diff options
Diffstat (limited to 'Documentation/devicetree/bindings/sound')
39 files changed, 2049 insertions, 589 deletions
diff --git a/Documentation/devicetree/bindings/sound/davinci-mcbsp.txt b/Documentation/devicetree/bindings/sound/davinci-mcbsp.txt deleted file mode 100644 index 3ffc2562fb31..000000000000 --- a/Documentation/devicetree/bindings/sound/davinci-mcbsp.txt +++ /dev/null @@ -1,50 +0,0 @@ -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/Documentation/devicetree/bindings/sound/davinci-mcbsp.yaml b/Documentation/devicetree/bindings/sound/davinci-mcbsp.yaml new file mode 100644 index 000000000000..4fa677023827 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/davinci-mcbsp.yaml @@ -0,0 +1,113 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/davinci-mcbsp.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: McBSP Controller for TI SoCs + +maintainers: + - Bastien Curutchet <bastien.curutchet@bootlin.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - ti,da850-mcbsp + + reg: + minItems: 1 + items: + - description: CFG registers + - description: data registers + + reg-names: + minItems: 1 + items: + - const: mpu + - const: dat + + dmas: + items: + - description: transmission DMA channel + - description: reception DMA channel + + dma-names: + items: + - const: tx + - const: rx + + interrupts: + items: + - description: RX interrupt + - description: TX interrupt + + interrupt-names: + items: + - const: rx + - const: tx + + clocks: + minItems: 1 + items: + - description: functional clock + - description: external input clock for sample rate generator. + + clock-names: + minItems: 1 + items: + - const: fck + - const: clks + + power-domains: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + ti,T1-framing-tx: + $ref: /schemas/types.yaml#/definitions/flag + description: + If the property is present, tx data delay is set to 2 bit clock periods. + McBSP will insert a blank period (high-impedance period) before the first + data bit. This can be used to interface to T1-framing devices. + + ti,T1-framing-rx: + $ref: /schemas/types.yaml#/definitions/flag + description: + If the property is present, rx data delay is set to 2 bit clock periods. + McBSP will discard the bit preceding the data stream (called framing bit). + This can be used to interface to T1-framing devices. + +required: + - "#sound-dai-cells" + - compatible + - reg + - reg-names + - dmas + - dma-names + - clocks + +unevaluatedProperties: false + +examples: + - | + mcbsp0@1d10000 { + #sound-dai-cells = <0>; + compatible = "ti,da850-mcbsp"; + pinctrl-names = "default"; + pinctrl-0 = <&mcbsp0_pins>; + + reg = <0x111000 0x1000>, + <0x311000 0x1000>; + reg-names = "mpu", "dat"; + interrupts = <97>, <98>; + interrupt-names = "rx", "tx"; + dmas = <&edma0 3 1>, + <&edma0 2 1>; + dma-names = "tx", "rx"; + + clocks = <&psc1 14>; + }; diff --git a/Documentation/devicetree/bindings/sound/fsl,audmix.txt b/Documentation/devicetree/bindings/sound/fsl,audmix.txt deleted file mode 100644 index 840b7e0d6a63..000000000000 --- a/Documentation/devicetree/bindings/sound/fsl,audmix.txt +++ /dev/null @@ -1,50 +0,0 @@ -NXP Audio Mixer (AUDMIX). - -The Audio Mixer is a on-chip functional module that allows mixing of two -audio streams into a single audio stream. Audio Mixer has two input serial -audio interfaces. These are driven by two Synchronous Audio interface -modules (SAI). Each input serial interface carries 8 audio channels in its -frame in TDM manner. Mixer mixes audio samples of corresponding channels -from two interfaces into a single sample. Before mixing, audio samples of -two inputs can be attenuated based on configuration. The output of the -Audio Mixer is also a serial audio interface. Like input interfaces it has -the same TDM frame format. This output is used to drive the serial DAC TDM -interface of audio codec and also sent to the external pins along with the -receive path of normal audio SAI module for readback by the CPU. - -The output of Audio Mixer can be selected from any of the three streams - - serial audio input 1 - - serial audio input 2 - - mixed audio - -Mixing operation is independent of audio sample rate but the two audio -input streams must have same audio sample rate with same number of channels -in TDM frame to be eligible for mixing. - -Device driver required properties: -================================= - - compatible : Compatible list, contains "fsl,imx8qm-audmix" - - - 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 "ipg" for register access. - - - power-domains : Must contain the phandle to AUDMIX power domain node - - - dais : Must contain a list of phandles to AUDMIX connected - DAIs. The current implementation requires two phandles - to SAI interfaces to be provided, the first SAI in the - list being used to route the AUDMIX output. - -Device driver configuration example: -====================================== - audmix: audmix@59840000 { - compatible = "fsl,imx8qm-audmix"; - reg = <0x0 0x59840000 0x0 0x10000>; - clocks = <&clk IMX8QXP_AUD_AUDMIX_IPG>; - clock-names = "ipg"; - power-domains = <&pd_audmix>; - dais = <&sai4>, <&sai5>; - }; diff --git a/Documentation/devicetree/bindings/sound/fsl,audmix.yaml b/Documentation/devicetree/bindings/sound/fsl,audmix.yaml new file mode 100644 index 000000000000..9413b901cf77 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,audmix.yaml @@ -0,0 +1,83 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,audmix.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NXP Audio Mixer (AUDMIX). + +maintainers: + - Shengjiu Wang <shengjiu.wang@nxp.com> + - Frank Li <Frank.Li@nxp.com> + +description: | + The Audio Mixer is a on-chip functional module that allows mixing of two + audio streams into a single audio stream. Audio Mixer has two input serial + audio interfaces. These are driven by two Synchronous Audio interface + modules (SAI). Each input serial interface carries 8 audio channels in its + frame in TDM manner. Mixer mixes audio samples of corresponding channels + from two interfaces into a single sample. Before mixing, audio samples of + two inputs can be attenuated based on configuration. The output of the + Audio Mixer is also a serial audio interface. Like input interfaces it has + the same TDM frame format. This output is used to drive the serial DAC TDM + interface of audio codec and also sent to the external pins along with the + receive path of normal audio SAI module for readback by the CPU. + + The output of Audio Mixer can be selected from any of the three streams + - serial audio input 1 + - serial audio input 2 + - mixed audio + + Mixing operation is independent of audio sample rate but the two audio + input streams must have same audio sample rate with same number of channels + in TDM frame to be eligible for mixing. + +properties: + compatible: + const: fsl,imx8qm-audmix + + reg: + maxItems: 1 + + clocks: + maxItems: 1 + + clock-names: + items: + - const: ipg + + power-domains: + maxItems: 1 + + dais: + description: contain a list of phandles to AUDMIX connected DAIs. + $ref: /schemas/types.yaml#/definitions/phandle-array + minItems: 2 + items: + - description: the AUDMIX output + maxItems: 1 + - description: serial audio input 1 + maxItems: 1 + - description: serial audio input 2 + maxItems: 1 + +required: + - compatible + - reg + - clocks + - clock-names + - power-domains + - dais + +unevaluatedProperties: false + +examples: + - | + audmix@59840000 { + compatible = "fsl,imx8qm-audmix"; + reg = <0x59840000 0x10000>; + clocks = <&amix_lpcg 0>; + clock-names = "ipg"; + power-domains = <&pd_audmix>; + dais = <&sai4>, <&sai5>; + }; diff --git a/Documentation/devicetree/bindings/sound/fsl,esai.txt b/Documentation/devicetree/bindings/sound/fsl,esai.txt deleted file mode 100644 index 90112ca1ff42..000000000000 --- a/Documentation/devicetree/bindings/sound/fsl,esai.txt +++ /dev/null @@ -1,68 +0,0 @@ -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, should contain one of the following - compatibles: - "fsl,imx35-esai", - "fsl,vf610-esai", - "fsl,imx6ull-esai", - "fsl,imx8qm-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 Transmission 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/Documentation/devicetree/bindings/sound/fsl,esai.yaml b/Documentation/devicetree/bindings/sound/fsl,esai.yaml new file mode 100644 index 000000000000..f99ed20fa684 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,esai.yaml @@ -0,0 +1,118 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,esai.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Freescale Enhanced Serial Audio Interface (ESAI) Controller + +maintainers: + - Shengjiu Wang <shengjiu.wang@nxp.com> + - Frank Li <Frank.Li@nxp.com> + +description: + 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. + +properties: + compatible: + enum: + - fsl,imx35-esai + - fsl,imx6ull-esai + - fsl,imx8qm-esai + - fsl,vf610-esai + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + minItems: 3 + items: + - description: + The core clock used to access registers. + - description: + The esai baud clock for esai controller used to + derive HCK, SCK and FS. + - description: + The system clock derived from ahb clock used to + derive HCK, SCK and FS. + - description: + 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. + + clock-names: + minItems: 3 + items: + - const: core + - const: extal + - const: fsys + - const: spba + + dmas: + minItems: 2 + maxItems: 2 + + dma-names: + items: + - const: rx + - const: tx + + fsl,fifo-depth: + $ref: /schemas/types.yaml#/definitions/uint32 + default: 64 + description: + 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: + $ref: /schemas/types.yaml#/definitions/flag + description: + 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 Transmission related registers. + + big-endian: + $ref: /schemas/types.yaml#/definitions/flag + description: + 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. + +required: + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + +unevaluatedProperties: false + +allOf: + - $ref: dai-common.yaml# + +examples: + - | + 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/Documentation/devicetree/bindings/sound/fsl,imx-asrc.yaml b/Documentation/devicetree/bindings/sound/fsl,imx-asrc.yaml index bfef2fcb75b1..76aa1f248488 100644 --- a/Documentation/devicetree/bindings/sound/fsl,imx-asrc.yaml +++ b/Documentation/devicetree/bindings/sound/fsl,imx-asrc.yaml @@ -74,6 +74,9 @@ properties: - const: asrck_f - const: spba + power-domains: + maxItems: 1 + fsl,asrc-rate: $ref: /schemas/types.yaml#/definitions/uint32 description: The mutual sample rate used by DPCM Back Ends @@ -131,6 +134,17 @@ allOf: properties: fsl,asrc-clk-map: false + - if: + properties: + compatible: + contains: + enum: + - fsl,imx8qm-asrc + - fsl,imx8qxp-asrc + then: + required: + - power-domains + additionalProperties: false examples: diff --git a/Documentation/devicetree/bindings/sound/fsl,imx-audio-spdif.yaml b/Documentation/devicetree/bindings/sound/fsl,imx-audio-spdif.yaml new file mode 100644 index 000000000000..5fc543d02ecb --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,imx-audio-spdif.yaml @@ -0,0 +1,66 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,imx-audio-spdif.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Freescale i.MX audio complex with S/PDIF transceiver + +maintainers: + - Shengjiu Wang <shengjiu.wang@nxp.com> + +properties: + compatible: + oneOf: + - items: + - enum: + - fsl,imx-sabreauto-spdif + - fsl,imx6sx-sdb-spdif + - const: fsl,imx-audio-spdif + - enum: + - fsl,imx-audio-spdif + + model: + $ref: /schemas/types.yaml#/definitions/string + description: User specified audio sound card name + + spdif-controller: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of the i.MX S/PDIF controller + + spdif-out: + type: boolean + description: + 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: + type: boolean + description: + If present, the receiving function of S/PDIF will be enabled, + indicating there is a physical S/PDIF in connector/jack on the board. + +required: + - compatible + - model + - spdif-controller + +anyOf: + - required: + - spdif-in + - required: + - spdif-out + +additionalProperties: false + +examples: + - | + sound-spdif { + compatible = "fsl,imx-audio-spdif"; + model = "imx-spdif"; + spdif-controller = <&spdif>; + spdif-out; + spdif-in; + }; diff --git a/Documentation/devicetree/bindings/sound/fsl,sai.yaml b/Documentation/devicetree/bindings/sound/fsl,sai.yaml index 2456d958adee..a5d9c246cc47 100644 --- a/Documentation/devicetree/bindings/sound/fsl,sai.yaml +++ b/Documentation/devicetree/bindings/sound/fsl,sai.yaml @@ -81,14 +81,12 @@ properties: dmas: minItems: 1 - items: - - description: DMA controller phandle and request line for RX - - description: DMA controller phandle and request line for TX + maxItems: 2 dma-names: minItems: 1 items: - - const: rx + - enum: [ rx, tx ] - const: tx interrupts: diff --git a/Documentation/devicetree/bindings/sound/fsl,spdif.yaml b/Documentation/devicetree/bindings/sound/fsl,spdif.yaml index 1d64e8337aa4..204f361cea27 100644 --- a/Documentation/devicetree/bindings/sound/fsl,spdif.yaml +++ b/Documentation/devicetree/bindings/sound/fsl,spdif.yaml @@ -31,7 +31,10 @@ properties: maxItems: 1 interrupts: - maxItems: 1 + minItems: 1 + items: + - description: Combined or receive interrupt + - description: Transmit interrupt dmas: items: @@ -86,6 +89,9 @@ properties: registers. Set this flag for HCDs with big endian descriptors and big endian registers. + power-domains: + maxItems: 1 + required: - compatible - reg @@ -97,6 +103,33 @@ required: additionalProperties: false +allOf: + - if: + properties: + compatible: + enum: + - fsl,imx8qm-spdif + - fsl,imx8qxp-spdif + then: + properties: + interrupts: + minItems: 2 + else: + properties: + interrupts: + maxItems: 1 + + - if: + properties: + compatible: + contains: + enum: + - fsl,imx8qm-spdif + - fsl,imx8qxp-spdif + then: + required: + - power-domains + examples: - | spdif@2004000 { diff --git a/Documentation/devicetree/bindings/sound/fsl,ssi.txt b/Documentation/devicetree/bindings/sound/fsl,ssi.txt deleted file mode 100644 index 7e15a85cecd2..000000000000 --- a/Documentation/devicetree/bindings/sound/fsl,ssi.txt +++ /dev/null @@ -1,87 +0,0 @@ -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/Documentation/devicetree/bindings/sound/fsl,ssi.yaml b/Documentation/devicetree/bindings/sound/fsl,ssi.yaml new file mode 100644 index 000000000000..4ab10cd3b520 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,ssi.yaml @@ -0,0 +1,194 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,ssi.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Freescale Synchronous Serial Interface + +maintainers: + - Shengjiu Wang <shengjiu.wang@nxp.com> + +description: + 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. + +properties: + compatible: + oneOf: + - items: + - enum: + - fsl,imx50-ssi + - fsl,imx53-ssi + - const: fsl,imx51-ssi + - const: fsl,imx21-ssi + - items: + - enum: + - fsl,imx25-ssi + - fsl,imx27-ssi + - fsl,imx35-ssi + - fsl,imx51-ssi + - const: fsl,imx21-ssi + - items: + - enum: + - fsl,imx6q-ssi + - fsl,imx6sl-ssi + - fsl,imx6sx-ssi + - const: fsl,imx51-ssi + - items: + - const: fsl,imx21-ssi + - items: + - const: fsl,mpc8610-ssi + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: The ipg clock for register access + - description: clock for SSI master mode + minItems: 1 + + clock-names: + items: + - const: ipg + - const: baud + minItems: 1 + + dmas: + oneOf: + - items: + - description: DMA controller phandle and request line for RX + - description: DMA controller phandle and request line for TX + - items: + - description: DMA controller phandle and request line for RX0 + - description: DMA controller phandle and request line for TX0 + - description: DMA controller phandle and request line for RX1 + - description: DMA controller phandle and request line for TX1 + + dma-names: + oneOf: + - items: + - const: rx + - const: tx + - items: + - const: rx0 + - const: tx0 + - const: rx1 + - const: tx1 + + "#sound-dai-cells": + const: 0 + description: optional, some dts node didn't add it. + + cell-index: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [0, 1, 2] + description: The SSI index + + ac97-gpios: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: Please refer to soc-ac97link.txt + + codec-handle: + $ref: /schemas/types.yaml#/definitions/phandle + description: + 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,fifo-depth: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + The number of elements in the transmit and receive FIFOs. + This number is the maximum allowed value for SFCSR[TFWM0]. + enum: [8, 15] + + fsl,fiq-stream-filter: + type: boolean + description: + 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. + + fsl,mode: + $ref: /schemas/types.yaml#/definitions/string + enum: [ ac97-slave, ac97-master, i2s-slave, i2s-master, + lj-slave, lj-master, rj-slave, rj-master ] + description: | + "ac97-slave" - AC97 mode, SSI is clock slave + "ac97-master" - AC97 mode, SSI is clock master + "i2s-slave" - I2S mode, SSI is clock slave + "i2s-master" - I2S mode, SSI is clock master + "lj-slave" - Left justified mode, SSI is clock slave + "lj-master" - Left justified mode, SSI is clock master + "rj-slave" - Right justified mode, SSI is clock slave + "rj-master" - Right justified mode, SSI is clock master + + fsl,ssi-asynchronous: + type: boolean + description: 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: + $ref: /schemas/types.yaml#/definitions/phandle + description: Phandle to a node for the DMA channel to use for + playback of audio. This is typically dictated by SOC + design. Only used on Power Architecture. + + fsl,capture-dma: + $ref: /schemas/types.yaml#/definitions/phandle + description: Phandle to a node for the DMA channel to use for + capture (recording) of audio. This is typically dictated + by SOC design. Only used on Power Architecture. + +required: + - compatible + - reg + - interrupts + - fsl,fifo-depth + +allOf: + - $ref: dai-common.yaml# + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/imx6qdl-clock.h> + ssi@2028000 { + compatible = "fsl,imx6q-ssi", "fsl,imx51-ssi"; + reg = <0x02028000 0x4000>; + interrupts = <GIC_SPI 46 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&clks IMX6QDL_CLK_SSI1_IPG>, + <&clks IMX6QDL_CLK_SSI1>; + clock-names = "ipg", "baud"; + dmas = <&sdma 37 1 0>, <&sdma 38 1 0>; + dma-names = "rx", "tx"; + #sound-dai-cells = <0>; + fsl,fifo-depth = <15>; + }; diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt deleted file mode 100644 index 4e8dbc5abfd1..000000000000 --- a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt +++ /dev/null @@ -1,117 +0,0 @@ -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" - - "fsl,imx-audio-mqs" - - "fsl,imx-audio-wm8524" - - "fsl,imx-audio-tlv320aic32x4" - - "fsl,imx-audio-tlv320aic31xx" - - "fsl,imx-audio-si476x" - - "fsl,imx-audio-wm8958" - - "fsl,imx-audio-nau8822" - -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 - -Optional properties: - - - audio-asrc : The phandle of ASRC. It can be absent if there's no - need to add ASRC support via DPCM. - - - 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. - - - hp-det-gpio : The GPIO that detect headphones are plugged in - - mic-det-gpio : The GPIO that detect microphones are plugged in - - bitclock-master : Indicates dai-link bit clock master; for details see simple-card.yaml. - - frame-master : Indicates dai-link frame master; for details see simple-card.yaml. - - dai-format : audio format, for details see simple-card.yaml. - - frame-inversion : dai-link uses frame clock inversion, for details see simple-card.yaml. - - bitclock-inversion : dai-link uses bit clock inversion, for details see simple-card.yaml. - - mclk-id : main clock id, specific for each card configuration. - -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/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml b/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml new file mode 100644 index 000000000000..9922664d5ccc --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml @@ -0,0 +1,197 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl-asoc-card.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Freescale Generic ASoC Sound Card with ASRC support + +description: + 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 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. + +maintainers: + - Shengjiu Wang <shengjiu.wang@nxp.com> + +properties: + compatible: + oneOf: + - items: + - enum: + - fsl,imx-sgtl5000 + - fsl,imx25-pdk-sgtl5000 + - fsl,imx53-cpuvo-sgtl5000 + - fsl,imx51-babbage-sgtl5000 + - fsl,imx53-m53evk-sgtl5000 + - fsl,imx53-qsb-sgtl5000 + - fsl,imx53-voipac-sgtl5000 + - fsl,imx6-armadeus-sgtl5000 + - fsl,imx6-rex-sgtl5000 + - fsl,imx6-sabreauto-cs42888 + - fsl,imx6-wandboard-sgtl5000 + - fsl,imx6dl-nit6xlite-sgtl5000 + - fsl,imx6q-ba16-sgtl5000 + - fsl,imx6q-nitrogen6_max-sgtl5000 + - fsl,imx6q-nitrogen6_som2-sgtl5000 + - fsl,imx6q-nitrogen6x-sgtl5000 + - fsl,imx6q-sabrelite-sgtl5000 + - fsl,imx6q-sabresd-wm8962 + - fsl,imx6q-udoo-ac97 + - fsl,imx6q-ventana-sgtl5000 + - fsl,imx6sl-evk-wm8962 + - fsl,imx6sx-sdb-mqs + - fsl,imx6sx-sdb-wm8962 + - fsl,imx7d-evk-wm8960 + - karo,tx53-audio-sgtl5000 + - tq,imx53-mba53-sgtl5000 + - enum: + - fsl,imx-audio-ac97 + - fsl,imx-audio-cs42888 + - fsl,imx-audio-mqs + - fsl,imx-audio-sgtl5000 + - fsl,imx-audio-wm8960 + - fsl,imx-audio-wm8962 + - items: + - enum: + - fsl,imx-audio-ac97 + - fsl,imx-audio-cs42888 + - fsl,imx-audio-cs427x + - fsl,imx-audio-mqs + - fsl,imx-audio-nau8822 + - fsl,imx-audio-sgtl5000 + - fsl,imx-audio-si476x + - fsl,imx-audio-tlv320aic31xx + - fsl,imx-audio-tlv320aic32x4 + - fsl,imx-audio-wm8524 + - fsl,imx-audio-wm8904 + - fsl,imx-audio-wm8960 + - fsl,imx-audio-wm8962 + - fsl,imx-audio-wm8958 + + model: + $ref: /schemas/types.yaml#/definitions/string + description: The user-visible name of this sound complex + + audio-asrc: + $ref: /schemas/types.yaml#/definitions/phandle + description: + The phandle of ASRC. It can be absent if there's no + need to add ASRC support via DPCM. + + audio-codec: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of an audio codec + + audio-cpu: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of an CPU DAI controller + + audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: + 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. "AMIC" stands for Analog Microphone Jack. + "DMIC" stands for Digital Microphone Jack. The "Mic Jack" and "AMIC" + are redundant while coexisting in order to support the old bindings + of wm8962 and sgtl5000. + + hp-det-gpio: + deprecated: true + maxItems: 1 + description: The GPIO that detect headphones are plugged in + + hp-det-gpios: + maxItems: 1 + description: The GPIO that detect headphones are plugged in + + mic-det-gpio: + deprecated: true + maxItems: 1 + description: The GPIO that detect microphones are plugged in + + mic-det-gpios: + maxItems: 1 + description: The GPIO that detect microphones are plugged in + + bitclock-master: + $ref: simple-card.yaml#/definitions/bitclock-master + description: Indicates dai-link bit clock master. + + frame-master: + $ref: simple-card.yaml#/definitions/frame-master + description: Indicates dai-link frame master. + + format: + $ref: simple-card.yaml#/definitions/format + description: audio format. + + frame-inversion: + $ref: simple-card.yaml#/definitions/frame-inversion + description: dai-link uses frame clock inversion. + + bitclock-inversion: + $ref: simple-card.yaml#/definitions/bitclock-inversion + description: dai-link uses bit clock inversion. + + mclk-id: + $ref: /schemas/types.yaml#/definitions/uint32 + description: main clock id, specific for each card configuration. + + mux-int-port: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [1, 2, 7] + description: The internal port of the i.MX audio muxer (AUDMUX) + + mux-ext-port: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [3, 4, 5, 6] + description: The external port of the i.MX audio muxer + + ssi-controller: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of an CPU DAI controller + +required: + - compatible + - model + +unevaluatedProperties: false + +examples: + - | + 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/Documentation/devicetree/bindings/sound/imx-audio-spdif.txt b/Documentation/devicetree/bindings/sound/imx-audio-spdif.txt deleted file mode 100644 index da84a442ccea..000000000000 --- a/Documentation/devicetree/bindings/sound/imx-audio-spdif.txt +++ /dev/null @@ -1,36 +0,0 @@ -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/Documentation/devicetree/bindings/sound/mediatek,mt2701-wm8960.yaml b/Documentation/devicetree/bindings/sound/mediatek,mt2701-wm8960.yaml new file mode 100644 index 000000000000..cf985461a995 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mediatek,mt2701-wm8960.yaml @@ -0,0 +1,54 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/mediatek,mt2701-wm8960.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: MediaTek MT2701 with WM8960 CODEC + +maintainers: + - Kartik Agarwala <agarwala.kartik@gmail.com> + +properties: + compatible: + const: mediatek,mt2701-wm8960-machine + + mediatek,platform: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of MT2701 ASoC platform. + + audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: + 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. + + mediatek,audio-codec: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of the WM8960 audio codec. + +unevaluatedProperties: false + +required: + - compatible + - mediatek,platform + - audio-routing + - mediatek,audio-codec + - pinctrl-names + - pinctrl-0 + +examples: + - | + 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/Documentation/devicetree/bindings/sound/mt2701-wm8960.txt b/Documentation/devicetree/bindings/sound/mt2701-wm8960.txt deleted file mode 100644 index 809b609ea9d0..000000000000 --- a/Documentation/devicetree/bindings/sound/mt2701-wm8960.txt +++ /dev/null @@ -1,24 +0,0 @@ -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/Documentation/devicetree/bindings/sound/mt8186-mt6366-da7219-max98357.yaml b/Documentation/devicetree/bindings/sound/mt8186-mt6366-da7219-max98357.yaml index 9853c11a1330..cbc641ecbe94 100644 --- a/Documentation/devicetree/bindings/sound/mt8186-mt6366-da7219-max98357.yaml +++ b/Documentation/devicetree/bindings/sound/mt8186-mt6366-da7219-max98357.yaml @@ -12,17 +12,46 @@ maintainers: description: This binding describes the MT8186 sound card. +allOf: + - $ref: sound-card-common.yaml# + properties: compatible: enum: - mediatek,mt8186-mt6366-da7219-max98357-sound + audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: + 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 the input or output widgets of audio components, + power supplies, MicBias of codec and the software switch. + minItems: 2 + items: + enum: + # Sinks + - HDMI1 + - Headphones + - Line Out + - MIC + - Speakers + + # Sources + - Headset Mic + - HPL + - HPR + - Speaker + - TX + mediatek,platform: $ref: /schemas/types.yaml#/definitions/phandle description: The phandle of MT8186 ASoC platform. headset-codec: type: object + deprecated: true additionalProperties: false properties: sound-dai: @@ -32,6 +61,7 @@ properties: playback-codecs: type: object + deprecated: true additionalProperties: false properties: sound-dai: @@ -53,32 +83,115 @@ properties: A list of the desired dai-links in the sound card. Each entry is a name defined in the machine driver. -additionalProperties: false +patternProperties: + ".*-dai-link$": + type: object + additionalProperties: false + description: + Container for dai-link level properties and CODEC sub-nodes. + + properties: + link-name: + description: Indicates dai-link name and PCM stream name + items: + enum: + - I2S0 + - I2S1 + - I2S2 + - I2S3 + + codec: + description: Holds subnode which indicates codec dai. + type: object + additionalProperties: false + properties: + sound-dai: + minItems: 1 + maxItems: 2 + required: + - sound-dai + + dai-format: + description: audio format + items: + enum: + - i2s + - right_j + - left_j + - dsp_a + - dsp_b + + mediatek,clk-provider: + $ref: /schemas/types.yaml#/definitions/string + description: Indicates dai-link clock master. + items: + enum: + - cpu + - codec + + required: + - link-name + +unevaluatedProperties: false required: - compatible - mediatek,platform - - headset-codec - - playback-codecs + +# Disallow legacy properties if xxx-dai-link nodes are specified +if: + not: + patternProperties: + ".*-dai-link$": false +then: + properties: + headset-codec: false + speaker-codecs: false examples: - | sound: mt8186-sound { compatible = "mediatek,mt8186-mt6366-da7219-max98357-sound"; - mediatek,platform = <&afe>; + model = "mt8186_da7219_m98357"; pinctrl-names = "aud_clk_mosi_off", "aud_clk_mosi_on"; pinctrl-0 = <&aud_clk_mosi_off>; pinctrl-1 = <&aud_clk_mosi_on>; + mediatek,platform = <&afe>; + + audio-routing = + "Headphones", "HPL", + "Headphones", "HPR", + "MIC", "Headset Mic", + "Speakers", "Speaker", + "HDMI1", "TX"; + + hs-playback-dai-link { + link-name = "I2S0"; + dai-format = "i2s"; + mediatek,clk-provider = "cpu"; + codec { + sound-dai = <&da7219>; + }; + }; - headset-codec { - sound-dai = <&da7219>; + hs-capture-dai-link { + link-name = "I2S1"; + dai-format = "i2s"; + mediatek,clk-provider = "cpu"; + codec { + sound-dai = <&da7219>; + }; }; - playback-codecs { - sound-dai = <&anx_bridge_dp>, - <&max98357a>; + spk-dp-playback-dai-link { + link-name = "I2S3"; + dai-format = "i2s"; + mediatek,clk-provider = "cpu"; + codec { + sound-dai = <&anx_bridge_dp>, <&max98357a>; + }; }; }; diff --git a/Documentation/devicetree/bindings/sound/mt8186-mt6366-rt1019-rt5682s.yaml b/Documentation/devicetree/bindings/sound/mt8186-mt6366-rt1019-rt5682s.yaml index bdf7b0960533..ed93f18ef985 100644 --- a/Documentation/devicetree/bindings/sound/mt8186-mt6366-rt1019-rt5682s.yaml +++ b/Documentation/devicetree/bindings/sound/mt8186-mt6366-rt1019-rt5682s.yaml @@ -12,6 +12,9 @@ maintainers: description: This binding describes the MT8186 sound card. +allOf: + - $ref: sound-card-common.yaml# + properties: compatible: enum: @@ -19,6 +22,34 @@ properties: - mediatek,mt8186-mt6366-rt5682s-max98360-sound - mediatek,mt8186-mt6366-rt5650-sound + audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: + 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 the input or output widgets of audio components, + power supplies, MicBias of codec and the software switch. + minItems: 2 + items: + enum: + # Sinks + - HDMI1 + - Headphone + - IN1P + - IN1N + - Line Out + - Speakers + + # Sources + - Headset Mic + - HPOL + - HPOR + - Speaker + - SPOL + - SPOR + - TX + mediatek,platform: $ref: /schemas/types.yaml#/definitions/phandle description: The phandle of MT8186 ASoC platform. @@ -32,6 +63,7 @@ properties: headset-codec: type: object + deprecated: true additionalProperties: false properties: sound-dai: @@ -41,6 +73,7 @@ properties: playback-codecs: type: object + deprecated: true additionalProperties: false properties: sound-dai: @@ -62,13 +95,56 @@ properties: A list of the desired dai-links in the sound card. Each entry is a name defined in the machine driver. -additionalProperties: false +patternProperties: + ".*-dai-link$": + type: object + additionalProperties: false + description: + Container for dai-link level properties and CODEC sub-nodes. + + properties: + link-name: + description: Indicates dai-link name and PCM stream name + enum: [ I2S0, I2S1, I2S2, I2S3 ] + + codec: + description: Holds subnode which indicates codec dai. + type: object + additionalProperties: false + properties: + sound-dai: + minItems: 1 + maxItems: 2 + required: + - sound-dai + + dai-format: + description: audio format + enum: [ i2s, right_j, left_j, dsp_a, dsp_b ] + + mediatek,clk-provider: + $ref: /schemas/types.yaml#/definitions/string + description: Indicates dai-link clock master. + enum: [ cpu, codec ] + + required: + - link-name + +unevaluatedProperties: false required: - compatible - mediatek,platform - - headset-codec - - playback-codecs + +# Disallow legacy properties if xxx-dai-link nodes are specified +if: + not: + patternProperties: + ".*-dai-link$": false +then: + properties: + headset-codec: false + speaker-codecs: false examples: - | @@ -76,23 +152,49 @@ examples: sound: mt8186-sound { compatible = "mediatek,mt8186-mt6366-rt1019-rt5682s-sound"; - mediatek,platform = <&afe>; + model = "mt8186_rt1019_rt5682s"; pinctrl-names = "aud_clk_mosi_off", "aud_clk_mosi_on", "aud_gpio_dmic_sec"; pinctrl-0 = <&aud_clk_mosi_off>; pinctrl-1 = <&aud_clk_mosi_on>; pinctrl-2 = <&aud_gpio_dmic_sec>; + mediatek,platform = <&afe>; dmic-gpios = <&pio 23 GPIO_ACTIVE_HIGH>; - headset-codec { - sound-dai = <&rt5682s>; + audio-routing = + "Headphone", "HPOL", + "Headphone", "HPOR", + "IN1P", "Headset Mic", + "Speakers", "Speaker", + "HDMI1", "TX"; + + hs-playback-dai-link { + link-name = "I2S0"; + dai-format = "i2s"; + mediatek,clk-provider = "cpu"; + codec { + sound-dai = <&rt5682s 0>; + }; + }; + + hs-capture-dai-link { + link-name = "I2S1"; + dai-format = "i2s"; + mediatek,clk-provider = "cpu"; + codec { + sound-dai = <&rt5682s 0>; + }; }; - playback-codecs { - sound-dai = <&it6505dptx>, - <&rt1019p>; + spk-hdmi-playback-dai-link { + link-name = "I2S3"; + dai-format = "i2s"; + mediatek,clk-provider = "cpu"; + codec { + sound-dai = <&it6505dptx>, <&rt1019p>; + }; }; }; diff --git a/Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml b/Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml index 7e50f5d65c8f..c4e68f31aaab 100644 --- a/Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml +++ b/Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml @@ -13,6 +13,9 @@ maintainers: description: This binding describes the MT8192 sound card. +allOf: + - $ref: sound-card-common.yaml# + properties: compatible: enum: @@ -20,6 +23,31 @@ properties: - mediatek,mt8192_mt6359_rt1015p_rt5682 - mediatek,mt8192_mt6359_rt1015p_rt5682s + audio-routing: + description: + 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 the input or output widgets of audio components, + power supplies, MicBias of codec and the software switch. + minItems: 2 + items: + enum: + # Sinks + - Speakers + - Headphone Jack + - IN1P + - Left Spk + - Right Spk + + # Sources + - Headset Mic + - HPOL + - HPOR + - Left SPO + - Right SPO + - Speaker + mediatek,platform: $ref: /schemas/types.yaml#/definitions/phandle description: The phandle of MT8192 ASoC platform. @@ -27,10 +55,12 @@ properties: mediatek,hdmi-codec: $ref: /schemas/types.yaml#/definitions/phandle description: The phandle of HDMI codec. + deprecated: true headset-codec: type: object additionalProperties: false + deprecated: true properties: sound-dai: @@ -41,6 +71,7 @@ properties: speaker-codecs: type: object additionalProperties: false + deprecated: true properties: sound-dai: @@ -51,33 +82,121 @@ properties: required: - sound-dai -additionalProperties: false +patternProperties: + ".*-dai-link$": + type: object + additionalProperties: false + + description: + Container for dai-link level properties and CODEC sub-nodes. + + properties: + link-name: + description: Indicates dai-link name and PCM stream name + enum: + - I2S0 + - I2S1 + - I2S2 + - I2S3 + - I2S4 + - I2S5 + - I2S6 + - I2S7 + - I2S8 + - I2S9 + - TDM + + codec: + description: Holds subnode which indicates codec dai. + type: object + additionalProperties: false + properties: + sound-dai: + minItems: 1 + maxItems: 2 + required: + - sound-dai + + dai-format: + description: audio format + enum: [ i2s, right_j, left_j, dsp_a, dsp_b ] + + mediatek,clk-provider: + $ref: /schemas/types.yaml#/definitions/string + description: Indicates dai-link clock master. + enum: [ cpu, codec ] + + required: + - link-name + +unevaluatedProperties: false required: - compatible - mediatek,platform - - headset-codec - - speaker-codecs + +# Disallow legacy properties if xxx-dai-link nodes are specified +if: + not: + patternProperties: + ".*-dai-link$": false +then: + properties: + headset-codec: false + speaker-codecs: false + mediatek,hdmi-codec: false examples: - | sound: mt8192-sound { compatible = "mediatek,mt8192_mt6359_rt1015_rt5682"; - mediatek,platform = <&afe>; - mediatek,hdmi-codec = <&anx_bridge_dp>; + model = "mt8192_mt6359_rt1015_rt5682"; pinctrl-names = "aud_clk_mosi_off", "aud_clk_mosi_on"; pinctrl-0 = <&aud_clk_mosi_off>; pinctrl-1 = <&aud_clk_mosi_on>; + mediatek,platform = <&afe>; + + audio-routing = + "Headphone Jack", "HPOL", + "Headphone Jack", "HPOR", + "IN1P", "Headset Mic", + "Speakers", "Speaker"; + + spk-playback-dai-link { + link-name = "I2S3"; + dai-format = "i2s"; + mediatek,clk-provider = "cpu"; + codec { + sound-dai = <&rt1015p>; + }; + }; + + hs-playback-dai-link { + link-name = "I2S8"; + dai-format = "i2s"; + mediatek,clk-provider = "cpu"; + codec { + sound-dai = <&rt5682 0>; + }; + }; - headset-codec { - sound-dai = <&rt5682>; + hs-capture-dai-link { + link-name = "I2S9"; + dai-format = "i2s"; + mediatek,clk-provider = "cpu"; + codec { + sound-dai = <&rt5682 0>; + }; }; - speaker-codecs { - sound-dai = <&rt1015_l>, - <&rt1015_r>; + displayport-dai-link { + link-name = "TDM"; + dai-format = "dsp_a"; + codec { + sound-dai = <&anx_bridge_dp>; + }; }; }; diff --git a/Documentation/devicetree/bindings/sound/mt8195-mt6359.yaml b/Documentation/devicetree/bindings/sound/mt8195-mt6359.yaml index c1ddbf672ca3..2af1d8ffbd8b 100644 --- a/Documentation/devicetree/bindings/sound/mt8195-mt6359.yaml +++ b/Documentation/devicetree/bindings/sound/mt8195-mt6359.yaml @@ -12,6 +12,9 @@ maintainers: description: This binding describes the MT8195 sound card. +allOf: + - $ref: sound-card-common.yaml# + properties: compatible: enum: @@ -23,6 +26,33 @@ properties: $ref: /schemas/types.yaml#/definitions/string description: User specified audio sound card name + audio-routing: + description: + 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 the input or output widgets of audio components, + power supplies, MicBias of codec and the software switch. + minItems: 2 + items: + enum: + # Sinks + - Ext Spk + - Headphone + - IN1P + - Left Spk + - Right Spk + + # Sources + - Headset Mic + - HPOL + - HPOR + - Left BE_OUT + - Left SPO + - Right BE_OUT + - Right SPO + - Speaker + mediatek,platform: $ref: /schemas/types.yaml#/definitions/phandle description: The phandle of MT8195 ASoC platform. @@ -30,10 +60,12 @@ properties: mediatek,dptx-codec: $ref: /schemas/types.yaml#/definitions/phandle description: The phandle of MT8195 Display Port Tx codec node. + deprecated: true mediatek,hdmi-codec: $ref: /schemas/types.yaml#/definitions/phandle description: The phandle of MT8195 HDMI codec node. + deprecated: true mediatek,adsp: $ref: /schemas/types.yaml#/definitions/phandle @@ -45,20 +77,122 @@ properties: A list of the desired dai-links in the sound card. Each entry is a name defined in the machine driver. +patternProperties: + ".*-dai-link$": + type: object + additionalProperties: false + description: + Container for dai-link level properties and CODEC sub-nodes. + + properties: + link-name: + description: Indicates dai-link name and PCM stream name + enum: + - DPTX_BE + - ETDM1_IN_BE + - ETDM2_IN_BE + - ETDM1_OUT_BE + - ETDM2_OUT_BE + - ETDM3_OUT_BE + - PCM1_BE + + codec: + description: Holds subnode which indicates codec dai. + type: object + additionalProperties: false + properties: + sound-dai: + minItems: 1 + maxItems: 2 + required: + - sound-dai + + dai-format: + description: audio format + enum: [ i2s, right_j, left_j, dsp_a, dsp_b ] + + mediatek,clk-provider: + $ref: /schemas/types.yaml#/definitions/string + description: Indicates dai-link clock master. + enum: [ cpu, codec ] + + required: + - link-name + additionalProperties: false required: - compatible - mediatek,platform +# Disallow legacy properties if xxx-dai-link nodes are specified +if: + not: + patternProperties: + ".*-dai-link$": false +then: + properties: + mediatek,dptx-codec: false + mediatek,hdmi-codec: false + examples: - | sound: mt8195-sound { compatible = "mediatek,mt8195_mt6359_rt1019_rt5682"; + model = "mt8195_r1019_5682"; mediatek,platform = <&afe>; pinctrl-names = "default"; pinctrl-0 = <&aud_pins_default>; + + audio-routing = + "Headphone", "HPOL", + "Headphone", "HPOR", + "IN1P", "Headset Mic", + "Ext Spk", "Speaker"; + + mm-dai-link { + link-name = "ETDM1_IN_BE"; + mediatek,clk-provider = "cpu"; + }; + + hs-playback-dai-link { + link-name = "ETDM1_OUT_BE"; + mediatek,clk-provider = "cpu"; + codec { + sound-dai = <&headset_codec>; + }; + }; + + hs-capture-dai-link { + link-name = "ETDM2_IN_BE"; + mediatek,clk-provider = "cpu"; + codec { + sound-dai = <&headset_codec>; + }; + }; + + spk-playback-dai-link { + link-name = "ETDM2_OUT_BE"; + mediatek,clk-provider = "cpu"; + codec { + sound-dai = <&spk_amplifier>; + }; + }; + + hdmi-dai-link { + link-name = "ETDM3_OUT_BE"; + codec { + sound-dai = <&hdmi_tx>; + }; + }; + + displayport-dai-link { + link-name = "DPTX_BE"; + codec { + sound-dai = <&dp_tx>; + }; + }; }; ... diff --git a/Documentation/devicetree/bindings/sound/nuvoton,nau8325.yaml b/Documentation/devicetree/bindings/sound/nuvoton,nau8325.yaml new file mode 100644 index 000000000000..979be0d336da --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nuvoton,nau8325.yaml @@ -0,0 +1,80 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nuvoton,nau8325.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NAU8325 audio Amplifier + +maintainers: + - Seven Lee <WTLI@nuvoton.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: nuvoton,nau8325 + + reg: + maxItems: 1 + + nuvoton,vref-impedance-ohms: + description: + The vref impedance to be used in ohms. Middle of voltage enables + Tie-Off selection options. Due to the high impedance of the VREF + pin, it is important to use a low-leakage capacitor. + + enum: [0, 25000, 125000, 2500] + + nuvoton,dac-vref-microvolt: + description: + The DAC vref to be used in voltage. DAC reference voltage setting. Can + be used for minor tuning of the output level. Since the VDDA is range + between 1.62 to 1.98 voltage, the typical value for design is 1.8V. After + the minor tuning, the final microvolt are as the below. + + enum: [1800000, 2700000, 2880000, 3060000] + + nuvoton,alc-enable: + description: + Enable digital automatic level control (ALC) function. + type: boolean + + nuvoton,clock-detection-disable: + description: + When clock detection is enabled, it will detect whether MCLK + and FS are within the range. MCLK range is from 2.048MHz to 24.576MHz. + FS range is from 8kHz to 96kHz. And also needs to detect the ratio + MCLK_SRC/LRCK of 256, 400 or 500, and needs to detect the BCLK + to make sure data is present. There needs to be at least 8 BCLK + cycles per Frame Sync. + type: boolean + + nuvoton,clock-det-data: + description: + Request clock detection to require 2048 non-zero samples before enabling + the audio paths. If set then non-zero samples is required, otherwise it + doesn't matter. + type: boolean + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@21 { + compatible = "nuvoton,nau8325"; + reg = <0x21>; + nuvoton,vref-impedance-ohms = <125000>; + nuvoton,dac-vref-microvolt = <2880000>; + nuvoton,alc-enable; + nuvoton,clock-det-data; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml b/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml index 054b53954ac3..9f44168efb3e 100644 --- a/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml +++ b/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml @@ -103,6 +103,12 @@ properties: just limited to the left adc for design demand. type: boolean + nuvoton,adc-delay-ms: + description: Delay (in ms) to make input path stable and avoid pop noise. + minimum: 125 + maximum: 500 + default: 125 + '#sound-dai-cells': const: 0 @@ -136,6 +142,7 @@ examples: nuvoton,jack-eject-debounce = <0>; nuvoton,dmic-clk-threshold = <3072000>; nuvoton,dmic-slew-rate = <0>; + nuvoton,adc-delay-ms = <125>; #sound-dai-cells = <0>; }; }; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra20-ac97.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra20-ac97.txt deleted file mode 100644 index eaf00102d92c..000000000000 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra20-ac97.txt +++ /dev/null @@ -1,36 +0,0 @@ -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/Documentation/devicetree/bindings/sound/nvidia,tegra20-ac97.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra20-ac97.yaml new file mode 100644 index 000000000000..4ea0a303d995 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra20-ac97.yaml @@ -0,0 +1,82 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra20-ac97.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NVIDIA Tegra20 AC97 controller + +maintainers: + - Thierry Reding <treding@nvidia.com> + - Jon Hunter <jonathanh@nvidia.com> + +properties: + compatible: + const: nvidia,tegra20-ac97 + + reg: + maxItems: 1 + + resets: + maxItems: 1 + + reset-names: + const: ac97 + + interrupts: + maxItems: 1 + + clocks: + maxItems: 1 + + dmas: + maxItems: 2 + + dma-names: + items: + - const: rx + - const: tx + + nvidia,codec-reset-gpios: + description: Reset pin of external AC97 codec + maxItems: 1 + + nvidia,codec-sync-gpios: + description: AC97 DAP _FS line + maxItems: 1 + +required: + - compatible + - reg + - resets + - reset-names + - interrupts + - clocks + - dmas + - dma-names + - nvidia,codec-reset-gpios + - nvidia,codec-sync-gpios + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/tegra20-car.h> + #include <dt-bindings/gpio/tegra-gpio.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + #include <dt-bindings/gpio/gpio.h> + + ac97@70002000 { + compatible = "nvidia,tegra20-ac97"; + reg = <0x70002000 0x200>; + resets = <&tegra_car 3>; + reset-names = "ac97"; + interrupts = <GIC_SPI 81 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&tegra_car 3>; + dmas = <&apbdma 12>, <&apbdma 12>; + dma-names = "rx", "tx"; + nvidia,codec-reset-gpios = <&gpio TEGRA_GPIO(V, 2) GPIO_ACTIVE_HIGH>; + nvidia,codec-sync-gpios = <&gpio TEGRA_GPIO(P, 0) GPIO_ACTIVE_HIGH>; + }; +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra20-das.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra20-das.txt deleted file mode 100644 index 6de3a7ee4efb..000000000000 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra20-das.txt +++ /dev/null @@ -1,12 +0,0 @@ -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/Documentation/devicetree/bindings/sound/nvidia,tegra20-das.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra20-das.yaml new file mode 100644 index 000000000000..44c5ce8ee6be --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra20-das.yaml @@ -0,0 +1,36 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra20-das.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NVIDIA Tegra 20 DAS (Digital Audio Switch) controller + +maintainers: + - Thierry Reding <treding@nvidia.com> + - Jon Hunter <jonathanh@nvidia.com> + +properties: + compatible: + const: nvidia,tegra20-das + + reg: + maxItems: 1 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + bus { + #address-cells = <1>; + #size-cells = <1>; + das@70000c00 { + compatible = "nvidia,tegra20-das"; + reg = <0x70000c00 0x80>; + }; + }; +... diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra30-i2s.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra30-i2s.txt deleted file mode 100644 index 38caa936f6f8..000000000000 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra30-i2s.txt +++ /dev/null @@ -1,27 +0,0 @@ -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/Documentation/devicetree/bindings/sound/nvidia,tegra30-i2s.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra30-i2s.yaml new file mode 100644 index 000000000000..89c3c6414ab1 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra30-i2s.yaml @@ -0,0 +1,67 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra30-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NVIDIA Tegra30 I2S controller + +maintainers: + - Thierry Reding <treding@nvidia.com> + - Jon Hunter <jonathanh@nvidia.com> + +properties: + compatible: + oneOf: + - enum: + - nvidia,tegra124-i2s + - nvidia,tegra30-i2s + - items: + - const: nvidia,tegra114-i2s + - const: nvidia,tegra30-i2s + + reg: + maxItems: 1 + + clocks: + maxItems: 1 + + clock-names: + const: i2s + + resets: + maxItems: 1 + + reset-names: + const: i2s + + nvidia,ahub-cif-ids: + description: list of AHUB CIF IDs + $ref: /schemas/types.yaml#/definitions/uint32-array + items: + - description: rx (playback) + - description: tx (capture) + +required: + - compatible + - reg + - clocks + - resets + - reset-names + - nvidia,ahub-cif-ids + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/tegra30-car.h> + + i2s@70080300 { + compatible = "nvidia,tegra30-i2s"; + reg = <0x70080300 0x100>; + nvidia,ahub-cif-ids = <4 4>; + clocks = <&tegra_car TEGRA30_CLK_I2S0>; + resets = <&tegra_car 30>; + reset-names = "i2s"; + }; +... diff --git a/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml index 2ab6871e89e5..b2e15ebbd1bc 100644 --- a/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml @@ -29,6 +29,8 @@ properties: - enum: - qcom,apq8016-sbc-sndcard - qcom,msm8916-qdsp6-sndcard + - qcom,qcm6490-idp-sndcard + - qcom,qcs6490-rb3gen2-sndcard - qcom,qrb5165-rb5-sndcard - qcom,sc7180-qdsp6-sndcard - qcom,sc8280xp-sndcard diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml index 0d7a6b576d88..07ec6247d9de 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml @@ -48,13 +48,16 @@ properties: - const: renesas,rcar_sound-gen3 # for Gen4 SoC - items: - - const: renesas,rcar_sound-r8a779g0 # R-Car V4H + - enum: + - renesas,rcar_sound-r8a779g0 # R-Car V4H + - renesas,rcar_sound-r8a779h0 # R-Car V4M - const: renesas,rcar_sound-gen4 # for Generic - enum: - renesas,rcar_sound-gen1 - renesas,rcar_sound-gen2 - renesas,rcar_sound-gen3 + - renesas,rcar_sound-gen4 reg: minItems: 1 diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk3308-codec.yaml b/Documentation/devicetree/bindings/sound/rockchip,rk3308-codec.yaml new file mode 100644 index 000000000000..ecf3d7d968c8 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip,rk3308-codec.yaml @@ -0,0 +1,98 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/rockchip,rk3308-codec.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Rockchip RK3308 Internal Codec + +description: | + This is the audio codec embedded in the Rockchip RK3308 + SoC. It has 8 24-bit ADCs and 2 24-bit DACs. The maximum supported + sampling rate is 192 kHz. + + It is connected internally to one out of a selection of the internal I2S + controllers. + + The RK3308 audio codec has 8 independent capture channels, but some + features work on stereo pairs called groups: + * grp 0 -- MIC1 / MIC2 + * grp 1 -- MIC3 / MIC4 + * grp 2 -- MIC5 / MIC6 + * grp 3 -- MIC7 / MIC8 + +maintainers: + - Luca Ceresoli <luca.ceresoli@bootlin.com> + +properties: + compatible: + const: rockchip,rk3308-codec + + reg: + maxItems: 1 + + rockchip,grf: + $ref: /schemas/types.yaml#/definitions/phandle + description: + Phandle to the General Register Files (GRF) + + clocks: + items: + - description: clock for TX + - description: clock for RX + - description: AHB clock driving the interface + + clock-names: + items: + - const: mclk_tx + - const: mclk_rx + - const: hclk + + resets: + maxItems: 1 + + reset-names: + items: + - const: codec + + "#sound-dai-cells": + const: 0 + + rockchip,micbias-avdd-percent: + description: | + Voltage setting for the MICBIAS pins expressed as a percentage of + AVDD. + + E.g. if rockchip,micbias-avdd-percent = 85 and AVDD = 3v3, then the + MIC BIAS voltage will be 3.3 V * 85% = 2.805 V. + + enum: [ 50, 55, 60, 65, 70, 75, 80, 85 ] + +required: + - compatible + - reg + - rockchip,grf + - clocks + - resets + - "#sound-dai-cells" + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/rk3308-cru.h> + + audio_codec: audio-codec@ff560000 { + compatible = "rockchip,rk3308-codec"; + reg = <0xff560000 0x10000>; + rockchip,grf = <&grf>; + clock-names = "mclk_tx", "mclk_rx", "hclk"; + clocks = <&cru SCLK_I2S2_8CH_TX_OUT>, + <&cru SCLK_I2S2_8CH_RX_OUT>, + <&cru PCLK_ACODEC>; + reset-names = "codec"; + resets = <&cru SRST_ACODEC_P>; + #sound-dai-cells = <0>; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/ti,pcm1681.txt b/Documentation/devicetree/bindings/sound/ti,pcm1681.txt deleted file mode 100644 index 4df17185ab80..000000000000 --- a/Documentation/devicetree/bindings/sound/ti,pcm1681.txt +++ /dev/null @@ -1,15 +0,0 @@ -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/Documentation/devicetree/bindings/sound/ti,pcm1681.yaml b/Documentation/devicetree/bindings/sound/ti,pcm1681.yaml new file mode 100644 index 000000000000..5aa00617291c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,pcm1681.yaml @@ -0,0 +1,43 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,pcm1681.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments PCM1681 8-channel PWM Processor + +maintainers: + - Shenghao Ding <shenghao-ding@ti.com> + - Kevin Lu <kevin-lu@ti.com> + - Baojun Xu <baojun.xu@ti.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: ti,pcm1681 + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + pcm1681: audio-codec@4c { + compatible = "ti,pcm1681"; + reg = <0x4c>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/ti,pcm6240.yaml b/Documentation/devicetree/bindings/sound/ti,pcm6240.yaml new file mode 100644 index 000000000000..dd5b08e3d7a1 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,pcm6240.yaml @@ -0,0 +1,177 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +# Copyright (C) 2022 - 2024 Texas Instruments Incorporated +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,pcm6240.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments PCM6240 Family Audio ADC/DAC + +maintainers: + - Shenghao Ding <shenghao-ding@ti.com> + +description: | + The PCM6240 Family is a big family of Audio ADC/DAC for + different Specifications, range from Personal Electric + to Automotive Electric, even some professional fields. + + Specifications about the audio chip can be found at: + https://www.ti.com/lit/gpn/tlv320adc3120 + https://www.ti.com/lit/gpn/tlv320adc5120 + https://www.ti.com/lit/gpn/tlv320adc6120 + https://www.ti.com/lit/gpn/dix4192 + https://www.ti.com/lit/gpn/pcm1690 + https://www.ti.com/lit/gpn/pcm3120-q1 + https://www.ti.com/lit/gpn/pcm3140-q1 + https://www.ti.com/lit/gpn/pcm5120-q1 + https://www.ti.com/lit/gpn/pcm6120-q1 + https://www.ti.com/lit/gpn/pcm6260-q1 + https://www.ti.com/lit/gpn/pcm9211 + https://www.ti.com/lit/gpn/pcmd3140 + https://www.ti.com/lit/gpn/pcmd3180 + https://www.ti.com/lit/gpn/taa5212 + https://www.ti.com/lit/gpn/tad5212 + +properties: + compatible: + description: | + ti,adc3120: Stereo-channel, 768-kHz, Burr-Brown™ audio analog-to- + digital converter (ADC) with 106-dB SNR. + + ti,adc5120: 2-Channel, 768-kHz, Burr-Brown™ Audio ADC with 120-dB SNR. + + ti,adc6120: Stereo-channel, 768-kHz, Burr-Brown™ audio analog-to- + digital converter (ADC) with 123-dB SNR. + + ti,dix4192: 216-kHz digital audio converter with Quad-Channel In + and One-Channel Out. + + ti,pcm1690: Automotive Catalog 113dB SNR 8-Channel Audio DAC with + Differential Outputs. + + ti,pcm3120: Automotive, stereo, 106-dB SNR, 768-kHz, low-power + software-controlled audio ADC. + + ti,pcm3140: Automotive, Quad-Channel, 768-kHz, Burr-Brown™ Audio ADC + with 106-dB SNR. + + ti,pcm5120: Automotive, stereo, 120-dB SNR, 768-kHz, low-power + software-controlled audio ADC. + + ti,pcm5140: Automotive, Quad-Channel, 768-kHz, Burr-Brown™ Audio ADC + with 120-dB SNR. + + ti,pcm6120: Automotive, stereo, 123-dB SNR, 768-kHz, low-power + software-controlled audio ADC. + + ti,pcm6140: Automotive, Quad-Channel, 768-kHz, Burr-Brown™ Audio ADC + with 123-dB SNR. + + ti,pcm6240: Automotive 4-ch audio ADC with integrated programmable mic + bias, boost and input diagnostics. + + ti,pcm6260: Automotive 6-ch audio ADC with integrated programmable mic + bias, boost and input diagnostics. + + ti,pcm9211: 216-kHz digital audio converter With Stereo ADC and + Routing. + + ti,pcmd3140: Four-channel PDM-input to TDM or I2S output converter. + + ti,pcmd3180: Eight-channel pulse-density-modulation input to TDM or + I2S output converter. + + ti,taa5212: Low-power high-performance stereo audio ADC with 118-dB + dynamic range. + + ti,tad5212: Low-power stereo audio DAC with 120-dB dynamic range. + oneOf: + - items: + - enum: + - ti,adc3120 + - ti,adc5120 + - ti,pcm3120 + - ti,pcm5120 + - ti,pcm6120 + - const: ti,adc6120 + - items: + - enum: + - ti,pcmd512x + - ti,pcm9211 + - ti,taa5212 + - ti,tad5212 + - const: ti,adc6120 + - items: + - enum: + - ti,pcm3140 + - ti,pcm5140 + - ti,dix4192 + - ti,pcm6140 + - ti,pcm6260 + - const: ti,pcm6240 + - items: + - enum: + - ti,pcmd3140 + - ti,pcmd3180 + - ti,pcm1690 + - ti,taa5412 + - ti,tad5412 + - const: ti,pcm6240 + - enum: + - ti,adc6120 + - ti,pcm6240 + + reg: + description: + I2C address, in multiple pcmdevices case, all the i2c address + aggregate as one Audio Device to support multiple audio slots. + minItems: 1 + maxItems: 4 + + reset-gpios: + maxItems: 1 + + interrupts: + maxItems: 1 + description: + Invalid only for ti,pcm1690 because of no INT pin. + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - reg + +allOf: + - $ref: dai-common.yaml# + - if: + properties: + compatible: + contains: + enum: + - ti,pcm1690 + then: + properties: + interrupts: false + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + /* example for two devices with interrupt support */ + #address-cells = <1>; + #size-cells = <0>; + pcm6240: audio-codec@48 { + compatible = "ti,pcm6240"; + reg = <0x48>, /* primary-device */ + <0x4b>; /* secondary-device */ + #sound-dai-cells = <0>; + reset-gpios = <&gpio1 10 GPIO_ACTIVE_HIGH>; + interrupt-parent = <&gpio1>; + interrupts = <15>; + }; + }; +... diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8776.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8776.yaml new file mode 100644 index 000000000000..7bbc96ee81be --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8776.yaml @@ -0,0 +1,41 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/wlf,wm8776.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: WM8776 audio CODEC + +maintainers: + - patches@opensource.cirrus.com + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: wlf,wm8776 + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@1a { + compatible = "wlf,wm8776"; + reg = <0x1a>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8974.txt b/Documentation/devicetree/bindings/sound/wlf,wm8974.txt deleted file mode 100644 index 01d3a7c83419..000000000000 --- a/Documentation/devicetree/bindings/sound/wlf,wm8974.txt +++ /dev/null @@ -1,15 +0,0 @@ -WM8974 audio CODEC - -This device supports both I2C and SPI (configured with pin strapping -on the board). - -Required properties: - - compatible: "wlf,wm8974" - - reg: the I2C address or SPI chip select number of the device - -Examples: - -codec: wm8974@1a { - compatible = "wlf,wm8974"; - reg = <0x1a>; -}; diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8974.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8974.yaml new file mode 100644 index 000000000000..d27300207c67 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8974.yaml @@ -0,0 +1,41 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/wlf,wm8974.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: WM8974 audio CODEC + +maintainers: + - patches@opensource.cirrus.com + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: wlf,wm8974 + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@1a { + compatible = "wlf,wm8974"; + reg = <0x1a>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/wm8776.txt b/Documentation/devicetree/bindings/sound/wm8776.txt deleted file mode 100644 index 01173369c3ed..000000000000 --- a/Documentation/devicetree/bindings/sound/wm8776.txt +++ /dev/null @@ -1,18 +0,0 @@ -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>; -}; |