diff options
| author | Linus Torvalds <torvalds@linux-foundation.org> | 2024-05-15 20:02:36 +0300 |
|---|---|---|
| committer | Linus Torvalds <torvalds@linux-foundation.org> | 2024-05-15 20:02:36 +0300 |
| commit | 33e02dc69afbd8f1b85a51d74d72f139ba4ca623 (patch) | |
| tree | 419637178f5dc6758703143d73eedf484d5b810e /Documentation | |
| parent | d34672777da3ea919e8adb0670ab91ddadf7dea0 (diff) | |
| parent | d731b1ed15052580b7b2f40559021012d280f1d9 (diff) | |
| download | linux-33e02dc69afbd8f1b85a51d74d72f139ba4ca623.tar.xz | |
Merge tag 'sound-6.10-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"This one became bigger than usual, not in the total size but rather
containing lots of small changes all over the places.
The majority of changes are about ASoC, especially SOF / Intel stuff,
and we see an interesting work for ASoC DAPM graph visualization,
while there are many other code cleanup and refactoring, too.
Core:
- A deadlock fix at device disconnection
- A new tool dapm-graph for visualising the DAPM state
ASoC:
- Large updates throughout the Intel audio drivers
- Fixes and clarifications for the DAPM documentation
- Cleanups of accessors for driver data, module labelling, and for
constification
- Modernsation and cleanup work in the Mediatek drivers
- Several fixes and features for the DaVinci I2S driver
- New drivers for several AMD and Intel platforms, Nuvoton NAU8325,
Rockchip RK3308 and Texas Instruments PCM6240
HD-audio:
- Cleanup for CONFIG_PM dependencies
- Cirrus HD-audio codec fixes and quirks
Others:
- Series of tree-wide fixes in Makefiles to use *-y
- Additions of missing module descriptions
- Scarlett2 USB mixer enhancements
- A series of legacy emu10k1 fixes and improvements"
* tag 'sound-6.10-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (603 commits)
ALSA: hda/realtek: Drop doubly quirk entry for 103c:8a2e
ALSA: hda/realtek - fixed headset Mic not show
ASoC: SOF: amd: Fix build error with built-in config
ALSA: scarlett2: Increase mixer range to +12dB
ALSA: scarlett2: Add S/PDIF source selection controls
ALSA: core: Remove superfluous CONFIG_PM
ALSA: Fix deadlocks with kctl removals at disconnection
ASoC: audio-graph-card2: call of_node_get() before of_get_next_child()
ASoC: SOF: amd: Correct spaces in Makefile
ASoC: rt715-sdca-sdw: Fix wrong complete waiting in rt715_dev_resume()
ASoC: Intel: sof_sdw_rt_amp: use dai parameter
ASoC: Intel: sof_sdw: add dai parameter to rtd_init callback
ASoC: Intel: sof_sdw: use .controls/.widgets to add controls/widgets
ASoC: Intel: sof_sdw: add controls and dapm widgets in codec_info
ASoC: Intel: sof_sdw: use generic name for controls/widgets
ASoC: Intel: sof_sdw_cs_amp: rename Speakers to Speaker
ASoC: Intel: maxim-common: change max98373 data to static
ASoC: Intel: sof_sdw: add max98373 dapm routes
ASoC: Intel: sof_rt5682: use max_98373_dai_link function
ASoC: Intel: sof_nau8825: use max_98373_dai_link function
...
Diffstat (limited to 'Documentation')
42 files changed, 2539 insertions, 645 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: + - $ref: dai-common.yaml# + +properties: + compatible: + const: wlf,wm8974 + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@1a { + compatible = "wlf,wm8974"; + reg = <0x1a>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/wm8776.txt b/Documentation/devicetree/bindings/sound/wm8776.txt deleted file mode 100644 index 01173369c3ed..000000000000 --- a/Documentation/devicetree/bindings/sound/wm8776.txt +++ /dev/null @@ -1,18 +0,0 @@ -WM8776 audio CODEC - -This device supports both I2C and SPI (configured with pin strapping -on the board). - -Required properties: - - - compatible : "wlf,wm8776" - - - reg : the I2C address of the device for I2C, the chip select - number for SPI. - -Example: - -wm8776: codec@1a { - compatible = "wlf,wm8776"; - reg = <0x1a>; -}; diff --git a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst index 2d2998faff62..801b0bb57e97 100644 --- a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst +++ b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst @@ -3976,7 +3976,7 @@ Driver with A Single Source File Suppose you have a file xyz.c. Add the following two lines:: - snd-xyz-objs := xyz.o + snd-xyz-y := xyz.o obj-$(CONFIG_SND_XYZ) += snd-xyz.o 2. Create the Kconfig entry @@ -4019,7 +4019,7 @@ located in the new subdirectory, sound/pci/xyz. 2. Under the directory ``sound/pci/xyz``, create a Makefile:: - snd-xyz-objs := xyz.o abc.o def.o + snd-xyz-y := xyz.o abc.o def.o obj-$(CONFIG_SND_XYZ) += snd-xyz.o 3. 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Playback->hdmi-audio-codec.1.auto_TX --> +<g id="edge22" class="edge"> +<title>hdmi-audio-codec.1.auto_I2S Playback->hdmi-audio-codec.1.auto_TX</title> +<path fill="none" stroke="black" d="M254.22,-544.83C253.05,-537.13 251.65,-527.97 250.34,-519.42"/> +<polygon fill="black" stroke="black" points="253.78,-518.77 248.81,-509.41 246.86,-519.83 253.78,-518.77"/> +</g> +<!-- hdmi-audio-codec.1.auto_RX --> +<g id="node29" class="node"> +<title>hdmi-audio-codec.1.auto_RX</title> +<polygon fill="#f2f2f2" stroke="#4d4d4d" points="189.5,-583 118.5,-583 118.5,-545 189.5,-545 189.5,-583"/> +<text text-anchor="middle" x="154" y="-567.8" font-family="sans-serif" font-size="14.00">RX</text> +<text text-anchor="middle" x="154" y="-552.8" font-family="sans-serif" font-size="14.00">[output]</text> +</g> +<!-- hdmi-audio-codec.1.auto_RX->hdmi-audio-codec.1.auto_Capture --> +<g id="edge20" class="edge"> +<title>hdmi-audio-codec.1.auto_RX->hdmi-audio-codec.1.auto_Capture</title> +<path fill="none" stroke="black" d="M154.25,-544.83C154.36,-537.13 154.49,-527.97 154.61,-519.42"/> +<polygon fill="black" stroke="black" points="158.1,-519.46 154.74,-509.41 151.11,-519.36 158.1,-519.46"/> +</g> +</g> +</svg> diff --git a/Documentation/sound/soc/dapm.rst b/Documentation/sound/soc/dapm.rst index c3154ce6e1b2..14c4dc026e6b 100644 --- a/Documentation/sound/soc/dapm.rst +++ b/Documentation/sound/soc/dapm.rst @@ -7,8 +7,8 @@ Description Dynamic Audio Power Management (DAPM) is designed to allow portable Linux devices to use the minimum amount of power within the audio -subsystem at all times. It is independent of other kernel PM and as -such, can easily co-exist with the other PM systems. +subsystem at all times. It is independent of other kernel power +management frameworks and, as such, can easily co-exist with them. DAPM is also completely transparent to all user space applications as all power switching is done within the ASoC core. No code changes or @@ -16,11 +16,29 @@ recompiling are required for user space applications. DAPM makes power switching decisions based upon any audio stream (capture/playback) activity and audio mixer settings within the device. -DAPM spans the whole machine. It covers power control within the entire -audio subsystem, this includes internal codec power blocks and machine -level power systems. +DAPM is based on two basic elements, called widgets and routes: -There are 4 power domains within DAPM + * a **widget** is every part of the audio hardware that can be enabled by + software when in use and disabled to save power when not in use + * a **route** is an interconnection between widgets that exists when sound + can flow from one widget to the other + +All DAPM power switching decisions are made automatically by consulting an +audio routing graph. This graph is specific to each sound card and spans +the whole sound card, so some DAPM routes connect two widgets belonging to +different components (e.g. the LINE OUT pin of a CODEC and the input pin of +an amplifier). + +The graph for the STM32MP1-DK1 sound card is shown in picture: + +.. kernel-figure:: dapm-graph.svg + :alt: Example DAPM graph + :align: center + +DAPM power domains +================== + +There are 4 power domains within DAPM: Codec bias domain VREF, VMID (core codec and audio power) @@ -47,17 +65,11 @@ Stream domain Enabled and disabled when stream playback/capture is started and stopped respectively. e.g. aplay, arecord. -All DAPM power switching decisions are made automatically by consulting an audio -routing map of the whole machine. This map is specific to each machine and -consists of the interconnections between every audio component (including -internal codec components). All audio components that effect power are called -widgets hereafter. - DAPM Widgets ============ -Audio DAPM widgets fall into a number of types:- +Audio DAPM widgets fall into a number of types: Mixer Mixes several analog signals into a single analog signal. @@ -141,14 +153,14 @@ Stream Widgets relate to the stream power domain and only consist of ADCs (analog to digital converters), DACs (digital to analog converters), AIF IN and AIF OUT. -Stream widgets have the following format:- +Stream widgets have the following format: :: SND_SOC_DAPM_DAC(name, stream name, reg, shift, invert), SND_SOC_DAPM_AIF_IN(name, stream, slot, reg, shift, invert) NOTE: the stream name must match the corresponding stream name in your codec -snd_soc_codec_dai. +snd_soc_dai_driver. e.g. stream widgets for HiFi playback and capture :: @@ -167,7 +179,7 @@ Path Domain Widgets ------------------- Path domain widgets have a ability to control or affect the audio signal or -audio paths within the audio subsystem. They have the following form:- +audio paths within the audio subsystem. They have the following form: :: SND_SOC_DAPM_PGA(name, reg, shift, invert, controls, num_controls) @@ -207,7 +219,7 @@ powered. e.g. A machine widget can have an optional call back. e.g. Jack connector widget for an external Mic that enables Mic Bias -when the Mic is inserted:-:: +when the Mic is inserted:: static int spitz_mic_bias(struct snd_soc_dapm_widget* w, int event) { @@ -221,7 +233,7 @@ when the Mic is inserted:-:: Codec (BIAS) Domain ------------------- -The codec bias power domain has no widgets and is handled by the codecs DAPM +The codec bias power domain has no widgets and is handled by the codec DAPM event handler. This handler is called when the codec powerstate is changed wrt to any stream event or by kernel PM events. @@ -229,17 +241,58 @@ to any stream event or by kernel PM events. Virtual Widgets --------------- -Sometimes widgets exist in the codec or machine audio map that don't have any +Sometimes widgets exist in the codec or machine audio graph that don't have any corresponding soft power control. In this case it is necessary to create a virtual widget - a widget with no control bits e.g. :: SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), -This can be used to merge to signal paths together in software. +This can be used to merge two signal paths together in software. -After all the widgets have been defined, they can then be added to the DAPM -subsystem individually with a call to snd_soc_dapm_new_control(). +Registering DAPM controls +========================= + +In many cases the DAPM widgets are implemented statically in a ``static +const struct snd_soc_dapm_widget`` array in a codec driver, and simply +declared via the ``dapm_widgets`` and ``num_dapm_widgets`` fields of the +``struct snd_soc_component_driver``. + +Similarly, routes connecting them are implemented statically in a ``static +const struct snd_soc_dapm_route`` array and declared via the +``dapm_routes`` and ``num_dapm_routes`` fields of the same struct. + +With the above declared, the driver registration will take care of +populating them:: + + static const struct snd_soc_dapm_widget wm2000_dapm_widgets[] = { + SND_SOC_DAPM_OUTPUT("SPKN"), + SND_SOC_DAPM_OUTPUT("SPKP"), + ... + }; + + /* Target, Path, Source */ + static const struct snd_soc_dapm_route wm2000_audio_map[] = { + { "SPKN", NULL, "ANC Engine" }, + { "SPKP", NULL, "ANC Engine" }, + ... + }; + + static const struct snd_soc_component_driver soc_component_dev_wm2000 = { + ... + .dapm_widgets = wm2000_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm2000_dapm_widgets), + .dapm_routes = wm2000_audio_map, + .num_dapm_routes = ARRAY_SIZE(wm2000_audio_map), + ... + }; + +In more complex cases the list of DAPM widgets and/or routes can be only +known at probe time. This happens for example when a driver supports +different models having a different set of features. In those cases +separate widgets and routes arrays implementing the case-specific features +can be registered programmatically by calling snd_soc_dapm_new_controls() +and snd_soc_dapm_add_routes(). Codec/DSP Widget Interconnections @@ -247,31 +300,29 @@ Codec/DSP Widget Interconnections Widgets are connected to each other within the codec, platform and machine by audio paths (called interconnections). Each interconnection must be defined in -order to create a map of all audio paths between widgets. +order to create a graph of all audio paths between widgets. This is easiest with a diagram of the codec or DSP (and schematic of the machine audio system), as it requires joining widgets together via their audio signal paths. -e.g., from the WM8731 output mixer (wm8731.c) - -The WM8731 output mixer has 3 inputs (sources) +For example the WM8731 output mixer (wm8731.c) has 3 inputs (sources): 1. Line Bypass Input 2. DAC (HiFi playback) 3. Mic Sidetone Input -Each input in this example has a kcontrol associated with it (defined in example -above) and is connected to the output mixer via its kcontrol name. We can now -connect the destination widget (wrt audio signal) with its source widgets. -:: +Each input in this example has a kcontrol associated with it (defined in +the example above) and is connected to the output mixer via its kcontrol +name. We can now connect the destination widget (wrt audio signal) with its +source widgets. :: /* output mixer */ {"Output Mixer", "Line Bypass Switch", "Line Input"}, {"Output Mixer", "HiFi Playback Switch", "DAC"}, {"Output Mixer", "Mic Sidetone Switch", "Mic Bias"}, -So we have :- +So we have: * Destination Widget <=== Path Name <=== Source Widget, or * Sink, Path, Source, or @@ -280,12 +331,11 @@ So we have :- When there is no path name connecting widgets (e.g. a direct connection) we pass NULL for the path name. -Interconnections are created with a call to:- -:: +Interconnections are created with a call to:: snd_soc_dapm_connect_input(codec, sink, path, source); -Finally, snd_soc_dapm_new_widgets(codec) must be called after all widgets and +Finally, snd_soc_dapm_new_widgets() must be called after all widgets and interconnections have been registered with the core. This causes the core to scan the codec and machine so that the internal DAPM state matches the physical state of the machine. @@ -326,35 +376,44 @@ jacks can also be switched OFF. DAPM Widget Events ================== -Some widgets can register their interest with the DAPM core in PM events. -e.g. A Speaker with an amplifier registers a widget so the amplifier can be -powered only when the spk is in use. -:: +Widgets needing to implement a more complex behaviour than what DAPM can do +can set a custom "event handler" by setting a function pointer. An example +is a power supply needing to enable a GPIO:: - /* turn speaker amplifier on/off depending on use */ - static int corgi_amp_event(struct snd_soc_dapm_widget *w, int event) + static int sof_es8316_speaker_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) { - gpio_set_value(CORGI_GPIO_APM_ON, SND_SOC_DAPM_EVENT_ON(event)); - return 0; + if (SND_SOC_DAPM_EVENT_ON(event)) + gpiod_set_value_cansleep(gpio_pa, true); + else + gpiod_set_value_cansleep(gpio_pa, false); + + return 0; } - /* corgi machine dapm widgets */ - static const struct snd_soc_dapm_widget wm8731_dapm_widgets = - SND_SOC_DAPM_SPK("Ext Spk", corgi_amp_event); + static const struct snd_soc_dapm_widget st_widgets[] = { + ... + SND_SOC_DAPM_SUPPLY("Speaker Power", SND_SOC_NOPM, 0, 0, + sof_es8316_speaker_power_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + }; -Please see soc-dapm.h for all other widgets that support events. +See soc-dapm.h for all other widgets that support events. Event types ----------- -The following event types are supported by event widgets. -:: +The following event types are supported by event widgets:: /* dapm event types */ - #define SND_SOC_DAPM_PRE_PMU 0x1 /* before widget power up */ - #define SND_SOC_DAPM_POST_PMU 0x2 /* after widget power up */ - #define SND_SOC_DAPM_PRE_PMD 0x4 /* before widget power down */ - #define SND_SOC_DAPM_POST_PMD 0x8 /* after widget power down */ - #define SND_SOC_DAPM_PRE_REG 0x10 /* before audio path setup */ - #define SND_SOC_DAPM_POST_REG 0x20 /* after audio path setup */ + #define SND_SOC_DAPM_PRE_PMU 0x1 /* before widget power up */ + #define SND_SOC_DAPM_POST_PMU 0x2 /* after widget power up */ + #define SND_SOC_DAPM_PRE_PMD 0x4 /* before widget power down */ + #define SND_SOC_DAPM_POST_PMD 0x8 /* after widget power down */ + #define SND_SOC_DAPM_PRE_REG 0x10 /* before audio path setup */ + #define SND_SOC_DAPM_POST_REG 0x20 /* after audio path setup */ + #define SND_SOC_DAPM_WILL_PMU 0x40 /* called at start of sequence */ + #define SND_SOC_DAPM_WILL_PMD 0x80 /* called at start of sequence */ + #define SND_SOC_DAPM_PRE_POST_PMD (SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD) + #define SND_SOC_DAPM_PRE_POST_PMU (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU) |
