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Merge tag 'sound-5.8-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai: "It was another busy development cycle, and the majority of changes are found in ASoC side. Below are Some highlights. ASoC core: - Lots of core cleanups and refactorings, still on-going work by Morimoto-san ASoC drivers: - Continued work on cleaning up and improving the Intel SOF stuff, along with new platform support including SoundWire - Fixes to make the Marvell SSPA driver work upstream - Support for AMD Renoir ACP, Dialog DA7212, Freescale EASRC and i.MX8M, Intel Elkhard Lake, Maxim MAX98390, Nuvoton NAU8812 and NAU8814 and Realtek RT1016. USB-audio: - Improvement for sync and implicit feedback streams with the more accurate frame size calculation and full-duplex support - Support for RME Babyface Pro and Prioneer DJ DJM HD-audio: - Fixes for Mic mute LED on HP machines - Re-enable support of Intel SST driver for SKL/KBL platforms FireWire: - Lots of refactoring, add support for RME FireFace and MOTU UltraLite-mk3" * tag 'sound-5.8-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (428 commits) ALSA: es1688: Add the missed snd_card_free() ALSA: hda: add sienna_cichlid audio asic id for sienna_cichlid up ALSA: usb-audio: Add Pioneer DJ DJM-900NXS2 support ASoC: qcom: q6asm-dai: kCFI fix ASoC: soc-card: add snd_soc_card_remove_dai_link() ASoC: soc-card: add snd_soc_card_add_dai_link() ASoC: soc-card: add snd_soc_card_set_bias_level_post() ASoC: soc-card: add snd_soc_card_set_bias_level() ASoC: soc-card: add snd_soc_card_remove() ASoC: soc-card: add snd_soc_card_late_probe() ASoC: soc-card: add snd_soc_card_probe() ASoC: soc-card: add probed bit field to snd_soc_card ASoC: soc-card: add snd_soc_card_resume_post() ASoC: soc-card: add snd_soc_card_resume_pre() ASoC: soc-card: add snd_soc_card_suspend_post() ASoC: soc-card: add snd_soc_card_suspend_pre() ASoC: soc-card: move snd_soc_card_subclass to soc-card ASoC: soc-card: move snd_soc_card_get_codec_dai() to soc-card ASoC: soc-card: move snd_soc_card_set/get_drvdata() to soc-card ASoC: soc-card: move snd_soc_card_jack_new() to soc-card ...
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Lines changed: 17128 additions & 4984 deletions

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‎Documentation/devicetree/bindings/dsp/fsl,dsp.yaml‎

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@@ -17,6 +17,8 @@ properties:
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compatible:
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enum:
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- fsl,imx8qxp-dsp
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- fsl,imx8qm-dsp
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- fsl,imx8mp-dsp
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reg:
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description: Should contain register location and length

‎Documentation/devicetree/bindings/sound/da7213.txt‎

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Dialog Semiconductor DA7213 Audio Codec bindings
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Dialog Semiconductor DA7212/DA7213 Audio Codec bindings
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======
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Required properties:
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- compatible : Should be "dlg,da7213"
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- compatible : Should be "dlg,da7212" or "dlg,da7213"
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- reg: Specifies the I2C slave address
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Optional properties:
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- dlg,dmic-clkrate : DMIC clock frequency (Hz).
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[<1500000>, <3000000>]
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- VDDA-supply : Regulator phandle for Analogue power supply
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- VDDMIC-supply : Regulator phandle for Mic Bias
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- VDDIO-supply : Regulator phandle for I/O power supply
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======
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Example:

‎Documentation/devicetree/bindings/sound/fsl,asrc.txt‎

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will be in use as default. Otherwise, the big endian
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mode will be in use for all the device registers.
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- fsl,asrc-format : Defines a mutual sample format used by DPCM Back
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Ends, which can replace the fsl,asrc-width.
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The value is 2 (S16_LE), or 6 (S24_LE).
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Example:
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asrc: asrc@2034000 {

‎Documentation/devicetree/bindings/sound/fsl,easrc.yaml‎

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# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
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%YAML 1.2
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---
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$id: http://devicetree.org/schemas/sound/fsl,easrc.yaml#
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$schema: http://devicetree.org/meta-schemas/core.yaml#
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title: NXP Asynchronous Sample Rate Converter (ASRC) Controller
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maintainers:
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- Shengjiu Wang <shengjiu.wang@nxp.com>
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properties:
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$nodename:
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pattern: "^easrc@.*"
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compatible:
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const: fsl,imx8mn-easrc
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reg:
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maxItems: 1
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interrupts:
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maxItems: 1
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clocks:
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items:
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- description: Peripheral clock
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clock-names:
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items:
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- const: mem
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dmas:
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maxItems: 8
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dma-names:
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items:
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- const: ctx0_rx
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- const: ctx0_tx
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- const: ctx1_rx
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- const: ctx1_tx
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- const: ctx2_rx
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- const: ctx2_tx
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- const: ctx3_rx
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- const: ctx3_tx
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firmware-name:
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allOf:
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- $ref: /schemas/types.yaml#/definitions/string
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- const: imx/easrc/easrc-imx8mn.bin
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description: The coefficient table for the filters
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fsl,asrc-rate:
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allOf:
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- $ref: /schemas/types.yaml#/definitions/uint32
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- minimum: 8000
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- maximum: 192000
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description: Defines a mutual sample rate used by DPCM Back Ends
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fsl,asrc-format:
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allOf:
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- $ref: /schemas/types.yaml#/definitions/uint32
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- enum: [2, 6, 10, 32, 36]
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default: 2
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description:
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Defines a mutual sample format used by DPCM Back Ends
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required:
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- compatible
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- reg
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- interrupts
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- clocks
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- clock-names
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- dmas
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- dma-names
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- firmware-name
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- fsl,asrc-rate
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- fsl,asrc-format
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examples:
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- |
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#include <dt-bindings/clock/imx8mn-clock.h>
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easrc: easrc@300c0000 {
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compatible = "fsl,imx8mn-easrc";
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reg = <0x0 0x300c0000 0x0 0x10000>;
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interrupts = <0x0 122 0x4>;
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clocks = <&clk IMX8MN_CLK_ASRC_ROOT>;
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clock-names = "mem";
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dmas = <&sdma2 16 23 0> , <&sdma2 17 23 0>,
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<&sdma2 18 23 0> , <&sdma2 19 23 0>,
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<&sdma2 20 23 0> , <&sdma2 21 23 0>,
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<&sdma2 22 23 0> , <&sdma2 23 23 0>;
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dma-names = "ctx0_rx", "ctx0_tx",
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"ctx1_rx", "ctx1_tx",
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"ctx2_rx", "ctx2_tx",
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"ctx3_rx", "ctx3_tx";
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firmware-name = "imx/easrc/easrc-imx8mn.bin";
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fsl,asrc-rate = <8000>;
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fsl,asrc-format = <2>;
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};

‎Documentation/devicetree/bindings/sound/fsl,esai.txt‎

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"fsl,imx35-esai",
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"fsl,vf610-esai",
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"fsl,imx6ull-esai",
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"fsl,imx8qm-esai",
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- reg : Offset and length of the register set for the device.
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‎Documentation/devicetree/bindings/sound/marvell,mmp-sspa.yaml‎

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# SPDX-License-Identifier: (GPL-2.0+ OR BSD-2-Clause)
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%YAML 1.2
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---
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$id: http://devicetree.org/schemas/sound/marvell,mmp-sspa.yaml#
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$schema: http://devicetree.org/meta-schemas/core.yaml#
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title: Marvel SSPA Digital Audio Interface Bindings
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maintainers:
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- Lubomir Rintel <lkundrak@v3.sk>
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properties:
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$nodename:
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pattern: "^audio-controller(@.*)?$"
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compatible:
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const: marvell,mmp-sspa
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reg:
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items:
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- description: RX block
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- description: TX block
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interrupts:
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maxItems: 1
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clocks:
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items:
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- description: Clock for the Audio block
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- description: I2S bit clock
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clock-names:
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items:
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- const: audio
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- const: bitclk
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power-domains:
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maxItems: 1
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'#sound-dai-cells':
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const: 0
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dmas:
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items:
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- description: TX DMA Channel
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- description: RX DMA Channel
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dma-names:
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items:
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- const: tx
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- const: rx
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port:
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type: object
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properties:
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endpoint:
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type: object
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properties:
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remote-endpoint: true
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frame-master:
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type: boolean
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description: SoC generates the frame clock
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bitclock-master:
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type: boolean
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description: SoC generates the bit clock
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dai-format:
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$ref: /schemas/types.yaml#/definitions/string
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description: The digital audio format
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const: i2s
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required:
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- remote-endpoint
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required:
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- endpoint
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additionalProperties: false
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required:
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- "#sound-dai-cells"
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- compatible
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- reg
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- interrupts
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- clocks
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- clock-names
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- dmas
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- dma-names
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- port
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additionalProperties: false
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examples:
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- |
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#include <dt-bindings/clock/marvell,mmp2.h>
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audio-controller@d42a0c00 {
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compatible = "marvell,mmp-sspa";
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reg = <0xd42a0c00 0x30>,
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<0xd42a0c80 0x30>;
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interrupts = <2>;
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clock-names = "audio", "bitclk";
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clocks = <&soc_clocks 127>,
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<&audio_clk 1>;
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#sound-dai-cells = <0>;
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dmas = <&adma0 0>, <&adma0 1>;
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dma-names = "tx", "rx";
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port {
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endpoint {
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remote-endpoint = <&rt5631_0>;
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frame-master;
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bitclock-master;
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dai-format = "i2s";
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};
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};
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};
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...

‎Documentation/devicetree/bindings/sound/nau8810.txt‎

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NAU8810 audio CODEC
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NAU8810/NAU8812/NAU8814 audio CODEC
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This device supports I2C only.
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Required properties:
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- compatible : "nuvoton,nau8810"
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- compatible : One of "nuvoton,nau8810" or "nuvoton,nau8812" or
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"nuvoton,nau8814"
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- reg : the I2C address of the device.
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‎Documentation/devicetree/bindings/sound/nau8825.txt‎

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nuvoton,crosstalk-enable;
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clock-names = "mclk";
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clocks = <&tegra_car TEGRA210_CLK_CLK_OUT_2>;
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clocks = <&tegra_pmc TEGRA_PMC_CLK_OUT_2>;
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};

‎Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt‎

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- nvidia,hp-det-gpios : The GPIO that detect headphones are plugged in
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- nvidia,int-mic-en-gpios : The GPIO that enables the internal microphone
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- nvidia,ext-mic-en-gpios : The GPIO that enables the external microphone
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- nvidia,headset : The Mic Jack represents state of the headset microphone pin
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Example:
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‎Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt‎

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- reg : Must contain an address for each entry in reg-names.
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- reg-names : A list which must include the following entries:
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* "lpass-lpaif"
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- #address-cells : Must be 1
34+
- #size-cells : Must be 0
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Optional properties:
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3840
- qcom,adsp : Phandle for the audio DSP node
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By default, the driver uses up to 4 MI2S SD lines, for a total of 8 channels.
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The SD lines to use can be configured by adding subnodes for each of the DAIs.
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Required properties for each DAI (represented by a subnode):
46+
- reg : Must be one of the DAI IDs
47+
(usually part of dt-bindings header)
48+
- qcom,playback-sd-lines: List of serial data lines to use for playback
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Each SD line should be represented by a number from 0-3.
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- qcom,capture-sd-lines : List of serial data lines to use for capture
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Each SD line should be represented by a number from 0-3.
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Note that adding a subnode changes the default to "no lines configured",
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so both playback and capture lines should be configured when a subnode is added.
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Example:
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lpass@28100000 {
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5167
reg = <0x28100000 0x10000>;
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reg-names = "lpass-lpaif";
5369
qcom,adsp = <&adsp>;
70+
71+
#address-cells = <1>;
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#size-cells = <0>;
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/* Optional to set different MI2S SD lines */
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dai@3 {
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reg = <MI2S_QUATERNARY>;
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qcom,playback-sd-lines = <0 1>;
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};
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};

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