC1", "MBIAS", + "MIC1", "Mic", + "MIC2", "HBIAS", + "MIC2", "Headset Mic"; +}; diff --git a/bindings/sound/sun4i-i2s.txt b/bindings/sound/sun4i-i2s.txt new file mode 100644 index 00000000..b9d50d6c --- /dev/null +++ b/bindings/sound/sun4i-i2s.txt @@ -0,0 +1,43 @@ +* Allwinner A10 I2S controller + +The I2S bus (Inter-IC sound bus) is a serial link for digital +audio data transfer between devices in the system. + +Required properties: + +- compatible: should be one of the following: + - "allwinner,sun4i-a10-i2s" + - "allwinner,sun6i-a31-i2s" + - "allwinner,sun8i-a83t-i2s" + - "allwinner,sun8i-h3-i2s" +- reg: physical base address of the controller and length of memory mapped + region. +- interrupts: should contain the I2S interrupt. +- dmas: DMA specifiers for tx and rx dma. See the DMA client binding, + Documentation/devicetree/bindings/dma/dma.txt +- dma-names: should include "tx" and "rx". +- clocks: a list of phandle + clock-specifer pairs, one for each entry in clock-names. +- clock-names: should contain the following: + - "apb" : clock for the I2S bus interface + - "mod" : module clock for the I2S controller +- #sound-dai-cells : Must be equal to 0 + +Required properties for the following compatibles: + - "allwinner,sun6i-a31-i2s" + - "allwinner,sun8i-a83t-i2s" + - "allwinner,sun8i-h3-i2s" +- resets: phandle to the reset line for this codec + +Example: + +i2s0: i2s@1c22400 { + #sound-dai-cells = <0>; + compatible = "allwinner,sun4i-a10-i2s"; + reg = <0x01c22400 0x400>; + interrupts = <GIC_SPI 16 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&apb0_gates 3>, <&i2s0_clk>; + clock-names = "apb", "mod"; + dmas = <&dma SUN4I_DMA_NORMAL 3>, + <&dma SUN4I_DMA_NORMAL 3>; + dma-names = "rx", "tx"; +}; diff --git a/bindings/sound/sun8i-a33-codec.txt b/bindings/sound/sun8i-a33-codec.txt new file mode 100644 index 00000000..2ca3d138 --- /dev/null +++ b/bindings/sound/sun8i-a33-codec.txt @@ -0,0 +1,63 @@ +Allwinner SUN8I audio codec +------------------------------------ + +On Sun8i-A33 SoCs, the audio is separated in different parts: + - A DAI driver. It uses the "sun4i-i2s" driver which is + documented here: + Documentation/devicetree/bindings/sound/sun4i-i2s.txt + - An analog part of the codec which is handled as PRCM registers. + See Documentation/devicetree/bindings/sound/sun8i-codec-analog.txt + - An digital part of the codec which is documented in this current + binding documentation. + - And finally, an audio card which links all the above components. + The simple-audio card will be used. + See Documentation/devicetree/bindings/sound/simple-card.txt + +This bindings documentation exposes Sun8i codec (digital part). + +Required properties: +- compatible: must be "allwinner,sun8i-a33-codec" +- reg: must contain the registers location and length +- interrupts: must contain the codec interrupt +- clocks: a list of phandle + clock-specifer pairs, one for each entry + in clock-names. +- clock-names: should contain followings: + - "bus": the parent APB clock for this controller + - "mod": the parent module clock + +Here is an example to add a sound card and the codec binding on sun8i SoCs that +are similar to A33 using simple-card: + + sound { + compatible = "simple-audio-card"; + simple-audio-card,name = "sun8i-a33-audio"; + simple-audio-card,format = "i2s"; + simple-audio-card,frame-master = <&link_codec>; + simple-audio-card,bitclock-master = <&link_codec>; + simple-audio-card,mclk-fs = <512>; + simple-audio-card,aux-devs = <&codec_analog>; + simple-audio-card,routing = + "Left DAC", "Digital Left DAC", + "Right DAC", "Digital Right DAC"; + + simple-audio-card,cpu { + sound-dai = <&dai>; + }; + + link_codec: simple-audio-card,codec { + sound-dai = <&codec>; + }; + + soc@1c00000 { + [...] + + audio-codec@1c22e00 { + #sound-dai-cells = <0>; + compatible = "allwinner,sun8i-a33-codec"; + reg = <0x01c22e00 0x400>; + interrupts = <GIC_SPI 29 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&ccu CLK_BUS_CODEC>, <&ccu CLK_AC_DIG>; + clock-names = "bus", "mod"; + }; + }; + diff --git a/bindings/sound/sun8i-codec-analog.txt b/bindings/sound/sun8i-codec-analog.txt new file mode 100644 index 00000000..07356758 --- /dev/null +++ b/bindings/sound/sun8i-codec-analog.txt @@ -0,0 +1,17 @@ +* Allwinner Codec Analog Controls + +Required properties: +- compatible: must be one of the following compatibles: + - "allwinner,sun8i-a23-codec-analog" + - "allwinner,sun8i-h3-codec-analog" + - "allwinner,sun8i-v3s-codec-analog" + +Required properties if not a sub-node of the PRCM node: +- reg: must contain the registers location and length + +Example: +prcm: prcm@1f01400 { + codec_analog: codec-analog { + compatible = "allwinner,sun8i-a23-codec-analog"; + }; +}; diff --git a/bindings/sound/sunxi,sun4i-spdif.txt b/bindings/sound/sunxi,sun4i-spdif.txt new file mode 100644 index 00000000..0c64a209 --- /dev/null +++ b/bindings/sound/sunxi,sun4i-spdif.txt @@ -0,0 +1,42 @@ +Allwinner Sony/Philips Digital Interface Format (S/PDIF) Controller + +The Allwinner S/PDIF audio block is a transceiver that allows the +processor to receive and transmit digital audio via an coaxial cable or +a fibre cable. +For now only playback is supported. + +Required properties: + + - compatible : should be one of the following: + - "allwinner,sun4i-a10-spdif": for the Allwinner A10 SoC + - "allwinner,sun6i-a31-spdif": for the Allwinner A31 SoC + - "allwinner,sun8i-h3-spdif": for the Allwinner H3 SoC + + - 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: + "apb" clock for the spdif bus. + "spdif" clock for spdif controller. + + - resets : reset specifier for the ahb reset (A31 and newer only) + +Example: + +spdif: spdif@1c21000 { + compatible = "allwinner,sun4i-a10-spdif"; + reg = <0x01c21000 0x40>; + interrupts = <13>; + clocks = <&apb0_gates 1>, <&spdif_clk>; + clock-names = "apb", "spdif"; + dmas = <&dma 0 2>, <&dma 0 2>; + dma-names = "rx", "tx"; +}; diff --git a/bindings/sound/tas2552.txt b/bindings/sound/tas2552.txt new file mode 100644 index 00000000..2d71eb05 --- /dev/null +++ b/bindings/sound/tas2552.txt @@ -0,0 +1,36 @@ +Texas Instruments - tas2552 Codec module + +The tas2552 serial control bus communicates through I2C protocols + +Required properties: + - compatible - One of: + "ti,tas2552" - TAS2552 + - reg - I2C slave address: it can be 0x40 if ADDR pin is 0 + or 0x41 if ADDR pin is 1. + - supply-*: Required supply regulators are: + "vbat" battery voltage + "iovdd" I/O Voltage + "avdd" Analog DAC Voltage + +Optional properties: + - enable-gpio - gpio pin to enable/disable the device + +tas2552 can receive its reference clock via MCLK, BCLK, IVCLKIN pin or use the +internal 1.8MHz. This CLKIN is used by the PLL. In addition to PLL, the PDM +reference clock is also selectable: PLL, IVCLKIN, BCLK or MCLK. +For system integration the dt-bindings/sound/tas2552.h header file provides +defined values to select and configure the PLL and PDM reference clocks. + +Example: + +tas2552: tas2552@41 { + compatible = "ti,tas2552"; + reg = <0x41>; + vbat-supply = <®_vbat>; + iovdd-supply = <®_iovdd>; + avdd-supply = <®_avdd>; + enable-gpio = <&gpio4 2 GPIO_ACTIVE_HIGH>; +}; + +For more product information please see the link below: +http://www.ti.com/product/TAS2552 diff --git a/bindings/sound/tas571x.txt b/bindings/sound/tas571x.txt new file mode 100644 index 00000000..7c8fd37c --- /dev/null +++ b/bindings/sound/tas571x.txt @@ -0,0 +1,48 @@ +Texas Instruments TAS5711/TAS5717/TAS5719/TAS5721 stereo power amplifiers + +The codec is controlled through an I2C interface. It also has two other +signals that can be wired up to GPIOs: reset (strongly recommended), and +powerdown (optional). + +Required properties: + +- compatible: should be one of the following: + - "ti,tas5707" + - "ti,tas5711", + - "ti,tas5717", + - "ti,tas5719", + - "ti,tas5721" +- reg: The I2C address of the device +- #sound-dai-cells: must be equal to 0 + +Optional properties: + +- reset-gpios: GPIO specifier for the TAS571x's active low reset line +- pdn-gpios: GPIO specifier for the TAS571x's active low powerdown line +- clocks: clock phandle for the MCLK input +- clock-names: should be "mclk" +- AVDD-supply: regulator phandle for the AVDD supply (all chips) +- DVDD-supply: regulator phandle for the DVDD supply (all chips) +- HPVDD-supply: regulator phandle for the HPVDD supply (5717/5719) +- PVDD_AB-supply: regulator phandle for the PVDD_AB supply (5717/5719) +- PVDD_CD-supply: regulator phandle for the PVDD_CD supply (5717/5719) +- PVDD_A-supply: regulator phandle for the PVDD_A supply (5711) +- PVDD_B-supply: regulator phandle for the PVDD_B supply (5711) +- PVDD_C-supply: regulator phandle for the PVDD_C supply (5711) +- PVDD_D-supply: regulator phandle for the PVDD_D supply (5711) +- DRVDD-supply: regulator phandle for the DRVDD supply (5721) +- PVDD-supply: regulator phandle for the PVDD supply (5721) + +Example: + + tas5717: audio-codec@2a { + compatible = "ti,tas5717"; + reg = <0x2a>; + #sound-dai-cells = <0>; + + reset-gpios = <&gpio5 1 GPIO_ACTIVE_LOW>; + pdn-gpios = <&gpio5 2 GPIO_ACTIVE_LOW>; + + clocks = <&clk_core CLK_I2S>; + clock-names = "mclk"; + }; diff --git a/bindings/sound/tas5720.txt b/bindings/sound/tas5720.txt new file mode 100644 index 00000000..7481653f --- /dev/null +++ b/bindings/sound/tas5720.txt @@ -0,0 +1,26 @@ +Texas Instruments TAS5720 Mono Audio amplifier + +The TAS5720 serial control bus communicates through the I2C protocol only. The +serial bus is also used for periodic codec fault checking/reporting during +audio playback. For more product information please see the links below: + +http://www.ti.com/product/TAS5720L +http://www.ti.com/product/TAS5720M +http://www.ti.com/product/TAS5722L + +Required properties: + +- compatible : "ti,tas5720", + "ti,tas5722" +- reg : I2C slave address +- dvdd-supply : phandle to a 3.3-V supply for the digital circuitry +- pvdd-supply : phandle to a supply used for the Class-D amp and the analog + +Example: + +tas5720: tas5720@6c { + compatible = "ti,tas5720"; + reg = <0x6c>; + dvdd-supply = <&vdd_3v3_reg>; + pvdd-supply = <&_supply_reg>; +}; diff --git a/bindings/sound/tda7419.txt b/bindings/sound/tda7419.txt new file mode 100644 index 00000000..6b85ec38 --- /dev/null +++ b/bindings/sound/tda7419.txt @@ -0,0 +1,38 @@ +TDA7419 audio processor + +This device supports I2C only. + +Required properties: + +- compatible : "st,tda7419" +- reg : the I2C address of the device. +- vdd-supply : a regulator spec for the common power supply (8-10V) + +Optional properties: + +- st,mute-gpios : a GPIO spec for the MUTE pin. + +Pins on the device (for linking into audio routes): + + * SE3L + * SE3R + * SE2L + * SE2R + * SE1L + * SE1R + * DIFFL + * DIFFR + * MIX + * OUTLF + * OUTRF + * OUTLR + * OUTRR + * OUTSW + +Example: + +ap: tda7419@44 { + compatible = "st,tda7419"; + reg = <0x44>; + vdd-supply = <&vdd_9v0_reg>; +}; diff --git a/bindings/sound/tdm-slot.txt b/bindings/sound/tdm-slot.txt new file mode 100644 index 00000000..34cf70e2 --- /dev/null +++ b/bindings/sound/tdm-slot.txt @@ -0,0 +1,29 @@ +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. +dai-tdm-slot-tx-mask : Transmit direction slot mask, optional +dai-tdm-slot-rx-mask : Receive direction slot mask, optional + +For instance: + dai-tdm-slot-num = <2>; + dai-tdm-slot-width = <8>; + dai-tdm-slot-tx-mask = <0 1>; + dai-tdm-slot-rx-mask = <1 0>; + +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. + +The explicit masks are given as array of integers, where the first +number presents bit-0 (LSB), second presents bit-1, etc. Any non zero +number is considered 1 and 0 is 0. snd_soc_of_xlate_tdm_slot_mask() +does not do anything, if either mask is set non zero value. diff --git a/bindings/sound/tfa9879.txt b/bindings/sound/tfa9879.txt new file mode 100644 index 00000000..1620e684 --- /dev/null +++ b/bindings/sound/tfa9879.txt @@ -0,0 +1,23 @@ +NXP TFA9879 class-D audio amplifier + +Required properties: + +- compatible : "nxp,tfa9879" + +- reg : the I2C address of the device + +- #sound-dai-cells : must be 0. + +Example: + +&i2c1 { + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_i2c1>; + + amp: amp@6c { + #sound-dai-cells = <0>; + compatible = "nxp,tfa9879"; + reg = <0x6c>; + }; +}; + diff --git a/bindings/sound/ti,ads117x.txt b/bindings/sound/ti,ads117x.txt new file mode 100644 index 00000000..7db19b50 --- /dev/null +++ b/bindings/sound/ti,ads117x.txt @@ -0,0 +1,11 @@ +Texas Intstruments ADS117x ADC + +Required properties: + + - compatible : "ti,ads1174" or "ti,ads1178" + +Example: + +ads1178 { + compatible = "ti,ads1178"; +}; diff --git a/bindings/sound/ti,pcm1681.txt b/bindings/sound/ti,pcm1681.txt new file mode 100644 index 00000000..4df17185 --- /dev/null +++ b/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/bindings/sound/ti,pcm3168a.txt b/bindings/sound/ti,pcm3168a.txt new file mode 100644 index 00000000..5d9cb84c --- /dev/null +++ b/bindings/sound/ti,pcm3168a.txt @@ -0,0 +1,48 @@ +Texas Instruments pcm3168a DT bindings + +This driver supports both SPI and I2C bus access for this codec + +Required properties: + + - compatible: "ti,pcm3168a" + + - clocks : Contains an entry for each entry in clock-names + + - clock-names : Includes the following entries: + "scki" The system clock + + - VDD1-supply : Digital power supply regulator 1 (+3.3V) + + - VDD2-supply : Digital power supply regulator 2 (+3.3V) + + - VCCAD1-supply : ADC power supply regulator 1 (+5V) + + - VCCAD2-supply : ADC power supply regulator 2 (+5V) + + - VCCDA1-supply : DAC power supply regulator 1 (+5V) + + - VCCDA2-supply : DAC power supply regulator 2 (+5V) + +For required properties on SPI/I2C, consult SPI/I2C device tree documentation + +Examples: + +i2c0: i2c0@0 { + + ... + + pcm3168a: audio-codec@44 { + compatible = "ti,pcm3168a"; + reg = <0x44>; + clocks = <&clk_core CLK_AUDIO>; + clock-names = "scki"; + VDD1-supply = <&supply3v3>; + VDD2-supply = <&supply3v3>; + VCCAD1-supply = <&supply5v0>; + VCCAD2-supply = <&supply5v0>; + VCCDA1-supply = <&supply5v0>; + VCCDA2-supply = <&supply5v0>; + pinctrl-names = "default"; + pinctrl-0 = <&dac_clk_pin>; + }; +}; diff --git a/bindings/sound/ti,tas5086.txt b/bindings/sound/ti,tas5086.txt new file mode 100644 index 00000000..234dad29 --- /dev/null +++ b/bindings/sound/ti,tas5086.txt @@ -0,0 +1,48 @@ +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. + + - avdd-supply: Power supply for AVDD, providing 3.3V + - dvdd-supply: Power supply for DVDD, providing 3.3V + +Examples: + + i2c_bus { + tas5086@1b { + compatible = "ti,tas5086"; + reg = <0x1b>; + reset-gpio = <&gpio 23 0>; + ti,charge-period = <156000>; + avdd-supply = <&vdd_3v3_reg>; + dvdd-supply = <&vdd_3v3_reg>; + }; + }; diff --git a/bindings/sound/ti,tas6424.txt b/bindings/sound/ti,tas6424.txt new file mode 100644 index 00000000..eacb54f3 --- /dev/null +++ b/bindings/sound/ti,tas6424.txt @@ -0,0 +1,22 @@ +Texas Instruments TAS6424 Quad-Channel Audio amplifier + +The TAS6424 serial control bus communicates through I2C protocols. + +Required properties: + - compatible: "ti,tas6424" - TAS6424 + - reg: I2C slave address + - sound-dai-cells: must be equal to 0 + - standby-gpios: GPIO used to shut the TAS6424 down. + - mute-gpios: GPIO used to mute all the outputs + +Example: + +tas6424: tas6424@6a { + compatible = "ti,tas6424"; + reg = <0x6a>; + + #sound-dai-cells = <0>; +}; + +For more product information please see the link below: +http://www.ti.com/product/TAS6424-Q1 diff --git a/bindings/sound/tlv320aic31xx.txt b/bindings/sound/tlv320aic31xx.txt new file mode 100644 index 00000000..5b3c33bb --- /dev/null +++ b/bindings/sound/tlv320aic31xx.txt @@ -0,0 +1,72 @@ +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) + "ti,tlv320dac3100" - TLV320DAC3100 (no ADC, mono speaker amp, no MiniDSP) + "ti,tlv320dac3101" - TLV320DAC3101 (no ADC, stereo speaker amp, no MiniDSP) + +- reg - <int> - I2C slave address +- 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 + + +Optional properties: + +- reset-gpios - GPIO specification for the active low RESET input. +- 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. + +Deprecated properties: + +- gpio-reset - gpio pin number used for codec reset + +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, devices with ADC + * MIC1RP, devices with ADC + * MIC1LM, devices with ADC + * AIN1, devices without ADC + * AIN2, devices without ADC + +The pins can be used in referring sound node's audio-routing property. + +Example: +#include <dt-bindings/gpio/gpio.h> +#include <dt-bindings/sound/tlv320aic31xx-micbias.h> + +tlv320aic31xx: tlv320aic31xx@18 { + compatible = "ti,tlv320aic311x"; + reg = <0x18>; + + ai31xx-micbias-vg = <MICBIAS_OFF>; + + reset-gpios = <&gpio1 17 GPIO_ACTIVE_LOW>; + + HPVDD-supply = <®ulator>; + SPRVDD-supply = <®ulator>; + SPLVDD-supply = <®ulator>; + AVDD-supply = <®ulator>; + IOVDD-supply = <®ulator>; + DVDD-supply = <®ulator>; +}; diff --git a/bindings/sound/tlv320aic32x4.txt b/bindings/sound/tlv320aic32x4.txt new file mode 100644 index 00000000..ca75890f --- /dev/null +++ b/bindings/sound/tlv320aic32x4.txt @@ -0,0 +1,41 @@ +Texas Instruments - tlv320aic32x4 Codec module + +The tlv320aic32x4 serial control bus communicates through I2C protocols + +Required properties: + - compatible - "string" - One of: + "ti,tlv320aic32x4" TLV320AIC3204 + "ti,tlv320aic32x6" TLV320AIC3206, TLV320AIC3256 + - 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. + - aic32x4-gpio-func - <array of 5 int> + - Types are defined in include/sound/tlv320aic32x4.h + + +Example: + +codec: tlv320aic32x4@18 { + compatible = "ti,tlv320aic32x4"; + reg = <0x18>; + clocks = <&clks 201>; + clock-names = "mclk"; + aic32x4-gpio-func= < + 0xff /* AIC32X4_MFPX_DEFAULT_VALUE */ + 0xff /* AIC32X4_MFPX_DEFAULT_VALUE */ + 0x04 /* MFP3 AIC32X4_MFP3_GPIO_ENABLED */ + 0xff /* AIC32X4_MFPX_DEFAULT_VALUE */ + 0x08 /* MFP5 AIC32X4_MFP5_GPIO_INPUT */ + >; +}; diff --git a/bindings/sound/tlv320aic3x.txt b/bindings/sound/tlv320aic3x.txt new file mode 100644 index 00000000..9796c463 --- /dev/null +++ b/bindings/sound/tlv320aic3x.txt @@ -0,0 +1,80 @@ +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 + "ti,tlv320aic3104" - TLV320AIC3104 + + +- reg - <int> - I2C slave address + + +Optional properties: + +- reset-gpios - GPIO specification for the active low RESET input. +- ai3x-gpio-func - <array of 2 int> - AIC3X_GPIO1 & AIC3X_GPIO2 Functionality + - Not supported on tlv320aic3104 +- 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. +- ai3x-ocmv - Output Common-Mode Voltage selection: + 0 - 1.35V, + 1 - 1.5V, + 2 - 1.65V, + 3 - 1.8V +- AVDD-supply, IOVDD-supply, DRVDD-supply, DVDD-supply : power supplies for the + device as covered in Documentation/devicetree/bindings/regulator/regulator.txt + +Deprecated properties: + +- gpio-reset - gpio pin number used for codec reset + +CODEC output pins: + * LLOUT + * RLOUT + * MONO_LOUT + * HPLOUT + * HPROUT + * HPLCOM + * HPRCOM + +CODEC input pins for TLV320AIC3104: + * MIC2L + * MIC2R + * LINE1L + * LINE1R + +CODEC input pins for other compatible codecs: + * MIC3L + * MIC3R + * LINE1L + * LINE2L + * LINE1R + * LINE2R + +The pins can be used in referring sound node's audio-routing property. + +Example: + +#include <dt-bindings/gpio/gpio.h> + +tlv320aic3x: tlv320aic3x@1b { + compatible = "ti,tlv320aic3x"; + reg = <0x1b>; + + reset-gpios = <&gpio1 17 GPIO_ACTIVE_LOW>; + + AVDD-supply = <®ulator>; + IOVDD-supply = <®ulator>; + DRVDD-supply = <®ulator>; + DVDD-supply = <®ulator>; +}; diff --git a/bindings/sound/tpa6130a2.txt b/bindings/sound/tpa6130a2.txt new file mode 100644 index 00000000..6dfa740e --- /dev/null +++ b/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/bindings/sound/ts3a227e.txt b/bindings/sound/ts3a227e.txt new file mode 100644 index 00000000..3ed83591 --- /dev/null +++ b/bindings/sound/ts3a227e.txt @@ -0,0 +1,30 @@ +Texas Instruments TS3A227E +Autonomous Audio Accessory Detection and Configuration Switch + +The TS3A227E detect headsets of 3-ring and 4-ring standards and +switches automatically to route the microphone correctly. It also +handles key press detection in accordance with the Android audio +headset specification v1.0. + +Required properties: + + - compatible: Should contain "ti,ts3a227e". + - reg: The i2c address. Should contain <0x3b>. + - interrupts: Interrupt number for /INT pin from the 227e + +Optional properies: + - ti,micbias: Intended MICBIAS voltage (datasheet section 9.6.7). + Select 0/1/2/3/4/5/6/7 to specify MACBIAS voltage + 2.1V/2.2V/2.3V/2.4V/2.5V/2.6V/2.7V/2.8V + Default value is "1" (2.2V). + +Examples: + + i2c { + ts3a227e@3b { + compatible = "ti,ts3a227e"; + reg = <0x3b>; + interrupt-parent = <&gpio>; + interrupts = <3 IRQ_TYPE_LEVEL_LOW>; + }; + }; diff --git a/bindings/sound/tscs42xx.txt b/bindings/sound/tscs42xx.txt new file mode 100644 index 00000000..7eea32e9 --- /dev/null +++ b/bindings/sound/tscs42xx.txt @@ -0,0 +1,22 @@ +TSCS42XX Audio CODEC + +Required Properties: + + - compatible : "tempo,tscs42A1" for analog mic + "tempo,tscs42A2" for digital mic + + - reg : <0x71> for analog mic + <0x69> for digital mic + + - clock-names: Must one of the following "mclk1", "xtal", "mclk2" + + - clocks: phandle of the clock that provides the codec sysclk + +Example: + +wookie: codec@69 { + compatible = "tempo,tscs42A2"; + reg = <0x69>; + clock-names = "xtal"; + clocks = <&audio_xtal>; +}; diff --git a/bindings/sound/tscs454.txt b/bindings/sound/tscs454.txt new file mode 100644 index 00000000..3ba3e2d2 --- /dev/null +++ b/bindings/sound/tscs454.txt @@ -0,0 +1,23 @@ +TSCS454 Audio CODEC + +Required Properties: + + - compatible : "tempo,tscs454" + + - reg : <0x69> + + - clock-names: Must one of the following "xtal", "mclk1", "mclk2" + + - clocks: phandle of the clock that provides the codec sysclk + + Note: If clock is not provided then bit clock is assumed + +Example: + +redwood: codec@69 { + #sound-dai-cells = <1>; + compatible = "tempo,tscs454"; + reg = <0x69>; + clock-names = "mclk1"; + clocks = <&audio_mclk>; +}; diff --git a/bindings/sound/uniphier,aio.txt b/bindings/sound/uniphier,aio.txt new file mode 100644 index 00000000..4ce68ed6 --- /dev/null +++ b/bindings/sound/uniphier,aio.txt @@ -0,0 +1,45 @@ +Socionext UniPhier SoC audio driver + +The Socionext UniPhier audio subsystem consists of I2S and S/PDIF blocks in +the same register space. + +Required properties: +- compatible : should be one of the following: + "socionext,uniphier-ld11-aio" + "socionext,uniphier-ld20-aio" + "socionext,uniphier-pxs2-aio" +- reg : offset and length of the register set for the device. +- interrupts : should contain I2S or S/PDIF interrupt. +- pinctrl-names : should be "default". +- pinctrl-0 : defined I2S signal pins for an external codec chip. +- clock-names : should include following entries: + "aio" +- clocks : a list of phandle, should contain an entry for each + entry in clock-names. +- reset-names : should include following entries: + "aio" +- resets : a list of phandle, should contain an entry for each + entry in reset-names. +- #sound-dai-cells: should be 1. + +Optional properties: +- socionext,syscon: a phandle, should contain soc-glue. + The soc-glue is used for changing mode of S/PDIF signal pin + to Output from Hi-Z. This property is optional if you use + I2S signal pins only. + +Example: + audio { + compatible = "socionext,uniphier-ld20-aio"; + reg = <0x56000000 0x80000>; + interrupts = <0 144 4>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_aout>; + clock-names = "aio"; + clocks = <&sys_clk 40>; + reset-names = "aio"; + resets = <&sys_rst 40>; + #sound-dai-cells = <1>; + + socionext,syscon = <&sg>; + }; diff --git a/bindings/sound/uniphier,evea.txt b/bindings/sound/uniphier,evea.txt new file mode 100644 index 00000000..3f31b235 --- /dev/null +++ b/bindings/sound/uniphier,evea.txt @@ -0,0 +1,26 @@ +Socionext EVEA - UniPhier SoC internal codec driver + +Required properties: +- compatible : should be "socionext,uniphier-evea". +- reg : offset and length of the register set for the device. +- clock-names : should include following entries: + "evea", "exiv" +- clocks : a list of phandle, should contain an entry for each + entries in clock-names. +- reset-names : should include following entries: + "evea", "exiv", "adamv" +- resets : a list of phandle, should contain reset entries of + reset-names. +- #sound-dai-cells: should be 1. + +Example: + + codec { + compatible = "socionext,uniphier-evea"; + reg = <0x57900000 0x1000>; + clock-names = "evea", "exiv"; + clocks = <&sys_clk 41>, <&sys_clk 42>; + reset-names = "evea", "exiv", "adamv"; + resets = <&sys_rst 41>, <&sys_rst 42>, <&adamv_rst 0>; + #sound-dai-cells = <1>; + }; diff --git a/bindings/sound/ux500-mop500.txt b/bindings/sound/ux500-mop500.txt new file mode 100644 index 00000000..48e071c9 --- /dev/null +++ b/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/bindings/sound/ux500-msp.txt b/bindings/sound/ux500-msp.txt new file mode 100644 index 00000000..7dd1b961 --- /dev/null +++ b/bindings/sound/ux500-msp.txt @@ -0,0 +1,42 @@ +* 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. + - <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/bindings/sound/wcd_codec.txt b/bindings/sound/wcd_codec.txt new file mode 100644 index 00000000..db50c09b --- /dev/null +++ b/bindings/sound/wcd_codec.txt @@ -0,0 +1,260 @@ +Qualcomm Technologies, Inc. WCD audio CODEC + +WSA macro in Bolero codec + +Required properties: + - compatible = "qcom,wsa-macro"; + - reg: Specifies the WSA macro base address for Bolero + soundwire core registers. + - clock-names : clock names defined for WSA macro + - clocks : clock handles defined for WSA macro + - qcom,default-clk-id: Default clk ID used for WSA macro + - qcom,wsa-swr-gpios: phandle for SWR data and clock GPIOs of WSA macro + - qcom,wsa-bcl-pmic-params: u8 array of PMIC ID, SID and PPID in same order + required to be configured to receive interrupts + in BCL block of WSA macro + +Example: + +&bolero { + wsa_macro: wsa-macro { + compatible = "qcom,wsa-macro"; + reg = <0x0C2C0000 0x0>; + clock-names = "wsa_core_clk", "wsa_npl_clk"; + clocks = <&clock_audio_wsa_1 0>, + <&clock_audio_wsa_2 0>; + qcom,wsa-swr-gpios = &wsa_swr_gpios; + qcom,wsa-bcl-pmic-params = /bits/ 8 <0x00 0x00 0x1E>; + qcom,default-clk-id = <TX_CORE_CLK>; + swr_0: wsa_swr_master { + compatible = "qcom,swr-mstr"; + wsa881x_1: wsa881x@20170212 { + compatible = "qcom,wsa881x"; + reg = <0x00 0x20170212>; + qcom,spkr-sd-n-gpio = <&tlmm 80 0>; + }; + }; + }; +}; + +VA macro in bolero codec + +Required properties: + - compatible = "qcom,va-macro"; + - reg: Specifies the VA macro base address for Bolero + soundwire core registers. + - clock-names : clock names defined for VA macro + - clocks : clock handles defined for VA macro + - qcom,default-clk-id: Default clk ID used for VA macro + - va-vdd-micb-supply： phandle of mic bias supply's regulator device tree node + - qcom,va-vdd-micb-voltage： mic bias supply's voltage level min and max in mV + - qcom,va-vdd-micb-current： mic bias supply's max current in mA + - qcom,va-dmic-sample-rate： Sample rate defined for DMIC connected to VA macro + +Optional properties: + - qcom,va-clk-mux-select： VA macro MCLK MUX selection + - qcom,va-island-mode-muxsel： VA macro island mode MUX selection + This property is required if qcom,va-clk-mux-select is provided + +Example: + +&bolero { + va_macro: va-macro { + compatible = "qcom,va-macro"; + reg = <0x0C490000 0x0>; + clock-names = "va_core_clk"; + clocks = <&clock_audio_va 0>; + qcom,default-clk-id = <TX_CORE_CLK>; + va-vdd-micb-supply = <&S4A>; + qcom,va-vdd-micb-voltage = <1800000 1800000>; + qcom,va-vdd-micb-current = <11200>; + qcom,va-dmic-sample-rate = <4800000>; + qcom,va-clk-mux-select = <1>; + qcom,va-island-mode-muxsel = <0x033A0000>; + }; +}; + +RX macro in bolero codec + +Required properties: + - compatible = "qcom,rx-macro"; + - reg: Specifies the Rx macro base address for Bolero + soundwire core registers. + - clock-names : clock names defined for RX macro + - clocks : clock handles defined for RX macro + - qcom,default-clk-id: Default clk ID used for RX macro + - qcom,rx-swr-gpios: phandle for SWR data and clock GPIOs of RX macro + - qcom,rx_mclk_mode_muxsel: register address for RX macro MCLK mode mux select + - qcom,rx-bcl-pmic-params: u8 array of PMIC ID, SID and PPID in same order + required to be configured to receive interrupts + in BCL block of WSA macro + +Example: + +&bolero { + rx_macro: rx-macro { + compatible = "qcom,rx-macro"; + reg = <0x62EE0000 0x0>; + clock-names = "rx_core_clk", "rx_npl_clk"; + clocks = <&clock_audio_rx_1 0>, + <&clock_audio_rx_2 0>; + qcom,rx-swr-gpios = <&rx_swr_gpios>; + qcom,rx_mclk_mode_muxsel = <0x62C25020>; + qcom,rx-bcl-pmic-params = /bits/ 8 <0x00 0x00 0x1E>; + qcom,default-clk-id = <TX_CORE_CLK>; + swr_1: rx_swr_master { + compatible = "qcom,swr-mstr"; + wcd938x_rx_slave: wcd938x-rx-slave { + compatible = "qcom,wcd938x-slave"; + }; + }; + }; +}; + +TX macro in bolero codec + +Required properties: + - compatible = "qcom,tx-macro"; + - reg: Specifies the Tx macro base address for Bolero + soundwire core registers. + - clock-names : clock names defined for TX macro + - clocks : clock handles defined for TX macro + - qcom,tx-swr-gpios: phandle for SWR data and clock GPIOs of TX macro + - qcom,tx-dmic-sample-rate: Sample rate defined for DMICs connected to TX macro + +Example: + +&bolero { + tx_macro: tx-macro { + compatible = "qcom,tx-macro"; + reg = <0x62EC0000 0x0>; + clock-names = "tx_core_clk", "tx_npl_clk"; + clocks = <&clock_audio_tx_1 0> + <&clock_audio_tx_2 0>; + qcom,tx-swr-gpios = <&tx_swr_gpios>; + qcom,tx-dmic-sample-rate = <4800000>; + swr_2: tx_swr_master { + compatible = "qcom,swr-mstr"; + wcd938x_tx_slave: wcd938x-tx-slave { + compatible = "qcom,wcd938x-slave"; + }; + }; + }; +}; + +Traverso Codec + +Required properties: + - compatible: "qcom,wcd938x-codec"; + - qcom,rx_swr_ch_map: mapping of swr rx slave port configuration to port_type and also + corresponding master port type it need to attach. + format: <port_id, slave_port_type, ch_mask, ch_rate, master_port_type> + same port_id configurations have to be grouped, and in ascending order. + - qcom,tx_swr_ch_map: mapping of swr tx slave port configuration to port_type and also + corresponding master port type it need to attach. + format: <port_id,slave_port_type, ch_mask, ch_rate, master_port_type> + same port_id configurations have to be grouped, and in ascending order. + - qcom,wcd-rst-gpio-node: Phandle reference to the DT node having codec reset gpio + configuration. If this property is not defined, it is + expected to atleast define "qcom,cdc-reset-gpio" property. + - qcom,rx-slave: phandle reference of Soundwire Rx slave device. + - qcom,tx-slave: phandle reference of Soundwire Tx slave device. + +Optional properties: + + - cdc-vdd-rxtx-supply: phandle of rxtx supply's regulator device tree node. + - qcom,cdc-vdd-rxtx-voltage: rxtx supply's voltage level min and max in mV. + - qcom,cdc-vdd-rxtx-current: rxtx supply's max current in mA. + + - cdc-vddio-supply: phandle of io supply's regulator device tree node. + - qcom,cdc-vddio-voltage: io supply's voltage level min and max in mV. + - qcom,cdc-vddio-current: io supply's max current in mA. + + - cdc-vdd-buck-supply: phandle of buck supply's regulator device tree node. + - qcom,cdc-vdd-buck-voltage: buck supply's voltage level min and max in mV. + - qcom,cdc-vdd-buck-current: buck supply's max current in mA. + + - cdc-vdd-mic-bias-supply: phandle of mic bias supply's regulator device tree node. + - qcom,cdc-vdd-mic-bias-voltage: mic bias supply's voltage level min and max in mV. + - qcom,cdc-vdd-mic-bias-current: mic bias supply's max current in mA. + + - qcom,cdc-static-supplies: List of supplies to be enabled prior to codec + hardware probe. Supplies in this list will be + stay enabled. + + - qcom,cdc-on-demand-supplies: List of supplies which can be enabled + dynamically. + Supplies in this list are off by default. + +Example: +wcd938x_codec: wcd938x-codec { + compatible = "qcom,wcd938x-codec"; + qcom,rx_swr_ch_map = <0 HPH_L 0x1 0 HPH_L>, + <0 HPH_R 0x2 0 HPH_R>, <1 CLSH 0x3 0 CLSH>, + <2 COMP_L 0x1 0 COMP_L>, <2 COMP_R 0x2 0 COMP_R>, + <3 LO 0x1 0 LO>, <4 DSD_L 0x1 0 DSD_L>, + <4 DSD_R 0x2 0 DSD_R>; + qcom,tx_swr_ch_map = <0 ADC1 0x1 0 ADC1>, + <1 ADC2 0x1 0 ADC3>, <1 ADC3 0x2 0 ADC4>, + <2 DMIC0 0x1 0 DMIC0>, <2 DMIC1 0x2 0 DMIC1>, + <2 MBHC 0x4 0 DMIC2>, <3 DMIC2 0x1 0 DMIC4>, + <3 DMIC3 0x2 0 DMIC5>, <3 DMIC4 0x4 0 DMIC6>, + <3 DMIC5 0x8 0 DMIC7>; + + qcom,wcd-rst-gpio-node = <&wcd938x_rst_gpio>; + qcom,rx-slave = <&wcd938x_rx_slave>; + qcom,tx-slave = <&wcd938x_tx_slave>; + + cdc-vdd-buck-supply = <&S4A>; + qcom,cdc-vdd-buck-voltage = <1800000 1800000>; + qcom,cdc-vdd-buck-current = <650000>; + + cdc-vdd-rxtx-supply = <&S4A>; + qcom,cdc-vdd-rxtx-voltage = <1800000 1800000>; + qcom,cdc-vdd-rxtx-current = <30000>; + + cdc-vddio-supply = <&S4A>; + qcom,cdc-vddio-voltage = <1800000 1800000>; + qcom,cdc-vddio-current = <30000>; + + cdc-vdd-mic-bias-supply = <&BOB>; + qcom,cdc-vdd-mic-bias-voltage = <3296000 3296000>; + qcom,cdc-vdd-mic-bias-current = <30000>; + + qcom,cdc-static-supplies = "cdc-vdd-rxtx", + "cdc-vddio"; + qcom,cdc-on-demand-supplies = "cdc-vdd-buck", + "cdc-vdd-mic-bias"; +}; + +Bolero Clock Resource Manager + +Required Properties: + - compatible = "qcom,bolero-clk-rsc-mngr"; + - qcom,fs-gen-sequence: Register sequence for fs clock generation + - clock-names : clock names defined for WSA macro + - clocks : clock handles defined for WSA macro + +Optional Properties: + - qcom,rx_mclk_mode_muxsel: register address for RX macro MCLK mode mux select + - qcom,wsa_mclk_mode_muxsel: register address for WSA macro MCLK mux select + - qcom,va_mclk_mode_muxsel: register address for VA macro MCLK mode mux select + +Example: +&bolero { + bolero-clock-rsc-manager { + compatible = "qcom,bolero-clk-rsc-mngr"; + qcom,fs-gen-sequence = <0x3000 0x1>, + <0x3004 0x1>, <0x3080 0x2>; + qcom,rx_mclk_mode_muxsel = <0x033240D8>; + qcom,wsa_mclk_mode_muxsel = <0x033220D8>; + qcom,va_mclk_mode_muxsel = <0x033A0000>; + clock-names = "tx_core_clk", "tx_npl_clk", "rx_core_clk", + "rx_npl_clk", "wsa_core_clk", "wsa_npl_clk", + "va_core_clk", "va_npl_clk"; + clocks = <&clock_audio_tx_1 0>, <&clock_audio_tx_2 0>, + <&clock_audio_rx_1 0>, <&clock_audio_rx_2 0>, + <&clock_audio_wsa_1 0>, <&clock_audio_wsa_2 0>, + <&clock_audio_va_1 0>, <&clock_audio_va_2 0>; + }; +}; diff --git a/bindings/sound/widgets.txt b/bindings/sound/widgets.txt new file mode 100644 index 00000000..b6de5ba3 --- /dev/null +++ b/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/bindings/sound/wlf,arizona.txt b/bindings/sound/wlf,arizona.txt new file mode 100644 index 00000000..e172c62d --- /dev/null +++ b/bindings/sound/wlf,arizona.txt @@ -0,0 +1,53 @@ +Cirrus Logic Arizona class audio SoCs + +These devices are audio SoCs with extensive digital capabilities and a range +of analogue I/O. + +This document lists sound specific bindings, see the primary binding +document: + ../mfd/arizona.txt + +Optional properties: + + - wlf,inmode : A list of INn_MODE register values, where n is the number + of input signals. Valid values are 0 (Differential), 1 (Single-ended) and + 2 (Digital Microphone). If absent, INn_MODE registers set to 0 by default. + If present, values must be specified less than or equal to the number of + input signals. If values less than the number of input signals, elements + that have not been specified are set to 0 by default. Entries are: + <IN1, IN2, IN3, IN4> (wm5102, wm5110, wm8280, wm8997) + <IN1A, IN2A, IN1B, IN2B> (wm8998, wm1814) + - wlf,out-mono : A list of boolean values indicating whether each output is + mono or stereo. Position within the list indicates the output affected + (eg. First entry in the list corresponds to output 1). A non-zero value + indicates a mono output. If present, the number of values should be less + than or equal to the number of outputs, if less values are supplied the + additional outputs will be treated as stereo. + + - wlf,dmic-ref : DMIC reference voltage source for each input, can be + selected from either MICVDD or one of the MICBIAS's, defines + (ARIZONA_DMIC_xxxx) are provided in <dt-bindings/mfd/arizona.txt>. If + present, the number of values should be less than or equal to the + number of inputs, unspecified inputs will use the chip default. + + - wlf,max-channels-clocked : The maximum number of channels to be clocked on + each AIF, useful for I2S systems with multiple data lines being mastered. + Specify one cell for each AIF to be configured, specify zero for AIFs that + should be handled normally. + If present, number of cells must be less than or equal to the number of + AIFs. If less than the number of AIFs, for cells that have not been + specified the corresponding AIFs will be treated as default setting. + + - wlf,spk-fmt : PDM speaker data format, must contain 2 cells (OUT5 and OUT6). + See the datasheet for values. + The second cell is ignored for codecs that do not have OUT6 (wm5102, wm8997, + wm8998, wm1814) + + - wlf,spk-mute : PDM speaker mute setting, must contain 2 cells (OUT5 and OUT6). + See the datasheet for values. + The second cell is ignored for codecs that do not have OUT6 (wm5102, wm8997, + wm8998, wm1814) + + - wlf,out-volume-limit : The volume limit value that should be applied to each + output channel. See the datasheet for exact values. Channels are specified + in the order OUT1L, OUT1R, OUT2L, OUT2R, etc. diff --git a/bindings/sound/wlf,wm8974.txt b/bindings/sound/wlf,wm8974.txt new file mode 100644 index 00000000..01d3a7c8 --- /dev/null +++ b/bindings/sound/wlf,wm8974.txt @@ -0,0 +1,15 @@ +WM8974 audio CODEC + +This device supports both I2C and SPI (configured with pin strapping +on the board). + +Required properties: + - compatible: "wlf,wm8974" + - reg: the I2C address or SPI chip select number of the device + +Examples: + +codec: wm8974@1a { + compatible = "wlf,wm8974"; + reg = <0x1a>; +}; diff --git a/bindings/sound/wm8510.txt b/bindings/sound/wm8510.txt new file mode 100644 index 00000000..e6b6cc04 --- /dev/null +++ b/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: + +wm8510: codec@1a { + compatible = "wlf,wm8510"; + reg = <0x1a>; +}; diff --git a/bindings/sound/wm8523.txt b/bindings/sound/wm8523.txt new file mode 100644 index 00000000..f3a6485f --- /dev/null +++ b/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: + +wm8523: codec@1a { + compatible = "wlf,wm8523"; + reg = <0x1a>; +}; diff --git a/bindings/sound/wm8524.txt b/bindings/sound/wm8524.txt new file mode 100644 index 00000000..f6c0c263 --- /dev/null +++ b/bindings/sound/wm8524.txt @@ -0,0 +1,16 @@ +WM8524 audio CODEC + +This device does not use I2C or SPI but a simple Hardware Control Interface. + +Required properties: + + - compatible : "wlf,wm8524" + + - wlf,mute-gpios: a GPIO spec for the MUTE pin. + +Example: + +wm8524: codec { + compatible = "wlf,wm8524"; + wlf,mute-gpios = <&gpio1 8 GPIO_ACTIVE_LOW>; +}; diff --git a/bindings/sound/wm8580.txt b/bindings/sound/wm8580.txt new file mode 100644 index 00000000..ff3f9f5f --- /dev/null +++ b/bindings/sound/wm8580.txt @@ -0,0 +1,16 @@ +WM8580 and WM8581 audio CODEC + +This device supports I2C only. + +Required properties: + + - compatible : "wlf,wm8580", "wlf,wm8581" + + - reg : the I2C address of the device. + +Example: + +wm8580: codec@1a { + compatible = "wlf,wm8580"; + reg = <0x1a>; +}; diff --git a/bindings/sound/wm8711.txt b/bindings/sound/wm8711.txt new file mode 100644 index 00000000..c30a1387 --- /dev/null +++ b/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: + +wm8711: codec@1a { + compatible = "wlf,wm8711"; + reg = <0x1a>; +}; diff --git a/bindings/sound/wm8728.txt b/bindings/sound/wm8728.txt new file mode 100644 index 00000000..a3608b4c --- /dev/null +++ b/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: + +wm8728: codec@1a { + compatible = "wlf,wm8728"; + reg = <0x1a>; +}; diff --git a/bindings/sound/wm8731.txt b/bindings/sound/wm8731.txt new file mode 100644 index 00000000..f660d9bb --- /dev/null +++ b/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: + +wm8731: codec@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/bindings/sound/wm8737.txt b/bindings/sound/wm8737.txt new file mode 100644 index 00000000..eda1ec6a --- /dev/null +++ b/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: + +wm8737: codec@1a { + compatible = "wlf,wm8737"; + reg = <0x1a>; +}; diff --git a/bindings/sound/wm8741.txt b/bindings/sound/wm8741.txt new file mode 100644 index 00000000..b69e196c --- /dev/null +++ b/bindings/sound/wm8741.txt @@ -0,0 +1,29 @@ +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. + +Optional properties: + + - diff-mode: Differential output mode configuration. Default value for field + DIFF in register R8 (MODE_CONTROL_2). If absent, the default is 0, shall be: + 0 = stereo + 1 = mono left + 2 = stereo reversed + 3 = mono right + +Example: + +wm8741: codec@1a { + compatible = "wlf,wm8741"; + reg = <0x1a>; + + diff-mode = <3>; +}; diff --git a/bindings/sound/wm8750.txt b/bindings/sound/wm8750.txt new file mode 100644 index 00000000..682f221f --- /dev/null +++ b/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: + +wm8750: codec@1a { + compatible = "wlf,wm8750"; + reg = <0x1a>; +}; diff --git a/bindings/sound/wm8753.txt b/bindings/sound/wm8753.txt new file mode 100644 index 00000000..eca9e5a8 --- /dev/null +++ b/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: + +wm8753: codec@1a { + compatible = "wlf,wm8753"; + reg = <0x1a>; +}; diff --git a/bindings/sound/wm8770.txt b/bindings/sound/wm8770.txt new file mode 100644 index 00000000..cac762a1 --- /dev/null +++ b/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: + +wm8770: codec@1 { + compatible = "wlf,wm8770"; + reg = <1>; +}; diff --git a/bindings/sound/wm8776.txt b/bindings/sound/wm8776.txt new file mode 100644 index 00000000..01173369 --- /dev/null +++ b/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: + +wm8776: codec@1a { + compatible = "wlf,wm8776"; + reg = <0x1a>; +}; diff --git a/bindings/sound/wm8804.txt b/bindings/sound/wm8804.txt new file mode 100644 index 00000000..2c1641c1 --- /dev/null +++ b/bindings/sound/wm8804.txt @@ -0,0 +1,25 @@ +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. + + - PVDD-supply, DVDD-supply : Power supplies for the device, as covered + in Documentation/devicetree/bindings/regulator/regulator.txt + +Optional properties: + + - wlf,reset-gpio: A GPIO specifier for the GPIO controlling the reset pin + +Example: + +wm8804: codec@1a { + compatible = "wlf,wm8804"; + reg = <0x1a>; +}; diff --git a/bindings/sound/wm8903.txt b/bindings/sound/wm8903.txt new file mode 100644 index 00000000..6371c243 --- /dev/null +++ b/bindings/sound/wm8903.txt @@ -0,0 +1,82 @@ +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. + + - AVDD-supply : Analog power supply regulator on the AVDD pin. + + - CPVDD-supply : Charge pump supply regulator on the CPVDD pin. + + - DBVDD-supply : Digital buffer supply regulator for the DBVDD pin. + + - DCVDD-supply : Digital core supply regulator for the DCVDD pin. + +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: + +wm8903: codec@1a { + compatible = "wlf,wm8903"; + reg = <0x1a>; + interrupts = < 347 >; + + AVDD-supply = <&fooreg_a>; + CPVDD-supply = <&fooreg_b>; + DBVDD-supply = <&fooreg_c>; + DCVDC-supply = <&fooreg_d>; + + 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/bindings/sound/wm8904.txt b/bindings/sound/wm8904.txt new file mode 100644 index 00000000..66bf2614 --- /dev/null +++ b/bindings/sound/wm8904.txt @@ -0,0 +1,33 @@ +WM8904 audio CODEC + +This device supports I2C only. + +Required properties: + - compatible: "wlf,wm8904" or "wlf,wm8912" + - reg: the I2C address of the device. + - clock-names: "mclk" + - clocks: reference to + <Documentation/devicetree/bindings/clock/clock-bindings.txt> + +Pins on the device (for linking into audio routes): + + * IN1L + * IN1R + * IN2L + * IN2R + * IN3L + * IN3R + * HPOUTL + * HPOUTR + * LINEOUTL + * LINEOUTR + * MICBIAS + +Examples: + +codec: wm8904@1a { + compatible = "wlf,wm8904"; + reg = <0x1a>; + clocks = <&pck0>; + clock-names = "mclk"; +}; diff --git a/bindings/sound/wm8960.txt b/bindings/sound/wm8960.txt new file mode 100644 index 00000000..6d29ac37 --- /dev/null +++ b/bindings/sound/wm8960.txt @@ -0,0 +1,31 @@ +WM8960 audio CODEC + +This device supports I2C only. + +Required properties: + + - compatible : "wlf,wm8960" + + - reg : the I2C address of the device. + +Optional properties: + - wlf,shared-lrclk: This is a boolean property. If present, the LRCM bit of + R24 (Additional control 2) gets set, indicating that ADCLRC and DACLRC pins + will be disabled only when ADC (Left and Right) and DAC (Left and Right) + are disabled. + When wm8960 works on synchronize mode and DACLRC pin is used to supply + frame clock, it will no frame clock for captrue unless enable DAC to enable + DACLRC pin. If shared-lrclk is present, no need to enable DAC for captrue. + + - wlf,capless: This is a boolean property. If present, OUT3 pin will be + enabled and disabled together with HP_L and HP_R pins in response to jack + detect events. + +Example: + +wm8960: codec@1a { + compatible = "wlf,wm8960"; + reg = <0x1a>; + + wlf,shared-lrclk; +}; diff --git a/bindings/sound/wm8962.txt b/bindings/sound/wm8962.txt new file mode 100644 index 00000000..dcfa9a33 --- /dev/null +++ b/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: + +wm8962: codec@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/bindings/sound/wm8994.txt b/bindings/sound/wm8994.txt new file mode 100644 index 00000000..68cccc46 --- /dev/null +++ b/bindings/sound/wm8994.txt @@ -0,0 +1,83 @@ +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-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/interrupt-controller/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. + + - wlf,spkmode-pu : If present enable the internal pull-up resistor on + the SPKMODE pin. + + - wlf,csnaddr-pd : If present enable the internal pull-down resistor on + the CS/ADDR pin. + +Example: + +wm8994: codec@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>; +}; diff --git a/bindings/sound/zte,tdm.txt b/bindings/sound/zte,tdm.txt new file mode 100644 index 00000000..2a07ca65 --- /dev/null +++ b/bindings/sound/zte,tdm.txt @@ -0,0 +1,30 @@ +ZTE TDM DAI driver + +Required properties: + +- compatible : should be one of the following. + * zte,zx296718-tdm +- reg : physical base address of the controller and length of memory mapped + region. +- clocks : Pairs of phandle and specifier referencing the controller's clocks. +- clock-names: "wclk" for the wclk. + "pclk" for the pclk. +-#clock-cells: should be 1. +- zte,tdm-dma-sysctrl : Reference to the sysctrl controller controlling + the dma. includes: + phandle of sysctrl. + register offset in sysctrl for control dma. + mask of the register that be written to sysctrl. + +Example: + + tdm: tdm@1487000 { + compatible = "zte,zx296718-tdm"; + reg = <0x01487000 0x1000>; + clocks = <&audiocrm AUDIO_TDM_WCLK>, <&audiocrm AUDIO_TDM_PCLK>; + clock-names = "wclk", "pclk"; + #clock-cells = <1>; + pinctrl-names = "default"; + pinctrl-0 = <&tdm_global_pin>; + zte,tdm-dma-sysctrl = <&sysctrl 0x10c 4>; + }; diff --git a/bindings/sound/zte,zx-aud96p22.txt b/bindings/sound/zte,zx-aud96p22.txt new file mode 100644 index 00000000..41bb1040 --- /dev/null +++ b/bindings/sound/zte,zx-aud96p22.txt @@ -0,0 +1,24 @@ +ZTE ZX AUD96P22 Audio Codec + +Required properties: + - compatible: Must be "zte,zx-aud96p22" + - #sound-dai-cells: Should be 0 + - reg: I2C bus slave address of AUD96P22 + +Example: + + i2c0: i2c@1486000 { + compatible = "zte,zx296718-i2c"; + reg = <0x01486000 0x1000>; + interrupts = <GIC_SPI 35 IRQ_TYPE_LEVEL_HIGH>; + #address-cells = <1>; + #size-cells = <0>; + clocks = <&audiocrm AUDIO_I2C0_WCLK>; + clock-frequency = <1600000>; + + aud96p22: codec@22 { + compatible = "zte,zx-aud96p22"; + #sound-dai-cells = <0>; + reg = <0x22>; + }; + }; diff --git a/bindings/sound/zte,zx-i2s.txt b/bindings/sound/zte,zx-i2s.txt new file mode 100644 index 00000000..39272514 --- /dev/null +++ b/bindings/sound/zte,zx-i2s.txt @@ -0,0 +1,45 @@ +ZTE ZX296702 I2S controller + +Required properties: + - compatible : Must be one of: + "zte,zx296718-i2s", "zte,zx296702-i2s" + "zte,zx296702-i2s" + - reg : Must contain I2S core's registers location and length + - clocks : Pairs of phandle and specifier referencing the controller's clocks. + - clock-names: "wclk" for the wclk, "pclk" for the pclk to the I2S interface. + - dmas: Pairs of phandle and specifier for the DMA channel that is used by + the core. The core expects two dma channels for transmit. + - dma-names : Must be "tx" and "rx" + +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: + i2s0: i2s@b005000 { + #sound-dai-cells = <0>; + compatible = "zte,zx296718-i2s", "zte,zx296702-i2s"; + reg = <0x0b005000 0x1000>; + clocks = <&audiocrm AUDIO_I2S0_WCLK>, <&audiocrm AUDIO_I2S0_PCLK>; + clock-names = "wclk", "pclk"; + interrupts = <GIC_SPI 22 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&dma 5>, <&dma 6>; + dma-names = "tx", "rx"; + }; + + sound { + compatible = "simple-audio-card"; + simple-audio-card,name = "zx296702_snd"; + simple-audio-card,format = "left_j"; + simple-audio-card,bitclock-master = <&sndcodec>; + simple-audio-card,frame-master = <&sndcodec>; + sndcpu: simple-audio-card,cpu { + sound-dai = <&i2s0>; + }; + + sndcodec: simple-audio-card,codec { + sound-dai = <&acodec>; + }; + }; diff --git a/bindings/sound/zte,zx-spdif.txt b/bindings/sound/zte,zx-spdif.txt new file mode 100644 index 00000000..09231d75 --- /dev/null +++ b/bindings/sound/zte,zx-spdif.txt @@ -0,0 +1,27 @@ +ZTE ZX296702 SPDIF controller + +Required properties: + - compatible : Must be "zte,zx296702-spdif" + - reg : Must contain SPDIF core's registers location and length + - clocks : Pairs of phandle and specifier referencing the controller's clocks. + - clock-names: "tx" for the clock to the SPDIF interface. + - 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: + spdif0: spdif0@b004000 { + compatible = "zte,zx296702-spdif"; + reg = <0x0b004000 0x1000>; + clocks = <&lsp0clk ZX296702_SPDIF0_DIV>; + clock-names = "tx"; + interrupts = <GIC_SPI 21 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&dma 4>; + dma-names = "tx"; + }; |