diff options
| author | Linus Torvalds <torvalds@linux-foundation.org> | 2023-02-22 21:29:05 +0300 |
|---|---|---|
| committer | Linus Torvalds <torvalds@linux-foundation.org> | 2023-02-22 21:29:05 +0300 |
| commit | 064d7dcf51a82b480e953a15cca47e5df0426502 (patch) | |
| tree | b34ab58ee35e827c14071024dfc79fe50a5f31e6 /Documentation | |
| parent | f3dd0c53370e70c0f9b7e931bbec12916f3bb8cc (diff) | |
| parent | 7933b90b42896f5b6596e6a829bb31c5121fc2a9 (diff) | |
| download | linux-064d7dcf51a82b480e953a15cca47e5df0426502.tar.xz | |
Merge tag 'sound-6.3-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"The majority of works in this cycle are about ASoC spread over trees.
Most of them are for new devices and cleanups / refactoring works, and
not much significant changes are seen in the core side.
Below are some highlights:
ASoC:
- Continued refactoring to move into common helper functions
- Lots of DT schema conversons and stylistic nits
- Continued work on building out the new SOF IPC4 scheme
- Continued work for Intel AVS
- New drivers for Awinc AT88395, Infineon PEB2466, Iron Device
SMA1303, Mediatek MT8188, Realtek RT712, Renesas IDT821034,
Samsung/Tesla FSD SoC I2S, and TI TAS5720A-Q1
ALSA:
- A few cleanups to make the remove callbacks to void returns
- FireWire refactoring and enhancements
- PCM kselftest enhancements"
* tag 'sound-6.3-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (398 commits)
ALSA: hda/hdmi: Register with vga_switcheroo on Dual GPU Macbooks
ASoC: soc-ac97: Return correct error codes
ASoC: soc-dapm.h: fixup warning struct snd_pcm_substream not declared
ASoC: cs35l45: Remove separate namespace for tables
ASoC: cs35l45: Remove separate tables module
ASoC: soc-ac97: Convert to agnostic GPIO API
ASoC: dt-bindings: renesas,rsnd.yaml: drop "dmas/dma-names" from "rcar_sound,ssi"
ALSA: hda: cs35l41: Enable Amp High Pass Filter
ALSA: hda: cs35l41: Ensure firmware/tuning pairs are always loaded
ALSA: hda: cs35l41: Correct error condition handling
ASoC: codecs: wcd934x: Use min macro for comparison and assignment
ASoC: Intel: Skylake: Fix struct definition
ASoC: tlv320adcx140: extend list of supported samplerates
ASoC: imx-pcm-rpmsg: Remove unused variable
SoC: rt5682s: Disable jack detection interrupt during suspend
ASoC: SOF: Intel: hda-dsp: Set streaming flag for d0i3
ASoC: SOF: Intel: Enable d0i3 work for ipc4
ASoC: SOF: ipc4: Wake up dsp core before sending ipc msg
ASoC: SOF: Intel: hda-dsp: use set_pm_gate according to ipc version
ASoC: SOF: Introduce a new set_pm_gate() IPC PM op
...
Diffstat (limited to 'Documentation')
67 files changed, 2093 insertions, 800 deletions
diff --git a/Documentation/devicetree/bindings/dsp/mediatek,mt8186-dsp.yaml b/Documentation/devicetree/bindings/dsp/mediatek,mt8186-dsp.yaml index 3e63f79890b4..88575da1e6d5 100644 --- a/Documentation/devicetree/bindings/dsp/mediatek,mt8186-dsp.yaml +++ b/Documentation/devicetree/bindings/dsp/mediatek,mt8186-dsp.yaml @@ -15,7 +15,9 @@ description: | properties: compatible: - const: mediatek,mt8186-dsp + enum: + - mediatek,mt8186-dsp + - mediatek,mt8188-dsp reg: items: diff --git a/Documentation/devicetree/bindings/sound/adi,adau7002.txt b/Documentation/devicetree/bindings/sound/adi,adau7002.txt deleted file mode 100644 index f144ee1abf85..000000000000 --- a/Documentation/devicetree/bindings/sound/adi,adau7002.txt +++ /dev/null @@ -1,19 +0,0 @@ -Analog Devices ADAU7002 Stereo PDM-to-I2S/TDM Converter - -Required properties: - - - compatible: Must be "adi,adau7002" - -Optional properties: - - - IOVDD-supply: Phandle and specifier for the power supply providing the IOVDD - supply as covered in Documentation/devicetree/bindings/regulator/regulator.txt - - If this property is not present it is assumed that the supply pin is - hardwired to always on. - -Example: - adau7002: pdm-to-i2s { - compatible = "adi,adau7002"; - IOVDD-supply = <&supply>; - }; diff --git a/Documentation/devicetree/bindings/sound/adi,adau7002.yaml b/Documentation/devicetree/bindings/sound/adi,adau7002.yaml new file mode 100644 index 000000000000..fcca0fde7d86 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,adau7002.yaml @@ -0,0 +1,40 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/adi,adau7002.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Analog Devices ADAU7002 Stereo PDM-to-I2S/TDM Converter + +maintainers: + - Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: adi,adau7002 + + IOVDD-supply: + description: + IOVDD power supply, if skipped then it is assumed that the supply pin is + hardwired to always on. + + wakeup-delay-ms: + description: + Delay after power up needed for device to settle. + +required: + - compatible + +unevaluatedProperties: false + +examples: + - | + audio-codec { + compatible = "adi,adau7002"; + IOVDD-supply = <&pp1800_l15a>; + #sound-dai-cells = <0>; + wakeup-delay-ms = <80>; + }; diff --git a/Documentation/devicetree/bindings/sound/ak4613.yaml b/Documentation/devicetree/bindings/sound/ak4613.yaml index 010574645e6a..75e13414d6eb 100644 --- a/Documentation/devicetree/bindings/sound/ak4613.yaml +++ b/Documentation/devicetree/bindings/sound/ak4613.yaml @@ -25,6 +25,13 @@ properties: "#sound-dai-cells": const: 0 + ports: + $ref: audio-graph-port.yaml#/definitions/ports + + port: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + patternProperties: "^asahi-kasei,in[1-2]-single-end$": description: Input Pin 1 - 2. diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-fifo.txt b/Documentation/devicetree/bindings/sound/amlogic,axg-fifo.txt deleted file mode 100644 index fa4545ed81ca..000000000000 --- a/Documentation/devicetree/bindings/sound/amlogic,axg-fifo.txt +++ /dev/null @@ -1,34 +0,0 @@ -* Amlogic Audio FIFO controllers - -Required properties: -- compatible: 'amlogic,axg-toddr' or - 'amlogic,axg-toddr' or - 'amlogic,g12a-frddr' or - 'amlogic,g12a-toddr' or - 'amlogic,sm1-frddr' or - 'amlogic,sm1-toddr' -- reg: physical base address of the controller and length of memory - mapped region. -- interrupts: interrupt specifier for the fifo. -- clocks: phandle to the fifo peripheral clock provided by the audio - clock controller. -- resets: list of reset phandle, one for each entry reset-names. -- reset-names: should contain the following: - * "arb" : memory ARB line (required) - * "rst" : dedicated device reset line (optional) -- #sound-dai-cells: must be 0. -- amlogic,fifo-depth: The size of the controller's fifo in bytes. This - is useful for determining certain configuration such - as the flush threshold of the fifo - -Example of FRDDR A on the A113 SoC: - -frddr_a: audio-controller@1c0 { - compatible = "amlogic,axg-frddr"; - reg = <0x0 0x1c0 0x0 0x1c>; - #sound-dai-cells = <0>; - interrupts = <GIC_SPI 88 IRQ_TYPE_EDGE_RISING>; - clocks = <&clkc_audio AUD_CLKID_FRDDR_A>; - resets = <&arb AXG_ARB_FRDDR_A>; - fifo-depth = <512>; -}; diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-fifo.yaml b/Documentation/devicetree/bindings/sound/amlogic,axg-fifo.yaml new file mode 100644 index 000000000000..b1b48d683101 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-fifo.yaml @@ -0,0 +1,112 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/amlogic,axg-fifo.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Amlogic AXG Audio FIFO controllers + +maintainers: + - Jerome Brunet <jbrunet@baylibre.com> + +properties: + compatible: + oneOf: + - enum: + - amlogic,axg-toddr + - amlogic,axg-frddr + - items: + - enum: + - amlogic,g12a-toddr + - amlogic,sm1-toddr + - const: amlogic,axg-toddr + - items: + - enum: + - amlogic,g12a-frddr + - amlogic,sm1-frddr + - const: amlogic,axg-frddr + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + clocks: + items: + - description: Peripheral clock + + interrupts: + maxItems: 1 + + resets: + minItems: 1 + maxItems: 2 + + reset-names: + minItems: 1 + maxItems: 2 + + amlogic,fifo-depth: + $ref: /schemas/types.yaml#/definitions/uint32 + description: Size of the controller's fifo in bytes + +required: + - compatible + - reg + - "#sound-dai-cells" + - clocks + - interrupts + - resets + - amlogic,fifo-depth + +allOf: + - $ref: dai-common.yaml# + - if: + properties: + compatible: + contains: + enum: + - amlogic,g12a-toddr + - amlogic,sm1-toddr + - amlogic,g12a-frddr + - amlogic,sm1-frddr + + then: + properties: + resets: + minItems: 2 + reset-names: + items: + - const: arb + - const: rst + required: + - reset-names + + else: + properties: + resets: + maxItems: 1 + reset-names: + const: arb + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/axg-audio-clkc.h> + #include <dt-bindings/interrupt-controller/irq.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/reset/amlogic,meson-axg-audio-arb.h> + #include <dt-bindings/reset/amlogic,meson-g12a-audio-reset.h> + + audio-controller@1c0 { + compatible = "amlogic,g12a-frddr", "amlogic,axg-frddr"; + reg = <0x1c0 0x1c>; + #sound-dai-cells = <0>; + clocks = <&clkc_audio AUD_CLKID_FRDDR_A>; + interrupts = <GIC_SPI 88 IRQ_TYPE_EDGE_RISING>; + resets = <&arb>, <&clkc_audio AUD_RESET_FRDDR_A>; + reset-names = "arb", "rst"; + amlogic,fifo-depth = <512>; + }; diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-pdm.txt b/Documentation/devicetree/bindings/sound/amlogic,axg-pdm.txt deleted file mode 100644 index 716878107a24..000000000000 --- a/Documentation/devicetree/bindings/sound/amlogic,axg-pdm.txt +++ /dev/null @@ -1,29 +0,0 @@ -* Amlogic Audio PDM input - -Required properties: -- compatible: 'amlogic,axg-pdm' or - 'amlogic,g12a-pdm' or - 'amlogic,sm1-pdm' -- reg: physical base address of the controller and length of memory - mapped region. -- clocks: list of clock phandle, one for each entry clock-names. -- clock-names: should contain the following: - * "pclk" : peripheral clock. - * "dclk" : pdm digital clock - * "sysclk" : dsp system clock -- #sound-dai-cells: must be 0. - -Optional property: -- resets: phandle to the dedicated reset line of the pdm input. - -Example of PDM on the A113 SoC: - -pdm: audio-controller@ff632000 { - compatible = "amlogic,axg-pdm"; - reg = <0x0 0xff632000 0x0 0x34>; - #sound-dai-cells = <0>; - clocks = <&clkc_audio AUD_CLKID_PDM>, - <&clkc_audio AUD_CLKID_PDM_DCLK>, - <&clkc_audio AUD_CLKID_PDM_SYSCLK>; - clock-names = "pclk", "dclk", "sysclk"; -}; diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-pdm.yaml b/Documentation/devicetree/bindings/sound/amlogic,axg-pdm.yaml new file mode 100644 index 000000000000..df21dd72fc65 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-pdm.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/amlogic,axg-pdm.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Amlogic Audio AXG PDM input + +maintainers: + - Jerome Brunet <jbrunet@baylibre.com> + +properties: + compatible: + oneOf: + - items: + - enum: + - amlogic,g12a-pdm + - amlogic,sm1-pdm + - const: amlogic,axg-pdm + - const: amlogic,axg-pdm + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + clocks: + items: + - description: Peripheral clock + - description: PDM digital clock + - description: DSP system clock + + clock-names: + items: + - const: pclk + - const: dclk + - const: sysclk + + resets: + maxItems: 1 + +required: + - compatible + - reg + - "#sound-dai-cells" + - clocks + - clock-names + +allOf: + - $ref: dai-common.yaml# + + - if: + properties: + compatible: + contains: + enum: + - amlogic,g12a-pdm + - amlogic,sm1-pdm + then: + required: + - resets + + else: + properties: + resets: false + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/axg-audio-clkc.h> + + audio-controller@ff632000 { + compatible = "amlogic,axg-pdm"; + reg = <0xff632000 0x34>; + #sound-dai-cells = <0>; + clocks = <&clkc_audio AUD_CLKID_PDM>, + <&clkc_audio AUD_CLKID_PDM_DCLK>, + <&clkc_audio AUD_CLKID_PDM_SYSCLK>; + clock-names = "pclk", "dclk", "sysclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.txt b/Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.txt deleted file mode 100644 index 80b411296480..000000000000 --- a/Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.txt +++ /dev/null @@ -1,124 +0,0 @@ -Amlogic AXG sound card: - -Required properties: - -- compatible: "amlogic,axg-sound-card" -- model : User specified audio sound card name, one string - -Optional properties: - -- audio-aux-devs : List of phandles pointing to auxiliary devices -- audio-widgets : Please refer to widgets.txt. -- audio-routing : A list of the connections between audio components. - -Subnodes: - -- dai-link: Container for dai-link level properties and the CODEC - sub-nodes. There should be at least one (and probably more) - subnode of this type. - -Required dai-link properties: - -- sound-dai: phandle and port of the CPU DAI. - -Required TDM Backend dai-link properties: -- dai-format : CPU/CODEC common audio format - -Optional TDM Backend dai-link properties: -- dai-tdm-slot-rx-mask-{0,1,2,3}: Receive direction slot masks -- dai-tdm-slot-tx-mask-{0,1,2,3}: Transmit direction slot masks - When omitted, mask is assumed to have to no - slots. A valid must have at one slot, so at - least one these mask should be provided with - an enabled slot. -- dai-tdm-slot-num : Please refer to tdm-slot.txt. - If omitted, slot number is set to accommodate the largest - mask provided. -- dai-tdm-slot-width : Please refer to tdm-slot.txt. default to 32 if omitted. -- mclk-fs : Multiplication factor between stream rate and mclk - -Backend dai-link subnodes: - -- codec: dai-link representing backend links should have at least one subnode. - One subnode for each codec of the dai-link. - dai-link representing frontend links have no codec, therefore have no - subnodes - -Required codec subnodes properties: - -- sound-dai: phandle and port of the CODEC DAI. - -Optional codec subnodes properties: - -- dai-tdm-slot-tx-mask : Please refer to tdm-slot.txt. -- dai-tdm-slot-rx-mask : Please refer to tdm-slot.txt. - -Example: - -sound { - compatible = "amlogic,axg-sound-card"; - model = "AXG-S420"; - audio-aux-devs = <&tdmin_a>, <&tdmout_c>; - audio-widgets = "Line", "Lineout", - "Line", "Linein", - "Speaker", "Speaker1 Left", - "Speaker", "Speaker1 Right"; - "Speaker", "Speaker2 Left", - "Speaker", "Speaker2 Right"; - audio-routing = "TDMOUT_C IN 0", "FRDDR_A OUT 2", - "SPDIFOUT IN 0", "FRDDR_A OUT 3", - "TDM_C Playback", "TDMOUT_C OUT", - "TDMIN_A IN 2", "TDM_C Capture", - "TDMIN_A IN 5", "TDM_C Loopback", - "TODDR_A IN 0", "TDMIN_A OUT", - "Lineout", "Lineout AOUTL", - "Lineout", "Lineout AOUTR", - "Speaker1 Left", "SPK1 OUT_A", - "Speaker2 Left", "SPK2 OUT_A", - "Speaker1 Right", "SPK1 OUT_B", - "Speaker2 Right", "SPK2 OUT_B", - "Linein AINL", "Linein", - "Linein AINR", "Linein"; - - dai-link@0 { - sound-dai = <&frddr_a>; - }; - - dai-link@1 { - sound-dai = <&toddr_a>; - }; - - dai-link@2 { - sound-dai = <&tdmif_c>; - dai-format = "i2s"; - dai-tdm-slot-tx-mask-2 = <1 1>; - dai-tdm-slot-tx-mask-3 = <1 1>; - dai-tdm-slot-rx-mask-1 = <1 1>; - mclk-fs = <256>; - - codec@0 { - sound-dai = <&lineout>; - }; - - codec@1 { - sound-dai = <&speaker_amp1>; - }; - - codec@2 { - sound-dai = <&speaker_amp2>; - }; - - codec@3 { - sound-dai = <&linein>; - }; - - }; - - dai-link@3 { - sound-dai = <&spdifout>; - - codec { - sound-dai = <&spdif_dit>; - }; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.yaml b/Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.yaml new file mode 100644 index 000000000000..bf1234550343 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.yaml @@ -0,0 +1,183 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/amlogic,axg-sound-card.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Amlogic AXG sound card + +maintainers: + - Jerome Brunet <jbrunet@baylibre.com> + +properties: + compatible: + const: amlogic,axg-sound-card + + audio-aux-devs: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: list of auxiliary devices + + 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. + + audio-widgets: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: + A list off component DAPM widget. Each entry is a pair of strings, + the first being the widget type, the second being the widget name + + model: + $ref: /schemas/types.yaml#/definitions/string + description: User specified audio sound card name + +patternProperties: + "^dai-link-[0-9]+$": + type: object + additionalProperties: false + description: + Container for dai-link level properties and the CODEC sub-nodes. + There should be at least one (and probably more) subnode of this type + + properties: + dai-format: + $ref: /schemas/types.yaml#/definitions/string + enum: [ i2s, left-j, dsp_a ] + + dai-tdm-slot-num: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Number of slots in use. If omitted, slot number is set to + accommodate the largest mask provided. + maximum: 32 + + dai-tdm-slot-width: + $ref: /schemas/types.yaml#/definitions/uint32 + description: Width in bits for each slot + enum: [ 8, 16, 20, 24, 32 ] + default: 32 + + mclk-fs: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Multiplication factor between the frame rate and master clock + rate + + sound-dai: + maxItems: 1 + description: phandle of the CPU DAI + + patternProperties: + "^dai-tdm-slot-(t|r)x-mask-[0-3]$": + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 1 + maxItems: 32 + description: + Transmit and receive cpu slot masks of each TDM lane + When omitted, mask is assumed to have to no slots. A valid + interface must have at least one slot, so at least one these + mask should be provided with an enabled slot. + + "^codec(-[0-9]+)?$": + type: object + additionalProperties: false + description: + dai-link representing backend links should have at least one subnode. + One subnode for each codec of the dai-link. dai-link representing + frontend links have no codec, therefore have no subnodes + + properties: + sound-dai: + maxItems: 1 + description: phandle of the codec DAI + + patternProperties: + "^dai-tdm-slot-(t|r)x-mask$": + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 1 + maxItems: 32 + description: Transmit and receive codec slot masks + + required: + - sound-dai + + required: + - sound-dai + +required: + - model + - dai-link-0 + +unevaluatedProperties: false + +examples: + - | + sound { + compatible = "amlogic,axg-sound-card"; + model = "AXG-S420"; + audio-aux-devs = <&tdmin_a>, <&tdmout_c>; + audio-widgets = "Line", "Lineout", + "Line", "Linein", + "Speaker", "Speaker1 Left", + "Speaker", "Speaker1 Right", + "Speaker", "Speaker2 Left", + "Speaker", "Speaker2 Right"; + audio-routing = "TDMOUT_C IN 0", "FRDDR_A OUT 2", + "SPDIFOUT IN 0", "FRDDR_A OUT 3", + "TDM_C Playback", "TDMOUT_C OUT", + "TDMIN_A IN 2", "TDM_C Capture", + "TDMIN_A IN 5", "TDM_C Loopback", + "TODDR_A IN 0", "TDMIN_A OUT", + "Lineout", "Lineout AOUTL", + "Lineout", "Lineout AOUTR", + "Speaker1 Left", "SPK1 OUT_A", + "Speaker2 Left", "SPK2 OUT_A", + "Speaker1 Right", "SPK1 OUT_B", + "Speaker2 Right", "SPK2 OUT_B", + "Linein AINL", "Linein", + "Linein AINR", "Linein"; + + dai-link-0 { + sound-dai = <&frddr_a>; + }; + + dai-link-1 { + sound-dai = <&toddr_a>; + }; + + dai-link-2 { + sound-dai = <&tdmif_c>; + dai-format = "i2s"; + dai-tdm-slot-tx-mask-2 = <1 1>; + dai-tdm-slot-tx-mask-3 = <1 1>; + dai-tdm-slot-rx-mask-1 = <1 1>; + mclk-fs = <256>; + + codec-0 { + sound-dai = <&lineout>; + }; + + codec-1 { + sound-dai = <&speaker_amp1>; + }; + + codec-2 { + sound-dai = <&speaker_amp2>; + }; + + codec-3 { + sound-dai = <&linein>; + }; + }; + + dai-link-3 { + sound-dai = <&spdifout>; + + codec { + sound-dai = <&spdif_dit>; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-spdifin.txt b/Documentation/devicetree/bindings/sound/amlogic,axg-spdifin.txt deleted file mode 100644 index df92a4ecf288..000000000000 --- a/Documentation/devicetree/bindings/sound/amlogic,axg-spdifin.txt +++ /dev/null @@ -1,27 +0,0 @@ -* Amlogic Audio SPDIF Input - -Required properties: -- compatible: 'amlogic,axg-spdifin' or - 'amlogic,g12a-spdifin' or - 'amlogic,sm1-spdifin' -- interrupts: interrupt specifier for the spdif input. -- clocks: list of clock phandle, one for each entry clock-names. -- clock-names: should contain the following: - * "pclk" : peripheral clock. - * "refclk" : spdif input reference clock -- #sound-dai-cells: must be 0. - -Optional property: -- resets: phandle to the dedicated reset line of the spdif input. - -Example on the A113 SoC: - -spdifin: audio-controller@400 { - compatible = "amlogic,axg-spdifin"; - reg = <0x0 0x400 0x0 0x30>; - #sound-dai-cells = <0>; - interrupts = <GIC_SPI 87 IRQ_TYPE_EDGE_RISING>; - clocks = <&clkc_audio AUD_CLKID_SPDIFIN>, - <&clkc_audio AUD_CLKID_SPDIFIN_CLK>; - clock-names = "pclk", "refclk"; -}; diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-spdifin.yaml b/Documentation/devicetree/bindings/sound/amlogic,axg-spdifin.yaml new file mode 100644 index 000000000000..a0bd7a5fb9b3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-spdifin.yaml @@ -0,0 +1,86 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/amlogic,axg-spdifin.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Amlogic Audio AXG SPDIF Input + +maintainers: + - Jerome Brunet <jbrunet@baylibre.com> + +properties: + compatible: + oneOf: + - const: amlogic,axg-spdifin + - items: + - enum: + - amlogic,g12a-spdifin + - amlogic,sm1-spdifin + - const: amlogic,axg-spdifin + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + clocks: + items: + - description: Peripheral clock + - description: SPDIF input reference clock + + clock-names: + items: + - const: pclk + - const: refclk + + interrupts: + maxItems: 1 + + resets: + maxItems: 1 + +required: + - compatible + - reg + - "#sound-dai-cells" + - clocks + - clock-names + - interrupts + +allOf: + - $ref: dai-common.yaml# + + - if: + properties: + compatible: + contains: + enum: + - amlogic,g12a-spdifin + - amlogic,sm1-spdifin + then: + required: + - resets + + else: + properties: + resets: false + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/axg-audio-clkc.h> + #include <dt-bindings/interrupt-controller/irq.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + + audio-controller@400 { + compatible = "amlogic,axg-spdifin"; + reg = <0x400 0x30>; + #sound-dai-cells = <0>; + interrupts = <GIC_SPI 87 IRQ_TYPE_EDGE_RISING>; + clocks = <&clkc_audio AUD_CLKID_SPDIFIN>, + <&clkc_audio AUD_CLKID_SPDIFIN_CLK>; + clock-names = "pclk", "refclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-spdifout.txt b/Documentation/devicetree/bindings/sound/amlogic,axg-spdifout.txt deleted file mode 100644 index 28381dd1f633..000000000000 --- a/Documentation/devicetree/bindings/sound/amlogic,axg-spdifout.txt +++ /dev/null @@ -1,25 +0,0 @@ -* Amlogic Audio SPDIF Output - -Required properties: -- compatible: 'amlogic,axg-spdifout' or - 'amlogic,g12a-spdifout' or - 'amlogic,sm1-spdifout' -- clocks: list of clock phandle, one for each entry clock-names. -- clock-names: should contain the following: - * "pclk" : peripheral clock. - * "mclk" : master clock -- #sound-dai-cells: must be 0. - -Optional property: -- resets: phandle to the dedicated reset line of the spdif output. - -Example on the A113 SoC: - -spdifout: audio-controller@480 { - compatible = "amlogic,axg-spdifout"; - reg = <0x0 0x480 0x0 0x50>; - #sound-dai-cells = <0>; - clocks = <&clkc_audio AUD_CLKID_SPDIFOUT>, - <&clkc_audio AUD_CLKID_SPDIFOUT_CLK>; - clock-names = "pclk", "mclk"; -}; diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-spdifout.yaml b/Documentation/devicetree/bindings/sound/amlogic,axg-spdifout.yaml new file mode 100644 index 000000000000..15be8dae9398 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-spdifout.yaml @@ -0,0 +1,79 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/amlogic,axg-spdifout.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Amlogic Audio AXG SPDIF Output + +maintainers: + - Jerome Brunet <jbrunet@baylibre.com> + +properties: + compatible: + oneOf: + - const: amlogic,axg-spdifout + - items: + - enum: + - amlogic,g12a-spdifout + - amlogic,sm1-spdifout + - const: amlogic,axg-spdifout + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + clocks: + items: + - description: Peripheral clock + - description: SPDIF output master clock + + clock-names: + items: + - const: pclk + - const: mclk + + resets: + maxItems: 1 + +required: + - compatible + - reg + - "#sound-dai-cells" + - clocks + - clock-names + +allOf: + - $ref: dai-common.yaml# + + - if: + properties: + compatible: + contains: + enum: + - amlogic,g12a-spdifout + - amlogic,sm1-spdifout + then: + required: + - resets + + else: + properties: + resets: false + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/axg-audio-clkc.h> + + audio-controller@480 { + compatible = "amlogic,axg-spdifout"; + reg = <0x480 0x50>; + #sound-dai-cells = <0>; + clocks = <&clkc_audio AUD_CLKID_SPDIFOUT>, + <&clkc_audio AUD_CLKID_SPDIFOUT_CLK>; + clock-names = "pclk", "mclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-formatters.txt b/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-formatters.txt deleted file mode 100644 index 5996c0cd89c2..000000000000 --- a/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-formatters.txt +++ /dev/null @@ -1,36 +0,0 @@ -* Amlogic Audio TDM formatters - -Required properties: -- compatible: 'amlogic,axg-tdmin' or - 'amlogic,axg-tdmout' or - 'amlogic,g12a-tdmin' or - 'amlogic,g12a-tdmout' or - 'amlogic,sm1-tdmin' or - 'amlogic,sm1-tdmout -- reg: physical base address of the controller and length of memory - mapped region. -- clocks: list of clock phandle, one for each entry clock-names. -- clock-names: should contain the following: - * "pclk" : peripheral clock. - * "sclk" : bit clock. - * "sclk_sel" : bit clock input multiplexer. - * "lrclk" : sample clock - * "lrclk_sel": sample clock input multiplexer - -Optional property: -- resets: phandle to the dedicated reset line of the tdm formatter. - -Example of TDMOUT_A on the S905X2 SoC: - -tdmout_a: audio-controller@500 { - compatible = "amlogic,axg-tdmout"; - reg = <0x0 0x500 0x0 0x40>; - resets = <&clkc_audio AUD_RESET_TDMOUT_A>; - clocks = <&clkc_audio AUD_CLKID_TDMOUT_A>, - <&clkc_audio AUD_CLKID_TDMOUT_A_SCLK>, - <&clkc_audio AUD_CLKID_TDMOUT_A_SCLK_SEL>, - <&clkc_audio AUD_CLKID_TDMOUT_A_LRCLK>, - <&clkc_audio AUD_CLKID_TDMOUT_A_LRCLK>; - clock-names = "pclk", "sclk", "sclk_sel", - "lrclk", "lrclk_sel"; -}; diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-formatters.yaml b/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-formatters.yaml new file mode 100644 index 000000000000..719ca8fc98c7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-formatters.yaml @@ -0,0 +1,88 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/amlogic,axg-tdm-formatters.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Amlogic Audio AXG TDM formatters + +maintainers: + - Jerome Brunet <jbrunet@baylibre.com> + +properties: + compatible: + enum: + - amlogic,g12a-tdmout + - amlogic,sm1-tdmout + - amlogic,axg-tdmout + - amlogic,g12a-tdmin + - amlogic,sm1-tdmin + - amlogic,axg-tdmin + + clocks: + items: + - description: Peripheral clock + - description: Bit clock + - description: Bit clock input multiplexer + - description: Sample clock + - description: Sample clock input multiplexer + + clock-names: + items: + - const: pclk + - const: sclk + - const: sclk_sel + - const: lrclk + - const: lrclk_sel + + reg: + maxItems: 1 + + resets: + maxItems: 1 + +required: + - compatible + - reg + - clocks + - clock-names + +allOf: + - $ref: component-common.yaml# + + - if: + properties: + compatible: + contains: + enum: + - amlogic,g12a-tdmin + - amlogic,sm1-tdmin + - amlogic,g12a-tdmout + - amlogic,sm1-tdmout + then: + required: + - resets + + else: + properties: + resets: false + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/axg-audio-clkc.h> + #include <dt-bindings/reset/amlogic,meson-g12a-audio-reset.h> + + audio-controller@500 { + compatible = "amlogic,g12a-tdmout"; + reg = <0x500 0x40>; + resets = <&clkc_audio AUD_RESET_TDMOUT_A>; + clocks = <&clkc_audio AUD_CLKID_TDMOUT_A>, + <&clkc_audio AUD_CLKID_TDMOUT_A_SCLK>, + <&clkc_audio AUD_CLKID_TDMOUT_A_SCLK_SEL>, + <&clkc_audio AUD_CLKID_TDMOUT_A_LRCLK>, + <&clkc_audio AUD_CLKID_TDMOUT_A_LRCLK>; + clock-names = "pclk", "sclk", "sclk_sel", + "lrclk", "lrclk_sel"; + }; diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-iface.txt b/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-iface.txt deleted file mode 100644 index cabfb26a5f22..000000000000 --- a/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-iface.txt +++ /dev/null @@ -1,22 +0,0 @@ -* Amlogic Audio TDM Interfaces - -Required properties: -- compatible: 'amlogic,axg-tdm-iface' -- clocks: list of clock phandle, one for each entry clock-names. -- clock-names: should contain the following: - * "sclk" : bit clock. - * "lrclk": sample clock - * "mclk" : master clock - -> optional if the interface is in clock slave mode. -- #sound-dai-cells: must be 0. - -Example of TDM_A on the A113 SoC: - -tdmif_a: audio-controller@0 { - compatible = "amlogic,axg-tdm-iface"; - #sound-dai-cells = <0>; - clocks = <&clkc_audio AUD_CLKID_MST_A_MCLK>, - <&clkc_audio AUD_CLKID_MST_A_SCLK>, - <&clkc_audio AUD_CLKID_MST_A_LRCLK>; - clock-names = "mclk", "sclk", "lrclk"; -}; diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-iface.yaml b/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-iface.yaml new file mode 100644 index 000000000000..320f0002649d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-tdm-iface.yaml @@ -0,0 +1,55 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/amlogic,axg-tdm-iface.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Amlogic Audio TDM Interfaces + +maintainers: + - Jerome Brunet <jbrunet@baylibre.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: amlogic,axg-tdm-iface + + "#sound-dai-cells": + const: 0 + + clocks: + minItems: 2 + items: + - description: Bit clock + - description: Sample clock + - description: Master clock #optional + + clock-names: + minItems: 2 + items: + - const: sclk + - const: lrclk + - const: mclk + +required: + - compatible + - "#sound-dai-cells" + - clocks + - clock-names + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/axg-audio-clkc.h> + + audio-controller { + compatible = "amlogic,axg-tdm-iface"; + #sound-dai-cells = <0>; + clocks = <&clkc_audio AUD_CLKID_MST_A_SCLK>, + <&clkc_audio AUD_CLKID_MST_A_LRCLK>, + <&clkc_audio AUD_CLKID_MST_A_MCLK>; + clock-names = "sclk", "lrclk", "mclk"; + }; diff --git a/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml b/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml index 5b8d59245f82..b358fd601ed3 100644 --- a/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml +++ b/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml @@ -62,7 +62,7 @@ patternProperties: description: phandle of the CPU DAI patternProperties: - "^codec-[0-9]+$": + "^codec(-[0-9]+)?$": type: object additionalProperties: false description: |- diff --git a/Documentation/devicetree/bindings/sound/audio-graph-port.yaml b/Documentation/devicetree/bindings/sound/audio-graph-port.yaml index f5b8b6d13077..6b4e02a0695a 100644 --- a/Documentation/devicetree/bindings/sound/audio-graph-port.yaml +++ b/Documentation/devicetree/bindings/sound/audio-graph-port.yaml @@ -11,32 +11,24 @@ maintainers: select: false -allOf: - - $ref: /schemas/graph.yaml#/$defs/port-base - -properties: - prefix: - description: "device name prefix" - $ref: /schemas/types.yaml#/definitions/string - convert-rate: - $ref: "/schemas/sound/dai-params.yaml#/$defs/dai-sample-rate" - convert-channels: - $ref: "/schemas/sound/dai-params.yaml#/$defs/dai-channels" - convert-sample-format: - $ref: "/schemas/sound/dai-params.yaml#/$defs/dai-sample-format" +definitions: + port-base: + $ref: /schemas/graph.yaml#/$defs/port-base + properties: + convert-rate: + $ref: "/schemas/sound/dai-params.yaml#/$defs/dai-sample-rate" + convert-channels: + $ref: "/schemas/sound/dai-params.yaml#/$defs/dai-channels" + convert-sample-format: + $ref: "/schemas/sound/dai-params.yaml#/$defs/dai-sample-format" + mclk-fs: + $ref: "simple-card.yaml#/definitions/mclk-fs" -patternProperties: - "^endpoint(@[0-9a-f]+)?": + endpoint-base: $ref: /schemas/graph.yaml#/$defs/endpoint-base - unevaluatedProperties: false - properties: mclk-fs: - description: | - Multiplication factor between stream rate and codec mclk. - When defined, mclk-fs property defined in dai-link sub nodes are - ignored. - $ref: /schemas/types.yaml#/definitions/uint32 + $ref: "simple-card.yaml#/definitions/mclk-fs" frame-inversion: description: dai-link uses frame clock inversion $ref: /schemas/types.yaml#/definitions/flag @@ -53,6 +45,15 @@ patternProperties: oneOf: - $ref: /schemas/types.yaml#/definitions/flag - $ref: /schemas/types.yaml#/definitions/phandle + clocks: + description: Indicates system clock + $ref: /schemas/types.yaml#/definitions/phandle + system-clock-frequency: + $ref: "simple-card.yaml#/definitions/system-clock-frequency" + system-clock-direction-out: + $ref: "simple-card.yaml#/definitions/system-clock-direction-out" + system-clock-fixed: + $ref: "simple-card.yaml#/definitions/system-clock-fixed" dai-format: description: audio format. @@ -100,4 +101,24 @@ patternProperties: minimum: 1 maximum: 64 + ports: + $ref: "#/definitions/port-base" + unevaluatedProperties: false + patternProperties: + "^port(@[0-9a-f]+)?$": + $ref: "#/definitions/port-base" + unevaluatedProperties: false + patternProperties: + "^endpoint(@[0-9a-f]+)?": + $ref: "#/definitions/endpoint-base" + unevaluatedProperties: false + +allOf: + - $ref: "#/definitions/port-base" + +patternProperties: + "^endpoint(@[0-9a-f]+)?": + $ref: "#/definitions/endpoint-base" + unevaluatedProperties: false + additionalProperties: true diff --git a/Documentation/devicetree/bindings/sound/awinic,aw88395.yaml b/Documentation/devicetree/bindings/sound/awinic,aw88395.yaml new file mode 100644 index 000000000000..35eef7d818a2 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/awinic,aw88395.yaml @@ -0,0 +1,53 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/awinic,aw88395.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Awinic AW88395 Smart Audio Amplifier + +maintainers: + - Weidong Wang <wangweidong.a@awinic.com> + +description: + The Awinic AW88395 is an I2S/TDM input, high efficiency + digital Smart K audio amplifier with an integrated 10.25V + smart boost convert. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: awinic,aw88395 + + reg: + maxItems: 1 + + '#sound-dai-cells': + const: 0 + + reset-gpios: + maxItems: 1 + +required: + - compatible + - reg + - '#sound-dai-cells' + - reset-gpios + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + audio-codec@34 { + compatible = "awinic,aw88395"; + reg = <0x34>; + #sound-dai-cells = <0>; + reset-gpios = <&gpio 10 GPIO_ACTIVE_LOW>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/cirrus,cs35l41.yaml b/Documentation/devicetree/bindings/sound/cirrus,cs35l41.yaml index 82062d80d700..18fb471aa891 100644 --- a/Documentation/devicetree/bindings/sound/cirrus,cs35l41.yaml +++ b/Documentation/devicetree/bindings/sound/cirrus,cs35l41.yaml @@ -22,6 +22,9 @@ properties: reg: maxItems: 1 + interrupts: + maxItems: 1 + '#sound-dai-cells': description: The first cell indicating the audio interface. @@ -42,7 +45,7 @@ properties: Configures the peak current by monitoring the current through the boost FET. Range starts at 1600 mA and goes to a maximum of 4500 mA with increments of 50 mA. See section 4.3.6 of the datasheet for details. - $ref: "/schemas/types.yaml#/definitions/uint32" + $ref: /schemas/types.yaml#/definitions/uint32 minimum: 1600 maximum: 4500 default: 4500 @@ -51,7 +54,7 @@ properties: description: Boost inductor value, expressed in nH. Valid values include 1000, 1200, 1500 and 2200. - $ref: "/schemas/types.yaml#/definitions/uint32" + $ref: /schemas/types.yaml#/definitions/uint32 minimum: 1000 maximum: 2200 @@ -60,7 +63,7 @@ properties: Total equivalent boost capacitance on the VBST and VAMP pins, derated at 11 volts DC. The value must be rounded to the nearest integer and expressed in uF. - $ref: "/schemas/types.yaml#/definitions/uint32" + $ref: /schemas/types.yaml#/definitions/uint32 cirrus,asp-sdout-hiz: description: @@ -70,7 +73,7 @@ properties: 1 = Hi-Z during unused slots but logic 0 while all transmit channels disabled 2 = (Default) Logic 0 during unused slots, but Hi-Z while all transmit channels disabled 3 = Hi-Z during unused slots and while all transmit channels disabled - $ref: "/schemas/types.yaml#/definitions/uint32" + $ref: /schemas/types.yaml#/definitions/uint32 minimum: 0 maximum: 3 default: 2 @@ -84,7 +87,7 @@ properties: enable boost voltage. 0 = Internal Boost 1 = External Boost - $ref: "/schemas/types.yaml#/definitions/uint32" + $ref: /schemas/types.yaml#/definitions/uint32 minimum: 0 maximum: 1 @@ -109,7 +112,7 @@ properties: 1 = GPIO 2 = Sync 3 = MCLK input - $ref: "/schemas/types.yaml#/definitions/uint32" + $ref: /schemas/types.yaml#/definitions/uint32 minimum: 0 maximum: 3 @@ -136,7 +139,7 @@ properties: 3 = MCLK input 4 = Push-pull INTB (active low) 5 = Push-pull INT (active high) - $ref: "/schemas/types.yaml#/definitions/uint32" + $ref: /schemas/types.yaml#/definitions/uint32 minimum: 0 maximum: 5 @@ -176,21 +179,23 @@ unevaluatedProperties: false examples: - | + #include <dt-bindings/gpio/gpio.h> + spi { #address-cells = <1>; #size-cells = <0>; - cs35l41: cs35l41@2 { - #sound-dai-cells = <1>; - compatible = "cirrus,cs35l41"; - reg = <2>; - VA-supply = <&dummy_vreg>; - VP-supply = <&dummy_vreg>; - reset-gpios = <&gpio 110 0>; - - cirrus,boost-type = <0>; - cirrus,boost-peak-milliamp = <4500>; - cirrus,boost-ind-nanohenry = <1000>; - cirrus,boost-cap-microfarad = <15>; + cs35l41: speaker-amp@2 { + #sound-dai-cells = <1>; + compatible = "cirrus,cs35l41"; + reg = <2>; + VA-supply = <&dummy_vreg>; + VP-supply = <&dummy_vreg>; + reset-gpios = <&gpio 110 GPIO_ACTIVE_HIGH>; + + cirrus,boost-type = <0>; + cirrus,boost-peak-milliamp = <4500>; + cirrus,boost-ind-nanohenry = <1000>; + cirrus,boost-cap-microfarad = <15>; }; }; diff --git a/Documentation/devicetree/bindings/sound/component-common.yaml b/Documentation/devicetree/bindings/sound/component-common.yaml new file mode 100644 index 000000000000..37766c5f3974 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/component-common.yaml @@ -0,0 +1,21 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/component-common.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Audio Component Common Properties + +maintainers: + - Jerome Brunet <jbrunet@baylibre.com> + +properties: + sound-name-prefix: + $ref: /schemas/types.yaml#/definitions/string + description: | + Card implementing the routing property define the connection between + audio components as list of string pair. Component using the same + sink/source names may use this property to prepend the name of their + sinks/sources with the provided string. + +additionalProperties: true diff --git a/Documentation/devicetree/bindings/sound/dai-common.yaml b/Documentation/devicetree/bindings/sound/dai-common.yaml index d858eea73ed7..1aed2f0f1775 100644 --- a/Documentation/devicetree/bindings/sound/dai-common.yaml +++ b/Documentation/devicetree/bindings/sound/dai-common.yaml @@ -9,15 +9,10 @@ title: Digital Audio Interface Common Properties maintainers: - Jerome Brunet <jbrunet@baylibre.com> -properties: - sound-name-prefix: - $ref: /schemas/types.yaml#/definitions/string - description: | - Card implementing the routing property define the connection between - audio components as list of string pair. Component using the same - sink/source names may use this property to prepend the name of their - sinks/sources with the provided string. +allOf: + - $ref: component-common.yaml# +properties: '#sound-dai-cells': true additionalProperties: true diff --git a/Documentation/devicetree/bindings/sound/fsl,sai.yaml b/Documentation/devicetree/bindings/sound/fsl,sai.yaml index 7e56337d8edc..088c26b001cc 100644 --- a/Documentation/devicetree/bindings/sound/fsl,sai.yaml +++ b/Documentation/devicetree/bindings/sound/fsl,sai.yaml @@ -76,10 +76,14 @@ properties: minItems: 4 dmas: - maxItems: 2 + items: + - description: DMA controller phandle and request line for RX + - description: DMA controller phandle and request line for TX dma-names: - maxItems: 2 + items: + - const: rx + - const: tx interrupts: items: @@ -143,31 +147,6 @@ properties: allOf: - $ref: dai-common.yaml# - if: - properties: - compatible: - contains: - const: fsl,vf610-sai - then: - properties: - dmas: - items: - - description: DMA controller phandle and request line for TX - - description: DMA controller phandle and request line for RX - dma-names: - items: - - const: tx - - const: rx - else: - properties: - dmas: - items: - - description: DMA controller phandle and request line for RX - - description: DMA controller phandle and request line for TX - dma-names: - items: - - const: rx - - const: tx - - if: required: - fsl,sai-asynchronous then: @@ -199,9 +178,8 @@ examples: <&clks VF610_CLK_SAI2>, <&clks 0>, <&clks 0>; clock-names = "bus", "mclk1", "mclk2", "mclk3"; - dma-names = "tx", "rx"; - dmas = <&edma0 0 21>, - <&edma0 0 20>; + dma-names = "rx", "tx"; + dmas = <&edma0 0 20>, <&edma0 0 21>; big-endian; lsb-first; }; diff --git a/Documentation/devicetree/bindings/sound/fsl,xcvr.yaml b/Documentation/devicetree/bindings/sound/fsl,xcvr.yaml index 223b8ea693dc..799b362ba498 100644 --- a/Documentation/devicetree/bindings/sound/fsl,xcvr.yaml +++ b/Documentation/devicetree/bindings/sound/fsl,xcvr.yaml @@ -21,6 +21,7 @@ properties: compatible: enum: - fsl,imx8mp-xcvr + - fsl,imx93-xcvr reg: items: diff --git a/Documentation/devicetree/bindings/sound/google,sc7280-herobrine.yaml b/Documentation/devicetree/bindings/sound/google,sc7280-herobrine.yaml index 869b40363af8..0b1a01a4c14e 100644 --- a/Documentation/devicetree/bindings/sound/google,sc7280-herobrine.yaml +++ b/Documentation/devicetree/bindings/sound/google,sc7280-herobrine.yaml @@ -75,6 +75,18 @@ patternProperties: additionalProperties: false + platform: + description: Holds subnode which includes the phandle of q6apm platform device. + type: object + properties: + sound-dai: + maxItems: 1 + + required: + - sound-dai + + additionalProperties: false + required: - link-name - cpu diff --git a/Documentation/devicetree/bindings/sound/infineon,peb2466.yaml b/Documentation/devicetree/bindings/sound/infineon,peb2466.yaml new file mode 100644 index 000000000000..66993d378aaf --- /dev/null +++ b/Documentation/devicetree/bindings/sound/infineon,peb2466.yaml @@ -0,0 +1,91 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/infineon,peb2466.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Infineon PEB2466 codec + +maintainers: + - Herve Codina <herve.codina@bootlin.com> + +description: | + The Infineon PEB2466 codec is a programmable DSP-based four channels codec + with filters capabilities. + + The time-slots used by the codec must be set and so, the properties + 'dai-tdm-slot-num', 'dai-tdm-slot-width', 'dai-tdm-slot-tx-mask' and + 'dai-tdm-slot-rx-mask' must be present in the sound card node for sub-nodes + that involve the codec. The codec uses one 8bit time-slot per channel. + 'dai-tdm-tdm-slot-with' must be set to 8. + + The PEB2466 codec also supports 28 gpios (signaling pins). + +allOf: + - $ref: /schemas/spi/spi-peripheral-props.yaml + - $ref: dai-common.yaml# + +properties: + compatible: + const: infineon,peb2466 + + reg: + description: + SPI device address. + maxItems: 1 + + clocks: + items: + - description: Master clock + + clock-names: + items: + - const: mclk + + spi-max-frequency: + maximum: 8192000 + + reset-gpios: + description: + GPIO used to reset the device. + maxItems: 1 + + firmware-name: + $ref: /schemas/types.yaml#/definitions/string + description: + Filters coefficients file to load. If this property is omitted, internal + filters are disabled. + + '#sound-dai-cells': + const: 0 + + '#gpio-cells': + const: 2 + + gpio-controller: true + +required: + - compatible + - reg + - '#sound-dai-cells' + - gpio-controller + - '#gpio-cells' + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + spi { + #address-cells = <1>; + #size-cells = <0>; + audio-codec@0 { + compatible = "infineon,peb2466"; + reg = <0>; + spi-max-frequency = <8192000>; + reset-gpios = <&gpio 10 GPIO_ACTIVE_LOW>; + #sound-dai-cells = <0>; + gpio-controller; + #gpio-cells = <2>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/irondevice,sma1303.yaml b/Documentation/devicetree/bindings/sound/irondevice,sma1303.yaml new file mode 100644 index 000000000000..b36c35e5da1a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/irondevice,sma1303.yaml @@ -0,0 +1,48 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/irondevice,sma1303.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Iron Device SMA1303 Audio Amplifier + +maintainers: + - Kiseok Jo <kiseok.jo@irondevice.com> + +description: + SMA1303 digital class-D audio amplifier + with an integrated boost converter. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - irondevice,sma1303 + + reg: + maxItems: 1 + + '#sound-dai-cells': + const: 1 + +required: + - compatible + - reg + - '#sound-dai-cells' + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + amplifier@1e { + compatible = "irondevice,sma1303"; + reg = <0x1e>; + #sound-dai-cells = <1>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/max98090.txt b/Documentation/devicetree/bindings/sound/max98090.txt deleted file mode 100644 index 39d640294c62..000000000000 --- a/Documentation/devicetree/bindings/sound/max98090.txt +++ /dev/null @@ -1,59 +0,0 @@ -MAX98090 audio CODEC - -This device supports I2C only. - -Required properties: - -- compatible : "maxim,max98090" or "maxim,max98091". - -- reg : The I2C address of the device. - -- interrupts : The CODEC's interrupt output. - -Optional properties: - -- clocks: The phandle of the master clock to the CODEC - -- clock-names: Should be "mclk" - -- #sound-dai-cells : should be 0. - -- maxim,dmic-freq: Frequency at which to clock DMIC - -- maxim,micbias: Micbias voltage applies to the analog mic, valid voltages value are: - 0 - 2.2v - 1 - 2.55v - 2 - 2.4v - 3 - 2.8v - -Pins on the device (for linking into audio routes): - - * MIC1 - * MIC2 - * DMICL - * DMICR - * IN1 - * IN2 - * IN3 - * IN4 - * IN5 - * IN6 - * IN12 - * IN34 - * IN56 - * HPL - * HPR - * SPKL - * SPKR - * RCVL - * RCVR - * MICBIAS - -Example: - -audio-codec@10 { - compatible = "maxim,max98090"; - reg = <0x10>; - interrupt-parent = <&gpio>; - interrupts = <TEGRA_GPIO(H, 4) IRQ_TYPE_LEVEL_HIGH>; -}; diff --git a/Documentation/devicetree/bindings/sound/max98095.txt b/Documentation/devicetree/bindings/sound/max98095.txt deleted file mode 100644 index 318a4c82f17f..000000000000 --- a/Documentation/devicetree/bindings/sound/max98095.txt +++ /dev/null @@ -1,22 +0,0 @@ -MAX98095 audio CODEC - -This device supports I2C only. - -Required properties: - -- compatible : "maxim,max98095". - -- reg : The I2C address of the device. - -Optional properties: - -- clocks: The phandle of the master clock to the CODEC - -- clock-names: Should be "mclk" - -Example: - -max98095: codec@11 { - compatible = "maxim,max98095"; - reg = <0x11>; -}; diff --git a/Documentation/devicetree/bindings/sound/maxim,max98090.yaml b/Documentation/devicetree/bindings/sound/maxim,max98090.yaml new file mode 100644 index 000000000000..65e4c516912f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/maxim,max98090.yaml @@ -0,0 +1,84 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/maxim,max98090.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Maxim Integrated MAX98090/MAX98091 audio codecs + +maintainers: + - Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org> + +description: | + Pins on the device (for linking into audio routes): + MIC1, MIC2, DMICL, DMICR, IN1, IN2, IN3, IN4, IN5, IN6, IN12, IN34, IN56, + HPL, HPR, SPKL, SPKR, RCVL, RCVR, MICBIAS + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - maxim,max98090 + - maxim,max98091 + + reg: + maxItems: 1 + + clocks: + items: + - description: master clock + + clock-names: + items: + - const: mclk + + interrupts: + maxItems: 1 + + maxim,dmic-freq: + $ref: /schemas/types.yaml#/definitions/uint32 + default: 2500000 + description: + DMIC clock frequency + + maxim,micbias: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [ 0, 1, 2, 3 ] + default: 3 + description: | + Micbias voltage applied to the analog mic, valid voltages value are: + 0 - 2.2v + 1 - 2.55v + 2 - 2.4v + 3 - 2.8v + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - reg + - interrupts + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/irq.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + + audio-codec@10 { + compatible = "maxim,max98090"; + reg = <0x10>; + interrupt-parent = <&gpx3>; + interrupts = <2 IRQ_TYPE_EDGE_FALLING>; + clocks = <&i2s0 0>; + clock-names = "mclk"; + #sound-dai-cells = <0>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/maxim,max98095.yaml b/Documentation/devicetree/bindings/sound/maxim,max98095.yaml new file mode 100644 index 000000000000..77544a9e1587 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/maxim,max98095.yaml @@ -0,0 +1,54 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/maxim,max98095.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Maxim Integrated MAX98095 audio codec + +maintainers: + - Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - maxim,max98095 + + reg: + maxItems: 1 + + clocks: + items: + - description: master clock + + clock-names: + items: + - const: mclk + + '#sound-dai-cells': + const: 1 + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/irq.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + + audio-codec@11 { + compatible = "maxim,max98095"; + reg = <0x11>; + clocks = <&i2s0 0>; + clock-names = "mclk"; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/mediatek,mt8188-afe.yaml b/Documentation/devicetree/bindings/sound/mediatek,mt8188-afe.yaml new file mode 100644 index 000000000000..82ccb32f08f2 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mediatek,mt8188-afe.yaml @@ -0,0 +1,208 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/mediatek,mt8188-afe.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: MediaTek AFE PCM controller for mt8188 + +maintainers: + - Trevor Wu <trevor.wu@mediatek.com> + +properties: + compatible: + const: mediatek,mt8188-afe + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + resets: + maxItems: 1 + + reset-names: + const: audiosys + + mediatek,topckgen: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of the mediatek topckgen controller + + power-domains: + maxItems: 1 + + clocks: + items: + - description: 26M clock + - description: audio pll1 clock + - description: audio pll2 clock + - description: clock divider for i2si1_mck + - description: clock divider for i2si2_mck + - description: clock divider for i2so1_mck + - description: clock divider for i2so2_mck + - description: clock divider for dptx_mck + - description: a1sys hoping clock + - description: audio intbus clock + - description: audio hires clock + - description: audio local bus clock + - description: mux for dptx_mck + - description: mux for i2so1_mck + - description: mux for i2so2_mck + - description: mux for i2si1_mck + - description: mux for i2si2_mck + - description: audio 26m clock + + clock-names: + items: + - const: clk26m + - const: apll1 + - const: apll2 + - const: apll12_div0 + - const: apll12_div1 + - const: apll12_div2 + - const: apll12_div3 + - const: apll12_div9 + - const: a1sys_hp_sel + - const: aud_intbus_sel + - const: audio_h_sel + - const: audio_local_bus_sel + - const: dptx_m_sel + - const: i2so1_m_sel + - const: i2so2_m_sel + - const: i2si1_m_sel + - const: i2si2_m_sel + - const: adsp_audio_26m + + mediatek,etdm-in1-cowork-source: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + etdm modules can share the same external clock pin. Specify + which etdm clock source is required by this etdm in module. + enum: + - 1 # etdm2_in + - 2 # etdm1_out + - 3 # etdm2_out + + mediatek,etdm-in2-cowork-source: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + etdm modules can share the same external clock pin. Specify + which etdm clock source is required by this etdm in module. + enum: + - 0 # etdm1_in + - 2 # etdm1_out + - 3 # etdm2_out + + mediatek,etdm-out1-cowork-source: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + etdm modules can share the same external clock pin. Specify + which etdm clock source is required by this etdm out module. + enum: + - 0 # etdm1_in + - 1 # etdm2_in + - 3 # etdm2_out + + mediatek,etdm-out2-cowork-source: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + etdm modules can share the same external clock pin. Specify + which etdm clock source is required by this etdm out module. + enum: + - 0 # etdm1_in + - 1 # etdm2_in + - 2 # etdm1_out + +patternProperties: + "^mediatek,etdm-in[1-2]-chn-disabled$": + $ref: /schemas/types.yaml#/definitions/uint8-array + minItems: 1 + maxItems: 16 + description: + This is a list of channel IDs which should be disabled. + By default, all data received from ETDM pins will be outputed to + memory. etdm in supports disable_out in direct mode(w/o interconn), + so user can disable the specified channels by the property. + uniqueItems: true + items: + minimum: 0 + maximum: 15 + + "^mediatek,etdm-in[1-2]-multi-pin-mode$": + type: boolean + description: if present, the etdm data mode is I2S. + + "^mediatek,etdm-out[1-3]-multi-pin-mode$": + type: boolean + description: if present, the etdm data mode is I2S. + +required: + - compatible + - reg + - interrupts + - resets + - reset-names + - mediatek,topckgen + - power-domains + - clocks + - clock-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + + afe@10b10000 { + compatible = "mediatek,mt8188-afe"; + reg = <0x10b10000 0x10000>; + interrupts = <GIC_SPI 822 IRQ_TYPE_LEVEL_HIGH 0>; + resets = <&watchdog 14>; + reset-names = "audiosys"; + mediatek,topckgen = <&topckgen>; + power-domains = <&spm 13>; //MT8188_POWER_DOMAIN_AUDIO + mediatek,etdm-in2-cowork-source = <2>; + mediatek,etdm-out2-cowork-source = <0>; + mediatek,etdm-in1-multi-pin-mode; + mediatek,etdm-in1-chn-disabled = /bits/ 8 <0x0 0x2>; + clocks = <&clk26m>, + <&apmixedsys 9>, //CLK_APMIXED_APLL1 + <&apmixedsys 10>, //CLK_APMIXED_APLL2 + <&topckgen 186>, //CLK_TOP_APLL12_CK_DIV0 + <&topckgen 187>, //CLK_TOP_APLL12_CK_DIV1 + <&topckgen 188>, //CLK_TOP_APLL12_CK_DIV2 + <&topckgen 189>, //CLK_TOP_APLL12_CK_DIV3 + <&topckgen 191>, //CLK_TOP_APLL12_CK_DIV9 + <&topckgen 83>, //CLK_TOP_A1SYS_HP + <&topckgen 31>, //CLK_TOP_AUD_INTBUS + <&topckgen 32>, //CLK_TOP_AUDIO_H + <&topckgen 69>, //CLK_TOP_AUDIO_LOCAL_BUS + <&topckgen 81>, //CLK_TOP_DPTX + <&topckgen 77>, //CLK_TOP_I2SO1 + <&topckgen 78>, //CLK_TOP_I2SO2 + <&topckgen 79>, //CLK_TOP_I2SI1 + <&topckgen 80>, //CLK_TOP_I2SI2 + <&adsp_audio26m 0>; //CLK_AUDIODSP_AUDIO26M + clock-names = "clk26m", + "apll1", + "apll2", + "apll12_div0", + "apll12_div1", + "apll12_div2", + "apll12_div3", + "apll12_div9", + "a1sys_hp_sel", + "aud_intbus_sel", + "audio_h_sel", + "audio_local_bus_sel", + "dptx_m_sel", + "i2so1_m_sel", + "i2so2_m_sel", + "i2si1_m_sel", + "i2si2_m_sel", + "adsp_audio_26m"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/mediatek,mt8188-mt6359.yaml b/Documentation/devicetree/bindings/sound/mediatek,mt8188-mt6359.yaml new file mode 100644 index 000000000000..6640272b3f4f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mediatek,mt8188-mt6359.yaml @@ -0,0 +1,97 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/mediatek,mt8188-mt6359.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: MediaTek MT8188 ASoC sound card + +maintainers: + - Trevor Wu <trevor.wu@mediatek.com> + +properties: + compatible: + const: mediatek,mt8188-mt6359-evb + + model: + $ref: /schemas/types.yaml#/definitions/string + description: User specified audio sound card name + + audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: + A list of the connections between audio components. Each entry is a + sink/source pair of strings. Valid names could be the input or output + widgets of audio components, power supplies, MicBias of codec and the + software switch. + + mediatek,platform: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of MT8188 ASoC platform. + +patternProperties: + "^dai-link-[0-9]+$": + type: object + description: + Container for dai-link level properties and CODEC sub-nodes. + + properties: + link-name: + description: + This property corresponds to the name of the BE dai-link to which + we are going to update parameters in this node. + items: + enum: + - ADDA_BE + - 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 + + additionalProperties: false + + required: + - link-name + - codec + +additionalProperties: false + +required: + - compatible + - mediatek,platform + +examples: + - | + sound { + compatible = "mediatek,mt8188-mt6359-evb"; + mediatek,platform = <&afe>; + pinctrl-names = "default"; + pinctrl-0 = <&aud_pins_default>; + audio-routing = + "Headphone", "Headphone L", + "Headphone", "Headphone R", + "AIN1", "Headset Mic"; + dai-link-0 { + link-name = "ETDM3_OUT_BE"; + + codec { + sound-dai = <&hdmi0>; + }; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/mchp,i2s-mcc.yaml b/Documentation/devicetree/bindings/sound/microchip,sama7g5-i2smcc.yaml index 621022872c8d..651f61c7c25a 100644 --- a/Documentation/devicetree/bindings/sound/mchp,i2s-mcc.yaml +++ b/Documentation/devicetree/bindings/sound/microchip,sama7g5-i2smcc.yaml @@ -1,7 +1,7 @@ # SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) %YAML 1.2 --- -$id: http://devicetree.org/schemas/sound/mchp,i2s-mcc.yaml# +$id: http://devicetree.org/schemas/sound/microchip,sama7g5-i2smcc.yaml# $schema: http://devicetree.org/meta-schemas/core.yaml# title: Microchip I2S Multi-Channel Controller diff --git a/Documentation/devicetree/bindings/sound/microchip,pdmc.yaml b/Documentation/devicetree/bindings/sound/microchip,sama7g5-pdmc.yaml index c37b89d94c12..c4cf1e5ab84b 100644 --- a/Documentation/devicetree/bindings/sound/microchip,pdmc.yaml +++ b/Documentation/devicetree/bindings/sound/microchip,sama7g5-pdmc.yaml @@ -1,7 +1,7 @@ # SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) %YAML 1.2 --- -$id: http://devicetree.org/schemas/sound/microchip,pdmc.yaml# +$id: http://devicetree.org/schemas/sound/microchip,sama7g5-pdmc.yaml# $schema: http://devicetree.org/meta-schemas/core.yaml# title: Microchip Pulse Density Microphone Controller diff --git a/Documentation/devicetree/bindings/sound/mchp,spdifrx.yaml b/Documentation/devicetree/bindings/sound/microchip,sama7g5-spdifrx.yaml index 70a47c6823b1..2f43c684ab88 100644 --- a/Documentation/devicetree/bindings/sound/mchp,spdifrx.yaml +++ b/Documentation/devicetree/bindings/sound/microchip,sama7g5-spdifrx.yaml @@ -1,7 +1,7 @@ # SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) %YAML 1.2 --- -$id: http://devicetree.org/schemas/sound/mchp,spdifrx.yaml# +$id: http://devicetree.org/schemas/sound/microchip,sama7g5-spdifrx.yaml# $schema: http://devicetree.org/meta-schemas/core.yaml# title: Microchip S/PDIF Rx Controller diff --git a/Documentation/devicetree/bindings/sound/mchp,spdiftx.yaml b/Documentation/devicetree/bindings/sound/microchip,sama7g5-spdiftx.yaml index c383162140bb..4702c528700d 100644 --- a/Documentation/devicetree/bindings/sound/mchp,spdiftx.yaml +++ b/Documentation/devicetree/bindings/sound/microchip,sama7g5-spdiftx.yaml @@ -1,7 +1,7 @@ # SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) %YAML 1.2 --- -$id: http://devicetree.org/schemas/sound/mchp,spdiftx.yaml# +$id: http://devicetree.org/schemas/sound/microchip,sama7g5-spdiftx.yaml# $schema: http://devicetree.org/meta-schemas/core.yaml# title: Microchip S/PDIF Tx Controller diff --git a/Documentation/devicetree/bindings/sound/nau8822.txt b/Documentation/devicetree/bindings/sound/nau8822.txt deleted file mode 100644 index a471d162d4e5..000000000000 --- a/Documentation/devicetree/bindings/sound/nau8822.txt +++ /dev/null @@ -1,16 +0,0 @@ -NAU8822 audio CODEC - -This device supports I2C only. - -Required properties: - - - compatible : "nuvoton,nau8822" - - - reg : the I2C address of the device. - -Example: - -codec: nau8822@1a { - compatible = "nuvoton,nau8822"; - reg = <0x1a>; -}; diff --git a/Documentation/devicetree/bindings/sound/nuvoton,nau8822.yaml b/Documentation/devicetree/bindings/sound/nuvoton,nau8822.yaml new file mode 100644 index 000000000000..65105402a53d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nuvoton,nau8822.yaml @@ -0,0 +1,46 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nuvoton,nau8822.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NAU8822 audio CODEC + +description: | + 24 bit stereo audio codec with speaker driver. + This device supports I2C/SPI. + +maintainers: + - David Lin <CTLIN0@nuvoton.com> + +properties: + compatible: + enum: + - nuvoton,nau8822 + + reg: + maxItems: 1 + + nuvoton,spk-btl: + description: + If set, configure the two loudspeaker outputs as a Bridge Tied Load output + to drive a high power external loudspeaker. + $ref: /schemas/types.yaml#/definitions/flag + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@1a { + compatible = "nuvoton,nau8822"; + reg = <0x1a>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml index bb42220916b3..6cc8f86c7531 100644 --- a/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml @@ -35,7 +35,7 @@ properties: clocks: minItems: 3 - maxItems: 7 + maxItems: 10 clock-names: minItems: 1 @@ -65,6 +65,9 @@ properties: power-domain-names: maxItems: 1 + required-opps: + maxItems: 1 + '#sound-dai-cells': const: 1 @@ -75,7 +78,7 @@ properties: const: 0 patternProperties: - "^dai-link@[0-9a-f]$": + "^dai-link@[0-9a-f]+$": type: object description: | LPASS CPU dai node for each I2S device or Soundwire device. Bindings of each node @@ -121,6 +124,8 @@ allOf: then: properties: + clocks: + maxItems: 3 clock-names: items: - const: ahbix-clk @@ -135,6 +140,9 @@ allOf: then: properties: + clocks: + minItems: 7 + maxItems: 7 clock-names: items: - const: ahbix-clk @@ -153,33 +161,31 @@ allOf: then: properties: + clocks: + minItems: 6 + maxItems: 6 clock-names: - oneOf: - - items: #for I2S - - const: pcnoc-sway-clk - - const: audio-core - - const: mclk0 - - const: pcnoc-mport-clk - - const: mi2s-bit-clk0 - - const: mi2s-bit-clk1 - - items: #for HDMI - - const: pcnoc-sway-clk - - const: audio-core - - const: pcnoc-mport-clk + items: + - const: pcnoc-sway-clk + - const: audio-core + - const: mclk0 + - const: pcnoc-mport-clk + - const: mi2s-bit-clk0 + - const: mi2s-bit-clk1 + reg: + minItems: 2 + maxItems: 2 reg-names: - anyOf: - - items: #for I2S - - const: lpass-lpaif - - items: #for I2S and HDMI - - const: lpass-hdmiif - - const: lpass-lpaif + items: + - const: lpass-hdmiif + - const: lpass-lpaif + interrupts: + minItems: 2 + maxItems: 2 interrupt-names: - anyOf: - - items: #for I2S - - const: lpass-irq-lpaif - - items: #for I2S and HDMI - - const: lpass-irq-lpaif - - const: lpass-irq-hdmi + items: + - const: lpass-irq-lpaif + - const: lpass-irq-hdmi required: - iommus - power-domains @@ -192,54 +198,44 @@ allOf: then: properties: + clocks: + minItems: 10 + maxItems: 10 clock-names: - oneOf: - - items: #for I2S - - const: aon_cc_audio_hm_h - - const: audio_cc_ext_mclk0 - - const: core_cc_sysnoc_mport_core - - const: core_cc_ext_if0_ibit - - const: core_cc_ext_if1_ibit - - items: #for Soundwire - - const: aon_cc_audio_hm_h - - const: audio_cc_codec_mem - - const: audio_cc_codec_mem0 - - const: audio_cc_codec_mem1 - - const: audio_cc_codec_mem2 - - const: aon_cc_va_mem0 - - items: #for HDMI - - const: core_cc_sysnoc_mport_core - + items: + - const: aon_cc_audio_hm_h + - const: audio_cc_ext_mclk0 + - const: core_cc_sysnoc_mport_core + - const: core_cc_ext_if0_ibit + - const: core_cc_ext_if1_ibit + - const: audio_cc_codec_mem + - const: audio_cc_codec_mem0 + - const: audio_cc_codec_mem1 + - const: audio_cc_codec_mem2 + - const: aon_cc_va_mem0 + reg: + minItems: 6 + maxItems: 6 reg-names: - anyOf: - - items: #for I2S - - const: lpass-lpaif - - items: #for I2S and HDMI - - const: lpass-hdmiif - - const: lpass-lpaif - - items: #for I2S, soundwire and HDMI - - const: lpass-hdmiif - - const: lpass-lpaif - - const: lpass-rxtx-cdc-dma-lpm - - const: lpass-rxtx-lpaif - - const: lpass-va-lpaif - - const: lpass-va-cdc-dma-lpm + items: + - const: lpass-hdmiif + - const: lpass-lpaif + - const: lpass-rxtx-cdc-dma-lpm + - const: lpass-rxtx-lpaif + - const: lpass-va-lpaif + - const: lpass-va-cdc-dma-lpm + interrupts: + minItems: 4 + maxItems: 4 interrupt-names: - anyOf: - - items: #for I2S - - const: lpass-irq-lpaif - - items: #for I2S and HDMI - - const: lpass-irq-lpaif - - const: lpass-irq-hdmi - - items: #for I2S, soundwire and HDMI - - const: lpass-irq-lpaif - - const: lpass-irq-hdmi - - const: lpass-irq-vaif - - const: lpass-irq-rxtxif + items: + - const: lpass-irq-lpaif + - const: lpass-irq-hdmi + - const: lpass-irq-vaif + - const: lpass-irq-rxtxif power-domain-names: - allOf: - - items: - - const: lcx + items: + - const: lcx required: - iommus diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-va-macro.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-va-macro.yaml index 26f0343b5aac..0a3c688ef1ec 100644 --- a/Documentation/devicetree/bindings/sound/qcom,lpass-va-macro.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-va-macro.yaml @@ -36,7 +36,7 @@ properties: oneOf: - items: #for ADSP based platforms - const: mclk - - const: core + - const: macro - const: dcodec - items: #for ADSP bypass based platforms - const: mclk @@ -77,7 +77,7 @@ examples: clocks = <&aoncc 0>, <&q6afecc LPASS_HW_MACRO_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>, <&q6afecc LPASS_HW_DCODEC_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>; - clock-names = "mclk", "core", "dcodec"; + clock-names = "mclk", "macro", "dcodec"; clock-output-names = "fsgen"; qcom,dmic-sample-rate = <600000>; vdd-micb-supply = <&vreg_s4a_1p8>; diff --git a/Documentation/devicetree/bindings/sound/qcom,q6apm-dai.yaml b/Documentation/devicetree/bindings/sound/qcom,q6apm-dai.yaml index a53c9ef938fa..cdbb4096fa44 100644 --- a/Documentation/devicetree/bindings/sound/qcom,q6apm-dai.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,q6apm-dai.yaml @@ -17,7 +17,8 @@ properties: const: qcom,q6apm-dais iommus: - maxItems: 1 + minItems: 1 + maxItems: 2 required: - compatible diff --git a/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml index 70080d04ddc9..262de7a60a73 100644 --- a/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml @@ -15,16 +15,20 @@ description: properties: compatible: - enum: - - lenovo,yoga-c630-sndcard - - qcom,apq8016-sbc-sndcard - - qcom,db845c-sndcard - - qcom,msm8916-qdsp6-sndcard - - qcom,qrb5165-rb5-sndcard - - qcom,sc8280xp-sndcard - - qcom,sdm845-sndcard - - qcom,sm8250-sndcard - - qcom,sm8450-sndcard + oneOf: + - items: + - enum: + - lenovo,yoga-c630-sndcard + - qcom,db845c-sndcard + - const: qcom,sdm845-sndcard + - enum: + - qcom,apq8016-sbc-sndcard + - qcom,msm8916-qdsp6-sndcard + - qcom,qrb5165-rb5-sndcard + - qcom,sc8280xp-sndcard + - qcom,sdm845-sndcard + - qcom,sm8250-sndcard + - qcom,sm8450-sndcard audio-routing: $ref: /schemas/types.yaml#/definitions/non-unique-string-array diff --git a/Documentation/devicetree/bindings/sound/qcom,wcd934x.yaml b/Documentation/devicetree/bindings/sound/qcom,wcd934x.yaml index 184e8ccbdd13..ea09590bfa30 100644 --- a/Documentation/devicetree/bindings/sound/qcom,wcd934x.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,wcd934x.yaml @@ -28,7 +28,9 @@ properties: description: GPIO spec for reset line to use maxItems: 1 - slim-ifc-dev: true + slim-ifc-dev: + description: IFC device interface + $ref: /schemas/types.yaml#/definitions/phandle clocks: maxItems: 1 @@ -147,21 +149,49 @@ patternProperties: required: - compatible - reg - - reset-gpios - - slim-ifc-dev - - interrupts - - interrupt-controller - - clock-frequency - - clock-output-names - - qcom,micbias1-microvolt - - qcom,micbias2-microvolt - - qcom,micbias3-microvolt - - qcom,micbias4-microvolt - - "#interrupt-cells" - - "#clock-cells" - - "#sound-dai-cells" - - "#address-cells" - - "#size-cells" + +allOf: + - if: + required: + - slim-ifc-dev + then: + required: + - reset-gpios + - slim-ifc-dev + - interrupt-controller + - clock-frequency + - clock-output-names + - qcom,micbias1-microvolt + - qcom,micbias2-microvolt + - qcom,micbias3-microvolt + - qcom,micbias4-microvolt + - "#interrupt-cells" + - "#clock-cells" + - "#sound-dai-cells" + - "#address-cells" + - "#size-cells" + oneOf: + - required: + - interrupts-extended + - required: + - interrupts + else: + properties: + reset-gpios: false + slim-ifc-dev: false + interrupts: false + interrupt-controller: false + clock-frequency: false + clock-output-names: false + qcom,micbias1-microvolt: false + qcom,micbias2-microvolt: false + qcom,micbias3-microvolt: false + qcom,micbias4-microvolt: false + "#interrupt-cells": false + "#clock-cells": false + "#sound-dai-cells": false + "#address-cells": false + "#size-cells": false additionalProperties: false diff --git a/Documentation/devicetree/bindings/sound/qcom,wsa881x.yaml b/Documentation/devicetree/bindings/sound/qcom,wsa881x.yaml index d702b489320f..ac03672ebf6d 100644 --- a/Documentation/devicetree/bindings/sound/qcom,wsa881x.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,wsa881x.yaml @@ -15,6 +15,9 @@ description: | Their primary operating mode uses a SoundWire digital audio interface. This binding is for SoundWire interface. +allOf: + - $ref: dai-common.yaml# + properties: compatible: const: sdw10217201000 @@ -39,7 +42,7 @@ required: - "#thermal-sensor-cells" - "#sound-dai-cells" -additionalProperties: false +unevaluatedProperties: false examples: - | diff --git a/Documentation/devicetree/bindings/sound/renesas,idt821034.yaml b/Documentation/devicetree/bindings/sound/renesas,idt821034.yaml new file mode 100644 index 000000000000..a2b92dba5529 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/renesas,idt821034.yaml @@ -0,0 +1,75 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/renesas,idt821034.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Renesas IDT821034 codec device + +maintainers: + - Herve Codina <herve.codina@bootlin.com> + +description: | + The IDT821034 codec is a four channel PCM codec with onchip filters and + programmable gain setting. + + The time-slots used by the codec must be set and so, the properties + 'dai-tdm-slot-num', 'dai-tdm-slot-width', 'dai-tdm-slot-tx-mask' and + 'dai-tdm-slot-rx-mask' must be present in the ALSA sound card node for + sub-nodes that involve the codec. The codec uses one 8bit time-slot per + channel. + 'dai-tdm-tdm-slot-with' must be set to 8. + + The IDT821034 codec also supports 5 gpios (SLIC signals) per channel. + +allOf: + - $ref: /schemas/spi/spi-peripheral-props.yaml# + - $ref: dai-common.yaml# + +properties: + compatible: + const: renesas,idt821034 + + reg: + description: + SPI device address. + maxItems: 1 + + spi-max-frequency: + maximum: 8192000 + + spi-cpha: true + + '#sound-dai-cells': + const: 0 + + '#gpio-cells': + const: 2 + + gpio-controller: true + +required: + - compatible + - reg + - spi-cpha + - '#sound-dai-cells' + - gpio-controller + - '#gpio-cells' + +unevaluatedProperties: false + +examples: + - | + spi { + #address-cells = <1>; + #size-cells = <0>; + audio-codec@0 { + compatible = "renesas,idt821034"; + reg = <0>; + spi-max-frequency = <8192000>; + spi-cpha; + #sound-dai-cells = <0>; + gpio-controller; + #gpio-cells = <2>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml index cb90463c7297..12ccf29338d9 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml @@ -18,8 +18,7 @@ properties: - enum: - renesas,rcar_sound-r8a7778 # R-Car M1A - renesas,rcar_sound-r8a7779 # R-Car H1 - - enum: - - renesas,rcar_sound-gen1 + - const: renesas,rcar_sound-gen1 # for Gen2 SoC - items: - enum: @@ -32,8 +31,7 @@ properties: - renesas,rcar_sound-r8a7791 # R-Car M2-W - renesas,rcar_sound-r8a7793 # R-Car M2-N - renesas,rcar_sound-r8a7794 # R-Car E2 - - enum: - - renesas,rcar_sound-gen2 + - const: renesas,rcar_sound-gen2 # for Gen3 SoC - items: - enum: @@ -47,14 +45,16 @@ properties: - renesas,rcar_sound-r8a77965 # R-Car M3-N - renesas,rcar_sound-r8a77990 # R-Car E3 - renesas,rcar_sound-r8a77995 # R-Car D3 - - enum: - - renesas,rcar_sound-gen3 - # for Generic + - const: renesas,rcar_sound-gen3 + # for Gen4 SoC - items: - - enum: - - renesas,rcar_sound-gen1 - - renesas,rcar_sound-gen2 - - renesas,rcar_sound-gen3 + - const: renesas,rcar_sound-r8a779g0 # R-Car V4H + - const: renesas,rcar_sound-gen4 + # for Generic + - enum: + - renesas,rcar_sound-gen1 + - renesas,rcar_sound-gen2 + - renesas,rcar_sound-gen3 reg: minItems: 1 @@ -68,6 +68,7 @@ properties: description: | it must be 0 if your system is using single DAI it must be 1 if your system is using multi DAIs + This is used on simple-audio-card enum: [0, 1] "#clock-cells": @@ -113,15 +114,34 @@ properties: - pattern: '^clk_(a|b|c|i)$' ports: - $ref: /schemas/graph.yaml#/properties/ports + $ref: audio-graph-port.yaml#/definitions/port-base + unevaluatedProperties: false patternProperties: '^port(@[0-9a-f]+)?$': - $ref: audio-graph-port.yaml# + $ref: audio-graph-port.yaml#/definitions/port-base unevaluatedProperties: false + patternProperties: + "^endpoint(@[0-9a-f]+)?": + $ref: audio-graph-port.yaml#/definitions/endpoint-base + properties: + playback: + $ref: /schemas/types.yaml#/definitions/phandle-array + capture: + $ref: /schemas/types.yaml#/definitions/phandle-array + unevaluatedProperties: false port: - $ref: audio-graph-port.yaml# + $ref: audio-graph-port.yaml#/definitions/port-base unevaluatedProperties: false + patternProperties: + "^endpoint(@[0-9a-f]+)?": + $ref: audio-graph-port.yaml#/definitions/endpoint-base + properties: + playback: + $ref: /schemas/types.yaml#/definitions/phandle-array + capture: + $ref: /schemas/types.yaml#/definitions/phandle-array + unevaluatedProperties: false rcar_sound,dvc: description: DVC subnode. @@ -178,10 +198,6 @@ properties: enum: - tx - rx - required: - - interrupts - - dmas - - dma-names additionalProperties: false rcar_sound,ssiu: @@ -240,8 +256,6 @@ properties: $ref: /schemas/types.yaml#/definitions/flag required: - interrupts - - dmas - - dma-names additionalProperties: false # For DAI base @@ -271,7 +285,6 @@ required: - reg-names - clocks - clock-names - - "#sound-dai-cells" allOf: - $ref: dai-common.yaml# @@ -285,7 +298,6 @@ allOf: reg: maxItems: 3 reg-names: - maxItems: 3 items: enum: - scu @@ -294,9 +306,8 @@ allOf: else: properties: reg: - maxItems: 5 + minItems: 5 reg-names: - maxItems: 5 items: enum: - scu diff --git a/Documentation/devicetree/bindings/sound/rt5640.txt b/Documentation/devicetree/bindings/sound/rt5640.txt index ff1228713f7e..0c398581d52b 100644 --- a/Documentation/devicetree/bindings/sound/rt5640.txt +++ b/Documentation/devicetree/bindings/sound/rt5640.txt @@ -20,6 +20,9 @@ Optional properties: - realtek,in3-differential Boolean. Indicate MIC1/2/3 input are differential, rather than single-ended. +- realtek,lout-differential + Boolean. Indicate LOUT output is differential, rather than stereo. + - realtek,ldo1-en-gpios : The GPIO that controls the CODEC's LDO1_EN pin. - realtek,dmic1-data-pin diff --git a/Documentation/devicetree/bindings/sound/samsung,odroid.yaml b/Documentation/devicetree/bindings/sound/samsung,odroid.yaml index 7b4e08ddef6a..7774543b8819 100644 --- a/Documentation/devicetree/bindings/sound/samsung,odroid.yaml +++ b/Documentation/devicetree/bindings/sound/samsung,odroid.yaml @@ -43,9 +43,10 @@ properties: type: object properties: sound-dai: + minItems: 1 items: - - description: phandle of the MAX98090 CODEC - description: phandle of the HDMI IP block node + - description: phandle of the MAX98090 CODEC samsung,audio-routing: $ref: /schemas/types.yaml#/definitions/non-unique-string-array diff --git a/Documentation/devicetree/bindings/sound/samsung-i2s.yaml b/Documentation/devicetree/bindings/sound/samsung-i2s.yaml index 8d5dcf9cd43e..30b3b6e9824b 100644 --- a/Documentation/devicetree/bindings/sound/samsung-i2s.yaml +++ b/Documentation/devicetree/bindings/sound/samsung-i2s.yaml @@ -37,12 +37,20 @@ properties: samsung,exynos7-i2s1: I2S1 on previous samsung platforms supports stereo channels. Exynos7 I2S1 upgraded to 5.1 multichannel with slightly modified bit offsets. + + tesla,fsd-i2s: for 8/16/24bit stereo channel I2S for playback and + capture, secondary FIFO using external DMA, s/w reset control, + internal mux for root clock source with all root clock sampling + frequencies supported by Exynos7 I2S and 7.1 channel TDM support + for playback and capture TDM (Time division multiplexing) to allow + transfer of multiple channel audio data on single data line. enum: - samsung,s3c6410-i2s - samsung,s5pv210-i2s - samsung,exynos5420-i2s - samsung,exynos7-i2s - samsung,exynos7-i2s1 + - tesla,fsd-i2s '#address-cells': const: 1 @@ -67,9 +75,6 @@ properties: - const: rx - const: tx-sec - assigned-clock-parents: true - assigned-clocks: true - clocks: minItems: 1 maxItems: 3 diff --git a/Documentation/devicetree/bindings/sound/simple-card.yaml b/Documentation/devicetree/bindings/sound/simple-card.yaml index ed19899bc94b..f0d81bfe2598 100644 --- a/Documentation/devicetree/bindings/sound/simple-card.yaml +++ b/Documentation/devicetree/bindings/sound/simple-card.yaml @@ -205,6 +205,8 @@ patternProperties: $ref: "#/definitions/dai" "^simple-audio-card,codec(@[0-9a-f]+)?$": $ref: "#/definitions/dai" + "^simple-audio-card,plat(@[0-9a-f]+)?$": + $ref: "#/definitions/dai" "^simple-audio-card,dai-link(@[0-9a-f]+)?$": description: | @@ -215,6 +217,10 @@ patternProperties: reg: maxItems: 1 + "#address-cells": + const: 1 + "#size-cells": + const: 0 # common properties frame-master: $ref: "#/definitions/frame-master" @@ -244,9 +250,9 @@ patternProperties: maxItems: 1 patternProperties: - "^cpu(@[0-9a-f]+)?": + "^cpu(-[0-9]+)?$": $ref: "#/definitions/dai" - "^codec(@[0-9a-f]+)?": + "^codec(-[0-9]+)?$": $ref: "#/definitions/dai" additionalProperties: false @@ -462,16 +468,16 @@ examples: convert-channels = <8>; /* TDM Split */ - sndcpu1: cpu0 { + sndcpu1: cpu-0 { sound-dai = <&rcar_sound 1>; }; - cpu1 { + cpu-1 { sound-dai = <&rcar_sound 2>; }; - cpu2 { + cpu-2 { sound-dai = <&rcar_sound 3>; }; - cpu3 { + cpu-3 { sound-dai = <&rcar_sound 4>; }; codec { diff --git a/Documentation/devicetree/bindings/sound/tas5720.txt b/Documentation/devicetree/bindings/sound/tas5720.txt index df99ca9451b0..7d851ae2bba2 100644 --- a/Documentation/devicetree/bindings/sound/tas5720.txt +++ b/Documentation/devicetree/bindings/sound/tas5720.txt @@ -6,11 +6,13 @@ audio playback. For more product information please see the links below: https://www.ti.com/product/TAS5720L https://www.ti.com/product/TAS5720M +https://www.ti.com/product/TAS5720A-Q1 https://www.ti.com/product/TAS5722L Required properties: - compatible : "ti,tas5720", + "ti,tas5720a-q1", "ti,tas5722" - reg : I2C slave address - dvdd-supply : phandle to a 3.3-V supply for the digital circuitry diff --git a/Documentation/devicetree/bindings/sound/ti,pcm3168a.txt b/Documentation/devicetree/bindings/sound/ti,pcm3168a.txt deleted file mode 100644 index a02ecaab5183..000000000000 --- a/Documentation/devicetree/bindings/sound/ti,pcm3168a.txt +++ /dev/null @@ -1,56 +0,0 @@ -Texas Instruments pcm3168a DT bindings - -This driver supports both SPI and I2C bus access for this codec - -Required properties: - - - compatible: "ti,pcm3168a" - - - clocks : Contains an entry for each entry in clock-names - - - clock-names : Includes the following entries: - "scki" The system clock - - - VDD1-supply : Digital power supply regulator 1 (+3.3V) - - - VDD2-supply : Digital power supply regulator 2 (+3.3V) - - - VCCAD1-supply : ADC power supply regulator 1 (+5V) - - - VCCAD2-supply : ADC power supply regulator 2 (+5V) - - - VCCDA1-supply : DAC power supply regulator 1 (+5V) - - - VCCDA2-supply : DAC power supply regulator 2 (+5V) - -For required properties on SPI/I2C, consult SPI/I2C device tree documentation - -Optional properties: - - - reset-gpios : Optional reset gpio line connected to RST pin of the codec. - The RST line is low active: - RST = low: device power-down - RST = high: device is enabled - -Examples: - -i2c0: i2c0@0 { - - ... - - pcm3168a: audio-codec@44 { - compatible = "ti,pcm3168a"; - reg = <0x44>; - reset-gpios = <&gpio0 4 GPIO_ACTIVE_LOW>; - clocks = <&clk_core CLK_AUDIO>; - clock-names = "scki"; - VDD1-supply = <&supply3v3>; - VDD2-supply = <&supply3v3>; - VCCAD1-supply = <&supply5v0>; - VCCAD2-supply = <&supply5v0>; - VCCDA1-supply = <&supply5v0>; - VCCDA2-supply = <&supply5v0>; - pinctrl-names = "default"; - pinctrl-0 = <&dac_clk_pin>; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/ti,pcm3168a.yaml b/Documentation/devicetree/bindings/sound/ti,pcm3168a.yaml new file mode 100644 index 000000000000..b6a4360ab845 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,pcm3168a.yaml @@ -0,0 +1,107 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,pcm3168a.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments PCM3168A Audio Codec + +maintainers: + - Damien Horsley <Damien.Horsley@imgtec.com> + - Geert Uytterhoeven <geert+renesas@glider.be> + - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> + +description: + The Texas Instruments PCM3168A is a 24-bit Multi-channel Audio CODEC with + 96/192kHz sampling rate, supporting both SPI and I2C bus access. + +properties: + compatible: + const: ti,pcm3168a + + reg: + maxItems: 1 + + clocks: + items: + - description: System clock input + + clock-names: + items: + - const: scki + + reset-gpios: + items: + - description: | + GPIO line connected to the active-low RST pin of the codec. + RST = low: device power-down + RST = high: device is enabled + + "#sound-dai-cells": + enum: [0, 1] + + VDD1-supply: + description: Digital power supply regulator 1 (+3.3V) + + VDD2-supply: + description: Digital power supply regulator 2 (+3.3V) + + VCCAD1-supply: + description: ADC power supply regulator 1 (+5V) + + VCCAD2-supply: + description: ADC power supply regulator 2 (+5V) + + VCCDA1-supply: + description: DAC power supply regulator 1 (+5V) + + VCCDA2-supply: + description: DAC power supply regulator 2 (+5V) + + ports: + $ref: audio-graph-port.yaml#/definitions/port-base + properties: + port@0: + $ref: audio-graph-port.yaml# + description: Audio input port. + + port@1: + $ref: audio-graph-port.yaml# + description: Audio output port. + +required: + - compatible + - reg + - clocks + - clock-names + - VDD1-supply + - VDD2-supply + - VCCAD1-supply + - VCCAD2-supply + - VCCDA1-supply + - VCCDA2-supply + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + + pcm3168a: audio-codec@44 { + compatible = "ti,pcm3168a"; + reg = <0x44>; + reset-gpios = <&gpio0 4 GPIO_ACTIVE_LOW>; + clocks = <&clk_core 42>; + clock-names = "scki"; + VDD1-supply = <&supply3v3>; + VDD2-supply = <&supply3v3>; + VCCAD1-supply = <&supply5v0>; + VCCAD2-supply = <&supply5v0>; + VCCDA1-supply = <&supply5v0>; + VCCDA2-supply = <&supply5v0>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/ti,tlv320aic3x.yaml b/Documentation/devicetree/bindings/sound/ti,tlv320aic3x.yaml new file mode 100644 index 000000000000..e8ca9f3369f8 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tlv320aic3x.yaml @@ -0,0 +1,165 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +# Copyright (C) 2022 Texas Instruments Incorporated +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,tlv320aic3x.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments TLV320AIC3x Codec + +description: | + TLV320AIC3x are a series of low-power stereo audio codecs with stereo + headphone amplifier, as well as multiple inputs and outputs programmable in + single-ended or fully differential configurations. + + The serial control bus supports SPI or I2C protocols, while the serial audio + data bus is programmable for I2S, left/right-justified, DSP, or TDM modes. + + The following pins can be referred in the sound node's audio routing property: + + CODEC output pins: + LLOUT + RLOUT + MONO_LOUT + HPLOUT + HPROUT + HPLCOM + HPRCOM + + CODEC input pins for TLV320AIC3104: + MIC2L + MIC2R + LINE1L + LINE1R + + CODEC input pins for other compatible codecs: + MIC3L + MIC3R + LINE1L + LINE2L + LINE1R + LINE2R + +maintainers: + - Jai Luthra <j-luthra@ti.com> + +properties: + compatible: + enum: + - ti,tlv320aic3x + - ti,tlv320aic33 + - ti,tlv320aic3007 + - ti,tlv320aic3106 + - ti,tlv320aic3104 + + reg: + maxItems: 1 + + reset-gpios: + maxItems: 1 + description: + GPIO specification for the active low RESET input. + + gpio-reset: + maxItems: 1 + description: + Deprecated, please use reset-gpios instead. + deprecated: true + + ai3x-gpio-func: + description: AIC3X_GPIO1 & AIC3X_GPIO2 Functionality + $ref: /schemas/types.yaml#/definitions/uint32-array + maxItems: 2 + + ai3x-micbias-vg: + description: MicBias required voltage. If node is omitted then MicBias is powered down. + $ref: /schemas/types.yaml#/definitions/uint32 + oneOf: + - const: 1 + description: MICBIAS output is powered to 2.0V. + - const: 2 + description: MICBIAS output is powered to 2.5V. + - const: 3 + description: MICBIAS output is connected to AVDD. + + ai3x-ocmv: + description: Output Common-Mode Voltage selection. + $ref: /schemas/types.yaml#/definitions/uint32 + oneOf: + - const: 0 + description: 1.35V + - const: 1 + description: 1.5V + - const: 2 + description: 1.65V + - const: 3 + description: 1.8V + + AVDD-supply: + description: Analog DAC voltage. + + IOVDD-supply: + description: I/O voltage. + + DRVDD-supply: + description: ADC analog and output driver voltage. + + DVDD-supply: + description: Digital core voltage. + + '#sound-dai-cells': + const: 0 + + clocks: + maxItems: 1 + + port: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + + tlv320aic3x_i2c: audio-codec@1b { + compatible = "ti,tlv320aic3x"; + reg = <0x1b>; + + reset-gpios = <&gpio1 17 GPIO_ACTIVE_LOW>; + + AVDD-supply = <®ulator>; + IOVDD-supply = <®ulator>; + DRVDD-supply = <®ulator>; + DVDD-supply = <®ulator>; + }; + }; + + - | + #include <dt-bindings/gpio/gpio.h> + spi { + #address-cells = <1>; + #size-cells = <0>; + + tlv320aic3x_spi: audio-codec@0 { + compatible = "ti,tlv320aic3x"; + reg = <0>; /* CS number */ + #sound-dai-cells = <0>; + + AVDD-supply = <®ulator>; + IOVDD-supply = <®ulator>; + DRVDD-supply = <®ulator>; + DVDD-supply = <®ulator>; + ai3x-ocmv = <0>; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt deleted file mode 100644 index 20931a63fd64..000000000000 --- a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt +++ /dev/null @@ -1,97 +0,0 @@ -Texas Instruments - tlv320aic3x Codec module - -The tlv320aic3x serial control bus communicates through both I2C and SPI bus protocols - -Required properties: - -- compatible - "string" - One of: - "ti,tlv320aic3x" - Generic TLV320AIC3x device - "ti,tlv320aic33" - TLV320AIC33 - "ti,tlv320aic3007" - TLV320AIC3007 - "ti,tlv320aic3106" - TLV320AIC3106 - "ti,tlv320aic3104" - TLV320AIC3104 - - -- reg - <int> - I2C slave address - - -Optional properties: - -- reset-gpios - GPIO specification for the active low RESET input. -- ai3x-gpio-func - <array of 2 int> - AIC3X_GPIO1 & AIC3X_GPIO2 Functionality - - Not supported on tlv320aic3104 -- ai3x-micbias-vg - MicBias Voltage required. - 1 - MICBIAS output is powered to 2.0V, - 2 - MICBIAS output is powered to 2.5V, - 3 - MICBIAS output is connected to AVDD, - If this node is not mentioned or if the value is incorrect, then MicBias - is powered down. -- ai3x-ocmv - Output Common-Mode Voltage selection: - 0 - 1.35V, - 1 - 1.5V, - 2 - 1.65V, - 3 - 1.8V -- AVDD-supply, IOVDD-supply, DRVDD-supply, DVDD-supply : power supplies for the - device as covered in Documentation/devicetree/bindings/regulator/regulator.txt - -Deprecated properties: - -- gpio-reset - gpio pin number used for codec reset - -CODEC output pins: - * LLOUT - * RLOUT - * MONO_LOUT - * HPLOUT - * HPROUT - * HPLCOM - * HPRCOM - -CODEC input pins for TLV320AIC3104: - * MIC2L - * MIC2R - * LINE1L - * LINE1R - -CODEC input pins for other compatible codecs: - * MIC3L - * MIC3R - * LINE1L - * LINE2L - * LINE1R - * LINE2R - -The pins can be used in referring sound node's audio-routing property. - -I2C example: - -#include <dt-bindings/gpio/gpio.h> - -tlv320aic3x: tlv320aic3x@1b { - compatible = "ti,tlv320aic3x"; - reg = <0x1b>; - - reset-gpios = <&gpio1 17 GPIO_ACTIVE_LOW>; - - AVDD-supply = <®ulator>; - IOVDD-supply = <®ulator>; - DRVDD-supply = <®ulator>; - DVDD-supply = <®ulator>; -}; - -SPI example: - -spi0: spi@f0000000 { - tlv320aic3x: codec@0 { - compatible = "ti,tlv320aic3x"; - reg = <0>; /* CS number */ - #sound-dai-cells = <0>; - spi-max-frequency = <1000000>; - - AVDD-supply = <®ulator>; - IOVDD-supply = <®ulator>; - DRVDD-supply = <®ulator>; - DVDD-supply = <®ulator>; - ai3x-ocmv = <0>; - }; -}; diff --git a/Documentation/devicetree/bindings/vendor-prefixes.yaml b/Documentation/devicetree/bindings/vendor-prefixes.yaml index 5c3c6abb983b..85bb190cdd2f 100644 --- a/Documentation/devicetree/bindings/vendor-prefixes.yaml +++ b/Documentation/devicetree/bindings/vendor-prefixes.yaml @@ -645,6 +645,8 @@ patternProperties: description: Inverse Path "^iom,.*": description: Iomega Corporation + "^irondevice,.*": + description: Iron Device Corporation "^isee,.*": description: ISEE 2007 S.L. "^isil,.*": diff --git a/Documentation/sound/alsa-configuration.rst b/Documentation/sound/alsa-configuration.rst index 21ab5e6f7062..5f31fa5e2435 100644 --- a/Documentation/sound/alsa-configuration.rst +++ b/Documentation/sound/alsa-configuration.rst @@ -70,7 +70,7 @@ dsp_map PCM device number maps assigned to the 1st OSS device; Default: 0 adsp_map - PCM device number maps assigned to the 2st OSS device; + PCM device number maps assigned to the 2nd OSS device; Default: 1 nonblock_open Don't block opening busy PCM devices; @@ -97,7 +97,7 @@ midi_map MIDI device number maps assigned to the 1st OSS device; Default: 0 amidi_map - MIDI device number maps assigned to the 2st OSS device; + MIDI device number maps assigned to the 2nd OSS device; Default: 1 Module snd-soc-core @@ -727,9 +727,9 @@ Module for EMU10K1/EMU10k2 based PCI sound cards. * Sound Blaster Audigy extin - bitmap of available external inputs for FX8010 (see bellow) + bitmap of available external inputs for FX8010 (see below) extout - bitmap of available external outputs for FX8010 (see bellow) + bitmap of available external outputs for FX8010 (see below) seq_ports allocated sequencer ports (4 by default) max_synth_voices diff --git a/Documentation/sound/cards/audigy-mixer.rst b/Documentation/sound/cards/audigy-mixer.rst index f3f4640ee2af..c506f8d16f2e 100644 --- a/Documentation/sound/cards/audigy-mixer.rst +++ b/Documentation/sound/cards/audigy-mixer.rst @@ -17,7 +17,7 @@ Digital mixer controls ====================== These controls are built using the DSP instructions. They offer extended -functionality. Only the default build-in code in the ALSA driver is described +functionality. Only the default built-in code in the ALSA driver is described here. Note that the controls work as attenuators: the maximum value is the neutral position leaving the signal unchanged. Note that if the same destination is mentioned in multiple controls, the signal is accumulated and can be wrapped diff --git a/Documentation/sound/cards/maya44.rst b/Documentation/sound/cards/maya44.rst index bf09a584b443..ab973f66c973 100644 --- a/Documentation/sound/cards/maya44.rst +++ b/Documentation/sound/cards/maya44.rst @@ -156,7 +156,7 @@ IEC958 Output S/PDIF should output the same signal as channel 3+4. [untested!] -Digitial output selectors +Digital output selectors These switches allow a direct digital routing from the ADCs to the DACs. Each switch determines where the digital input data to one of the DACs comes from. They are not supported by the ESI windows driver. diff --git a/Documentation/sound/cards/sb-live-mixer.rst b/Documentation/sound/cards/sb-live-mixer.rst index 2ce41d3822d8..357fcd619d39 100644 --- a/Documentation/sound/cards/sb-live-mixer.rst +++ b/Documentation/sound/cards/sb-live-mixer.rst @@ -31,7 +31,7 @@ Digital mixer controls ====================== These controls are built using the DSP instructions. They offer extended -functionality. Only the default build-in code in the ALSA driver is described +functionality. Only the default built-in code in the ALSA driver is described here. Note that the controls work as attenuators: the maximum value is the neutral position leaving the signal unchanged. Note that if the same destination is mentioned in multiple controls, the signal is accumulated and can be wrapped diff --git a/Documentation/sound/designs/jack-controls.rst b/Documentation/sound/designs/jack-controls.rst index ae25b1531bb0..e8a18f126a63 100644 --- a/Documentation/sound/designs/jack-controls.rst +++ b/Documentation/sound/designs/jack-controls.rst @@ -8,7 +8,7 @@ Why we need Jack kcontrols ALSA uses kcontrols to export audio controls(switch, volume, Mux, ...) to user space. This means userspace applications like pulseaudio can switch off headphones and switch on speakers when no headphones are -pluged in. +plugged in. The old ALSA jack code only created input devices for each registered jack. These jack input devices are not readable by userspace devices diff --git a/Documentation/sound/designs/seq-oss.rst b/Documentation/sound/designs/seq-oss.rst index e82ffe0e7f43..ec6304a07441 100644 --- a/Documentation/sound/designs/seq-oss.rst +++ b/Documentation/sound/designs/seq-oss.rst @@ -96,7 +96,7 @@ if you use an AWE64 card, you'll see like the following: Number of synth devices: 1 synth 0: [EMU8000] type 0x1 : subtype 0x20 : voices 32 - capabilties : ioctl enabled / load_patch enabled + capabilities : ioctl enabled / load_patch enabled Number of MIDI devices: 3 midi 0: [Emu8000 Port-0] ALSA port 65:0 diff --git a/Documentation/sound/hd-audio/notes.rst b/Documentation/sound/hd-audio/notes.rst index d118b6fe269b..3432feb0fa33 100644 --- a/Documentation/sound/hd-audio/notes.rst +++ b/Documentation/sound/hd-audio/notes.rst @@ -500,7 +500,7 @@ add_jack_modes (bool) change the headphone amp and mic bias VREF capabilities power_save_node (bool) advanced power management for each widget, controlling the power - sate (D0/D3) of each widget node depending on the actual pin and + state (D0/D3) of each widget node depending on the actual pin and stream states power_down_unused (bool) power down the unused widgets, a subset of power_save_node, and diff --git a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst index 07a620c5ca74..5c9523b7d55c 100644 --- a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst +++ b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst @@ -1720,16 +1720,16 @@ Typically, you'll have a hardware descriptor as below: - ``rate_min`` and ``rate_max`` define the minimum and maximum sample rate. This should correspond somehow to ``rates`` bits. -- ``channel_min`` and ``channel_max`` define, as you might already +- ``channels_min`` and ``channels_max`` define, as you might already expected, the minimum and maximum number of channels. - ``buffer_bytes_max`` defines the maximum buffer size in bytes. There is no ``buffer_bytes_min`` field, since it can be calculated from the minimum period size and the minimum number of - periods. Meanwhile, ``period_bytes_min`` and define the minimum and - maximum size of the period in bytes. ``periods_max`` and - ``periods_min`` define the maximum and minimum number of periods in - the buffer. + periods. Meanwhile, ``period_bytes_min`` and ``period_bytes_max`` + define the minimum and maximum size of the period in bytes. + ``periods_max`` and ``periods_min`` define the maximum and minimum + number of periods in the buffer. The “period” is a term that corresponds to a fragment in the OSS world. The period defines the size at which a PCM interrupt is |