sound updates for 5.9
This became wide and scattered updates all over the sound tree as
diffstat shows: lots of (still ongoing) refactoring works in ASoC,
fixes and cleanups caught by static analysis, inclusive term
conversions as well as lots of new drivers. Below are highlights:
ASoC core:
* API cleanups and conversions to the unified mute_stream() call
* Simplify I/O helper functions
* Use helper macros to retrieve RTD from substreams
ASoC drivers:
* Lots of fixes and cleanups in Intel ASoC drivers
* Lots of new stuff: Freescale MQS and i.MX6sx, Intel KeemBay I2S,
Maxim MAX98360A and MAX98373 SoundWire, various Mediatek boards,
nVidia Tegra 186 and 210, RealTek RL6231, Samsung Midas and Aries
boards, TI J721e EVM
ALSA core:
* Minor code refacotring for SG-buffer handling
HD-audio:
* Generalization of mute-LED handling with LED classdev
* Intel silent stream support for HDMI
* Device-specific fixes: CA0132, Loongson-3
Others:
* Usual USB- and HD-audio quirks for various devices
* Fixes for echoaudio DMA position handling
* Various documents and trivial fixes for sparse warnings
* Conversion to adapt inclusive terms
-----BEGIN PGP SIGNATURE-----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=sy7n
-----END PGP SIGNATURE-----
Merge tag 'sound-5.9-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"This became wide and scattered updates all over the sound tree as
diffstat shows: lots of (still ongoing) refactoring works in ASoC,
fixes and cleanups caught by static analysis, inclusive term
conversions as well as lots of new drivers. Below are highlights:
ASoC core:
- API cleanups and conversions to the unified mute_stream() call
- Simplify I/O helper functions
- Use helper macros to retrieve RTD from substreams
ASoC drivers:
- Lots of fixes and cleanups in Intel ASoC drivers
- Lots of new stuff: Freescale MQS and i.MX6sx, Intel KeemBay I2S,
Maxim MAX98360A and MAX98373 SoundWire, various Mediatek boards,
nVidia Tegra 186 and 210, RealTek RL6231, Samsung Midas and Aries
boards, TI J721e EVM
ALSA core:
- Minor code refacotring for SG-buffer handling
HD-audio:
- Generalization of mute-LED handling with LED classdev
- Intel silent stream support for HDMI
- Device-specific fixes: CA0132, Loongson-3
Others:
- Usual USB- and HD-audio quirks for various devices
- Fixes for echoaudio DMA position handling
- Various documents and trivial fixes for sparse warnings
- Conversion to adopt inclusive terms"
* tag 'sound-5.9-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (479 commits)
ALSA: pci: delete repeated words in comments
ALSA: isa: delete repeated words in comments
ALSA: hda/tegra: Add 100us dma stop delay
ALSA: hda: Add dma stop delay variable
ASoC: hda/tegra: Set buffer alignment to 128 bytes
ALSA: seq: oss: Serialize ioctls
ALSA: hda/hdmi: Add quirk to force connectivity
ALSA: usb-audio: add startech usb audio dock name
ALSA: usb-audio: Add support for Lenovo ThinkStation P620
Revert "ALSA: hda: call runtime_allow() for all hda controllers"
ALSA: hda/ca0132 - Fix AE-5 microphone selection commands.
ALSA: hda/ca0132 - Add new quirk ID for Recon3D.
ALSA: hda/ca0132 - Fix ZxR Headphone gain control get value.
ALSA: hda/realtek: Add alc269/alc662 pin-tables for Loongson-3 laptops
ALSA: docs: fix typo
ALSA: doc: use correct config variable name
ASoC: core: Two step component registration
ASoC: core: Simplify snd_soc_component_initialize declaration
ASoC: core: Relocate and expose snd_soc_component_initialize
ASoC: sh: Replace 'select' DMADEVICES 'with depends on'
...
This commit is contained in:
Linus Torvalds
commit
3f9df56480
|
|
@ -1,9 +1,9 @@
|
|||
Analog Devices ADAU1977/ADAU1978/ADAU1979
|
||||
|
||||
Datasheets:
|
||||
http://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1977.pdf
|
||||
http://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1978.pdf
|
||||
http://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1979.pdf
|
||||
https://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1977.pdf
|
||||
https://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1978.pdf
|
||||
https://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1979.pdf
|
||||
|
||||
This driver supports both the I2C and SPI bus.
|
||||
|
||||
|
|
|
|||
|
|
@ -1,27 +0,0 @@
|
|||
AK4613 I2C transmitter
|
||||
|
||||
This device supports I2C mode only.
|
||||
|
||||
Required properties:
|
||||
|
||||
- compatible : "asahi-kasei,ak4613"
|
||||
- reg : The chip select number on the I2C bus
|
||||
|
||||
Optional properties:
|
||||
- asahi-kasei,in1-single-end : Boolean. Indicate input / output pins are single-ended.
|
||||
- asahi-kasei,in2-single-end rather than differential.
|
||||
- asahi-kasei,out1-single-end
|
||||
- asahi-kasei,out2-single-end
|
||||
- asahi-kasei,out3-single-end
|
||||
- asahi-kasei,out4-single-end
|
||||
- asahi-kasei,out5-single-end
|
||||
- asahi-kasei,out6-single-end
|
||||
|
||||
Example:
|
||||
|
||||
&i2c {
|
||||
ak4613: ak4613@10 {
|
||||
compatible = "asahi-kasei,ak4613";
|
||||
reg = <0x10>;
|
||||
};
|
||||
};
|
||||
|
|
@ -0,0 +1,49 @@
|
|||
# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
|
||||
%YAML 1.2
|
||||
---
|
||||
$id: http://devicetree.org/schemas/sound/ak4613.yaml#
|
||||
$schema: http://devicetree.org/meta-schemas/core.yaml#
|
||||
|
||||
title: AK4613 I2C transmitter Device Tree Bindings
|
||||
|
||||
maintainers:
|
||||
- Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
|
||||
|
||||
properties:
|
||||
compatible:
|
||||
const: asahi-kasei,ak4613
|
||||
|
||||
reg:
|
||||
maxItems: 1
|
||||
|
||||
clocks:
|
||||
maxItems: 1
|
||||
|
||||
"#sound-dai-cells":
|
||||
const: 0
|
||||
|
||||
patternProperties:
|
||||
"^asahi-kasei,in[1-2]-single-end$":
|
||||
description: Input Pin 1 - 2.
|
||||
$ref: /schemas/types.yaml#/definitions/flag
|
||||
|
||||
"^asahi-kasei,out[1-6]-single-end$":
|
||||
description: Output Pin 1 - 6.
|
||||
$ref: /schemas/types.yaml#/definitions/flag
|
||||
|
||||
required:
|
||||
- compatible
|
||||
- reg
|
||||
|
||||
additionalProperties: false
|
||||
|
||||
examples:
|
||||
- |
|
||||
i2c {
|
||||
#address-cells = <1>;
|
||||
#size-cells = <0>;
|
||||
ak4613: codec@10 {
|
||||
compatible = "asahi-kasei,ak4613";
|
||||
reg = <0x10>;
|
||||
};
|
||||
};
|
||||
|
|
@ -1,37 +0,0 @@
|
|||
AK4642 I2C transmitter
|
||||
|
||||
This device supports I2C mode only.
|
||||
|
||||
Required properties:
|
||||
|
||||
- compatible : "asahi-kasei,ak4642" or "asahi-kasei,ak4643" or "asahi-kasei,ak4648"
|
||||
- reg : The chip select number on the I2C bus
|
||||
|
||||
Optional properties:
|
||||
|
||||
- #clock-cells : common clock binding; shall be set to 0
|
||||
- clocks : common clock binding; MCKI clock
|
||||
- clock-frequency : common clock binding; frequency of MCKO
|
||||
- clock-output-names : common clock binding; MCKO clock name
|
||||
|
||||
Example 1:
|
||||
|
||||
&i2c {
|
||||
ak4648: ak4648@12 {
|
||||
compatible = "asahi-kasei,ak4642";
|
||||
reg = <0x12>;
|
||||
};
|
||||
};
|
||||
|
||||
Example 2:
|
||||
|
||||
&i2c {
|
||||
ak4643: codec@12 {
|
||||
compatible = "asahi-kasei,ak4643";
|
||||
reg = <0x12>;
|
||||
#clock-cells = <0>;
|
||||
clocks = <&audio_clock>;
|
||||
clock-frequency = <12288000>;
|
||||
clock-output-names = "ak4643_mcko";
|
||||
};
|
||||
};
|
||||
|
|
@ -0,0 +1,58 @@
|
|||
# SPDX-License-Identifier: GPL-2.0
|
||||
%YAML 1.2
|
||||
---
|
||||
$id: http://devicetree.org/schemas/sound/ak4642.yaml#
|
||||
$schema: http://devicetree.org/meta-schemas/core.yaml#
|
||||
|
||||
title: AK4642 I2C transmitter Device Tree Bindings
|
||||
|
||||
maintainers:
|
||||
- Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
|
||||
|
||||
properties:
|
||||
compatible:
|
||||
enum:
|
||||
- asahi-kasei,ak4642
|
||||
- asahi-kasei,ak4643
|
||||
- asahi-kasei,ak4648
|
||||
|
||||
reg:
|
||||
maxItems: 1
|
||||
|
||||
"#clock-cells":
|
||||
const: 0
|
||||
"#sound-dai-cells":
|
||||
const: 0
|
||||
|
||||
clocks:
|
||||
maxItems: 1
|
||||
|
||||
clock-frequency:
|
||||
description: common clock binding; frequency of MCKO
|
||||
$ref: /schemas/types.yaml#/definitions/uint32
|
||||
|
||||
clock-output-names:
|
||||
description: common clock name
|
||||
$ref: /schemas/types.yaml#/definitions/string
|
||||
|
||||
required:
|
||||
- compatible
|
||||
- reg
|
||||
|
||||
additionalProperties: false
|
||||
|
||||
examples:
|
||||
- |
|
||||
i2c {
|
||||
#address-cells = <1>;
|
||||
#size-cells = <0>;
|
||||
ak4643: codec@12 {
|
||||
compatible = "asahi-kasei,ak4643";
|
||||
#sound-dai-cells = <0>;
|
||||
reg = <0x12>;
|
||||
#clock-cells = <0>;
|
||||
clocks = <&audio_clock>;
|
||||
clock-frequency = <12288000>;
|
||||
clock-output-names = "ak4643_mcko";
|
||||
};
|
||||
};
|
||||
|
|
@ -1,23 +0,0 @@
|
|||
Everest ES8316 audio CODEC
|
||||
|
||||
This device supports both I2C and SPI.
|
||||
|
||||
Required properties:
|
||||
|
||||
- compatible : should be "everest,es8316"
|
||||
- reg : the I2C address of the device for I2C
|
||||
|
||||
Optional properties:
|
||||
|
||||
- clocks : a list of phandle, should contain entries for clock-names
|
||||
- clock-names : should include as follows:
|
||||
"mclk" : master clock (MCLK) of the device
|
||||
|
||||
Example:
|
||||
|
||||
es8316: codec@11 {
|
||||
compatible = "everest,es8316";
|
||||
reg = <0x11>;
|
||||
clocks = <&clks 10>;
|
||||
clock-names = "mclk";
|
||||
};
|
||||
|
|
@ -0,0 +1,50 @@
|
|||
# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
|
||||
%YAML 1.2
|
||||
---
|
||||
$id: http://devicetree.org/schemas/sound/everest,es8316.yaml#
|
||||
$schema: http://devicetree.org/meta-schemas/core.yaml#
|
||||
|
||||
title: Everest ES8316 audio CODEC
|
||||
|
||||
maintainers:
|
||||
- Daniel Drake <drake@endlessm.com>
|
||||
- Katsuhiro Suzuki <katsuhiro@katsuster.net>
|
||||
|
||||
properties:
|
||||
compatible:
|
||||
const: everest,es8316
|
||||
|
||||
reg:
|
||||
maxItems: 1
|
||||
|
||||
clocks:
|
||||
items:
|
||||
- description: clock for master clock (MCLK)
|
||||
|
||||
clock-names:
|
||||
items:
|
||||
- const: mclk
|
||||
|
||||
"#sound-dai-cells":
|
||||
const: 0
|
||||
|
||||
required:
|
||||
- compatible
|
||||
- reg
|
||||
- "#sound-dai-cells"
|
||||
|
||||
additionalProperties: false
|
||||
|
||||
examples:
|
||||
- |
|
||||
i2c0 {
|
||||
#address-cells = <1>;
|
||||
#size-cells = <0>;
|
||||
es8316: codec@11 {
|
||||
compatible = "everest,es8316";
|
||||
reg = <0x11>;
|
||||
clocks = <&clks 10>;
|
||||
clock-names = "mclk";
|
||||
#sound-dai-cells = <0>;
|
||||
};
|
||||
};
|
||||
|
|
@ -6,7 +6,11 @@ a fibre cable.
|
|||
|
||||
Required properties:
|
||||
|
||||
- compatible : Compatible list, must contain "fsl,imx35-spdif".
|
||||
- compatible : Compatible list, should contain one of the following
|
||||
compatibles:
|
||||
"fsl,imx35-spdif",
|
||||
"fsl,vf610-spdif",
|
||||
"fsl,imx6sx-spdif",
|
||||
|
||||
- reg : Offset and length of the register set for the device.
|
||||
|
||||
|
|
|
|||
|
|
@ -34,6 +34,10 @@ The compatible list for this generic sound card currently:
|
|||
|
||||
"fsl,imx-audio-wm8960"
|
||||
|
||||
"fsl,imx-audio-mqs"
|
||||
|
||||
"fsl,imx-audio-wm8524"
|
||||
|
||||
Required properties:
|
||||
|
||||
- compatible : Contains one of entries in the compatible list.
|
||||
|
|
@ -44,6 +48,11 @@ Required properties:
|
|||
|
||||
- audio-codec : The phandle of an audio codec
|
||||
|
||||
Optional properties:
|
||||
|
||||
- audio-asrc : The phandle of ASRC. It can be absent if there's no
|
||||
need to add ASRC support via DPCM.
|
||||
|
||||
- audio-routing : A list of the connections between audio components.
|
||||
Each entry is a pair of strings, the first being the
|
||||
connection's sink, the second being the connection's
|
||||
|
|
@ -60,10 +69,13 @@ Required properties:
|
|||
coexisting in order to support the old bindings
|
||||
of wm8962 and sgtl5000.
|
||||
|
||||
Optional properties:
|
||||
|
||||
- audio-asrc : The phandle of ASRC. It can be absent if there's no
|
||||
need to add ASRC support via DPCM.
|
||||
- hp-det-gpio : The GPIO that detect headphones are plugged in
|
||||
- mic-det-gpio : The GPIO that detect microphones are plugged in
|
||||
- bitclock-master : Indicates dai-link bit clock master; for details see simple-card.yaml.
|
||||
- frame-master : Indicates dai-link frame master; for details see simple-card.yaml.
|
||||
- dai-format : audio format, for details see simple-card.yaml.
|
||||
- frame-inversion : dai-link uses frame clock inversion, for details see simple-card.yaml.
|
||||
- bitclock-inversion : dai-link uses bit clock inversion, for details see simple-card.yaml.
|
||||
|
||||
Optional unless SSI is selected as a CPU DAI:
|
||||
|
||||
|
|
|
|||
|
|
@ -0,0 +1,70 @@
|
|||
# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause)
|
||||
# Copyright 2020 Intel Corporation
|
||||
%YAML 1.2
|
||||
---
|
||||
$id: http://devicetree.org/schemas/sound/intel,keembay-i2s.yaml#
|
||||
$schema: http://devicetree.org/meta-schemas/core.yaml#
|
||||
|
||||
title: Intel KeemBay I2S Device Tree Bindings
|
||||
|
||||
maintainers:
|
||||
- Sia, Jee Heng <jee.heng.sia@intel.com>
|
||||
|
||||
description: |
|
||||
Intel KeemBay I2S
|
||||
|
||||
properties:
|
||||
compatible:
|
||||
enum:
|
||||
- intel,keembay-i2s
|
||||
|
||||
"#sound-dai-cells":
|
||||
const: 0
|
||||
|
||||
reg:
|
||||
items:
|
||||
- description: I2S registers
|
||||
- description: I2S gen configuration
|
||||
|
||||
reg-names:
|
||||
items:
|
||||
- const: i2s-regs
|
||||
- const: i2s_gen_cfg
|
||||
|
||||
interrupts:
|
||||
maxItems: 1
|
||||
|
||||
clocks:
|
||||
items:
|
||||
- description: Bus Clock
|
||||
- description: Module Clock
|
||||
|
||||
clock-names:
|
||||
items:
|
||||
- const: osc
|
||||
- const: apb_clk
|
||||
|
||||
required:
|
||||
- compatible
|
||||
- "#sound-dai-cells"
|
||||
- reg
|
||||
- clocks
|
||||
- clock-names
|
||||
- interrupts
|
||||
|
||||
examples:
|
||||
- |
|
||||
#include <dt-bindings/interrupt-controller/arm-gic.h>
|
||||
#include <dt-bindings/interrupt-controller/irq.h>
|
||||
#define KEEM_BAY_PSS_AUX_I2S3
|
||||
#define KEEM_BAY_PSS_I2S3
|
||||
i2s3: i2s@20140000 {
|
||||
compatible = "intel,keembay-i2s";
|
||||
#sound-dai-cells = <0>;
|
||||
reg = <0x20140000 0x200>, /* I2S registers */
|
||||
<0x202a00a4 0x4>; /* I2S gen configuration */
|
||||
reg-names = "i2s-regs", "i2s_gen_cfg";
|
||||
interrupts = <GIC_SPI 120 IRQ_TYPE_LEVEL_HIGH>;
|
||||
clock-names = "osc", "apb_clk";
|
||||
clocks = <&scmi_clk KEEM_BAY_PSS_AUX_I2S3>, <&scmi_clk KEEM_BAY_PSS_I2S3>;
|
||||
};
|
||||
|
|
@ -1,9 +1,10 @@
|
|||
Maxim MAX98357A audio DAC
|
||||
Maxim MAX98357A/MAX98360A audio DAC
|
||||
|
||||
This node models the Maxim MAX98357A DAC.
|
||||
This node models the Maxim MAX98357A/MAX98360A DAC.
|
||||
|
||||
Required properties:
|
||||
- compatible : "maxim,max98357a"
|
||||
- compatible : "maxim,max98357a" for MAX98357A.
|
||||
"maxim,max98360a" for MAX98360A.
|
||||
|
||||
Optional properties:
|
||||
- sdmode-gpios : GPIO specifier for the chip's SD_MODE pin.
|
||||
|
|
@ -20,3 +21,8 @@ max98357a {
|
|||
compatible = "maxim,max98357a";
|
||||
sdmode-gpios = <&qcom_pinmux 25 0>;
|
||||
};
|
||||
|
||||
max98360a {
|
||||
compatible = "maxim,max98360a";
|
||||
sdmode-gpios = <&qcom_pinmux 25 0>;
|
||||
};
|
||||
|
|
|
|||
|
|
@ -0,0 +1,51 @@
|
|||
# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
|
||||
%YAML 1.2
|
||||
---
|
||||
$id: http://devicetree.org/schemas/sound/maxim,max98390.yaml#
|
||||
$schema: http://devicetree.org/meta-schemas/core.yaml#
|
||||
|
||||
title: Maxim Integrated MAX98390 Speaker Amplifier with Integrated Dynamic Speaker Management
|
||||
|
||||
maintainers:
|
||||
- Steve Lee <steves.lee@maximintegrated.com>
|
||||
|
||||
properties:
|
||||
compatible:
|
||||
const: maxim,max98390
|
||||
|
||||
reg:
|
||||
maxItems: 1
|
||||
description: I2C address of the device.
|
||||
|
||||
maxim,temperature_calib:
|
||||
allOf:
|
||||
- $ref: /schemas/types.yaml#/definitions/uint32
|
||||
description: The calculated temperature data was measured while doing the calibration.
|
||||
minimum: 0
|
||||
maximum: 65535
|
||||
|
||||
maxim,r0_calib:
|
||||
allOf:
|
||||
- $ref: /schemas/types.yaml#/definitions/uint32
|
||||
description: This is r0 calibration data which was measured in factory mode.
|
||||
minimum: 1
|
||||
maximum: 8388607
|
||||
|
||||
required:
|
||||
- compatible
|
||||
- reg
|
||||
|
||||
additionalProperties: false
|
||||
|
||||
examples:
|
||||
- |
|
||||
i2c {
|
||||
#address-cells = <1>;
|
||||
#size-cells = <0>;
|
||||
max98390: amplifier@38 {
|
||||
compatible = "maxim,max98390";
|
||||
reg = <0x38>;
|
||||
maxim,temperature_calib = <1024>;
|
||||
maxim,r0_calib = <100232>;
|
||||
};
|
||||
};
|
||||
|
|
@ -10,9 +10,15 @@ Required properties:
|
|||
- compatible : "mediatek,mt6358-sound".
|
||||
- Avdd-supply : power source of AVDD
|
||||
|
||||
Optional properties:
|
||||
- mediatek,dmic-mode : Indicates how many data pins are used to transmit two
|
||||
channels of PDM signal. 0 means two wires, 1 means one wire. Default
|
||||
value is 0.
|
||||
|
||||
Example:
|
||||
|
||||
mt6358_snd {
|
||||
compatible = "mediatek,mt6358-sound";
|
||||
Avdd-supply = <&mt6358_vaud28_reg>;
|
||||
mediatek,dmic-mode = <0>;
|
||||
};
|
||||
|
|
|
|||
|
|
@ -1,15 +1,20 @@
|
|||
MT8183 with MT6358, DA7219 and MAX98357 CODECS
|
||||
MT8183 with MT6358, DA7219, MAX98357, and RT1015 CODECS
|
||||
|
||||
Required properties:
|
||||
- compatible : "mediatek,mt8183_da7219_max98357"
|
||||
- compatible : "mediatek,mt8183_da7219_max98357" for MAX98357A codec
|
||||
"mediatek,mt8183_da7219_rt1015" for RT1015 codec
|
||||
- mediatek,headset-codec: the phandles of da7219 codecs
|
||||
- mediatek,platform: the phandle of MT8183 ASoC platform
|
||||
|
||||
Optional properties:
|
||||
- mediatek,hdmi-codec: the phandles of HDMI codec
|
||||
|
||||
Example:
|
||||
|
||||
sound {
|
||||
compatible = "mediatek,mt8183_da7219_max98357";
|
||||
mediatek,headset-codec = <&da7219>;
|
||||
mediatek,hdmi-codec = <&it6505dptx>;
|
||||
mediatek,platform = <&afe>;
|
||||
};
|
||||
|
||||
|
|
|
|||
|
|
@ -1,13 +1,16 @@
|
|||
MT8183 with MT6358, TS3A227 and MAX98357 CODECS
|
||||
MT8183 with MT6358, TS3A227, MAX98357, and RT1015 CODECS
|
||||
|
||||
Required properties:
|
||||
- compatible : "mediatek,mt8183_mt6358_ts3a227_max98357"
|
||||
- compatible : "mediatek,mt8183_mt6358_ts3a227_max98357" for MAX98357A codec
|
||||
"mediatek,mt8183_mt6358_ts3a227_max98357b" for MAX98357B codec
|
||||
"mediatek,mt8183_mt6358_ts3a227_rt1015" for RT1015 codec
|
||||
- mediatek,platform: the phandle of MT8183 ASoC platform
|
||||
|
||||
Optional properties:
|
||||
- mediatek,headset-codec: the phandles of ts3a227 codecs
|
||||
- mediatek,ec-codec: the phandle of EC codecs.
|
||||
See google,cros-ec-codec.txt for more details.
|
||||
- mediatek,hdmi-codec: the phandles of HDMI codec
|
||||
|
||||
Example:
|
||||
|
||||
|
|
@ -15,6 +18,7 @@ Example:
|
|||
compatible = "mediatek,mt8183_mt6358_ts3a227_max98357";
|
||||
mediatek,headset-codec = <&ts3a227>;
|
||||
mediatek,ec-codec = <&ec_codec>;
|
||||
mediatek,hdmi-codec = <&it6505dptx>;
|
||||
mediatek,platform = <&afe>;
|
||||
};
|
||||
|
||||
|
|
|
|||
|
|
@ -0,0 +1,83 @@
|
|||
# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
|
||||
%YAML 1.2
|
||||
---
|
||||
$id: http://devicetree.org/schemas/sound/nvidia,tegra186-dspk.yaml#
|
||||
$schema: http://devicetree.org/meta-schemas/core.yaml#
|
||||
|
||||
title: Tegra186 DSPK Controller Device Tree Bindings
|
||||
|
||||
description: |
|
||||
The Digital Speaker Controller (DSPK) can be viewed as a Pulse
|
||||
Density Modulation (PDM) transmitter that up-samples the input to
|
||||
the desired sampling rate by interpolation and then converts the
|
||||
over sampled Pulse Code Modulation (PCM) input to the desired 1-bit
|
||||
output via Delta Sigma Modulation (DSM).
|
||||
|
||||
maintainers:
|
||||
- Jon Hunter <jonathanh@nvidia.com>
|
||||
- Sameer Pujar <spujar@nvidia.com>
|
||||
|
||||
properties:
|
||||
$nodename:
|
||||
pattern: "^dspk@[0-9a-f]*$"
|
||||
|
||||
compatible:
|
||||
oneOf:
|
||||
- const: nvidia,tegra186-dspk
|
||||
- items:
|
||||
- const: nvidia,tegra194-dspk
|
||||
- const: nvidia,tegra186-dspk
|
||||
|
||||
reg:
|
||||
maxItems: 1
|
||||
|
||||
clocks:
|
||||
maxItems: 1
|
||||
|
||||
clock-names:
|
||||
const: dspk
|
||||
|
||||
assigned-clocks:
|
||||
maxItems: 1
|
||||
|
||||
assigned-clock-parents:
|
||||
maxItems: 1
|
||||
|
||||
assigned-clock-rates:
|
||||
maxItems: 1
|
||||
|
||||
sound-name-prefix:
|
||||
pattern: "^DSPK[1-9]$"
|
||||
allOf:
|
||||
- $ref: /schemas/types.yaml#/definitions/string
|
||||
description:
|
||||
Used as prefix for sink/source names of the component. Must be a
|
||||
unique string among multiple instances of the same component.
|
||||
The name can be "DSPK1" or "DSPKx", where x depends on the maximum
|
||||
available instances on a Tegra SoC.
|
||||
|
||||
required:
|
||||
- compatible
|
||||
- reg
|
||||
- clocks
|
||||
- clock-names
|
||||
- assigned-clocks
|
||||
- assigned-clock-parents
|
||||
- sound-name-prefix
|
||||
|
||||
examples:
|
||||
- |
|
||||
#include<dt-bindings/clock/tegra186-clock.h>
|
||||
|
||||
dspk@2905000 {
|
||||
compatible = "nvidia,tegra186-dspk";
|
||||
reg = <0x2905000 0x100>;
|
||||
clocks = <&bpmp TEGRA186_CLK_DSPK1>;
|
||||
clock-names = "dspk";
|
||||
assigned-clocks = <&bpmp TEGRA186_CLK_DSPK1>;
|
||||
assigned-clock-parents = <&bpmp TEGRA186_CLK_PLL_A_OUT0>;
|
||||
assigned-clock-rates = <12288000>;
|
||||
sound-name-prefix = "DSPK1";
|
||||
};
|
||||
|
||||
...
|
||||
|
|
@ -0,0 +1,111 @@
|
|||
# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
|
||||
%YAML 1.2
|
||||
---
|
||||
$id: http://devicetree.org/schemas/sound/nvidia,tegra210-admaif.yaml#
|
||||
$schema: http://devicetree.org/meta-schemas/core.yaml#
|
||||
|
||||
title: Tegra210 ADMAIF Device Tree Bindings
|
||||
|
||||
description: |
|
||||
ADMAIF is the interface between ADMA and AHUB. Each ADMA channel
|
||||
that sends/receives data to/from AHUB must interface through an
|
||||
ADMAIF channel. ADMA channel sending data to AHUB pairs with ADMAIF
|
||||
Tx channel and ADMA channel receiving data from AHUB pairs with
|
||||
ADMAIF Rx channel.
|
||||
|
||||
maintainers:
|
||||
- Jon Hunter <jonathanh@nvidia.com>
|
||||
- Sameer Pujar <spujar@nvidia.com>
|
||||
|
||||
properties:
|
||||
$nodename:
|
||||
pattern: "^admaif@[0-9a-f]*$"
|
||||
|
||||
compatible:
|
||||
oneOf:
|
||||
- enum:
|
||||
- nvidia,tegra210-admaif
|
||||
- nvidia,tegra186-admaif
|
||||
- items:
|
||||
- const: nvidia,tegra194-admaif
|
||||
- const: nvidia,tegra186-admaif
|
||||
|
||||
reg:
|
||||
maxItems: 1
|
||||
|
||||
dmas: true
|
||||
|
||||
dma-names: true
|
||||
|
||||
if:
|
||||
properties:
|
||||
compatible:
|
||||
contains:
|
||||
const: nvidia,tegra210-admaif
|
||||
|
||||
then:
|
||||
properties:
|
||||
dmas:
|
||||
description:
|
||||
DMA channel specifiers, equally divided for Tx and Rx.
|
||||
minItems: 1
|
||||
maxItems: 20
|
||||
dma-names:
|
||||
items:
|
||||
pattern: "^[rt]x(10|[1-9])$"
|
||||
description:
|
||||
Should be "rx1", "rx2" ... "rx10" for DMA Rx channel
|
||||
Should be "tx1", "tx2" ... "tx10" for DMA Tx channel
|
||||
minItems: 1
|
||||
maxItems: 20
|
||||
|
||||
else:
|
||||
properties:
|
||||
dmas:
|
||||
description:
|
||||
DMA channel specifiers, equally divided for Tx and Rx.
|
||||
minItems: 1
|
||||
maxItems: 40
|
||||
dma-names:
|
||||
items:
|
||||
pattern: "^[rt]x(1[0-9]|[1-9]|20)$"
|
||||
description:
|
||||
Should be "rx1", "rx2" ... "rx20" for DMA Rx channel
|
||||
Should be "tx1", "tx2" ... "tx20" for DMA Tx channel
|
||||
minItems: 1
|
||||
maxItems: 40
|
||||
|
||||
required:
|
||||
- compatible
|
||||
- reg
|
||||
- dmas
|
||||
- dma-names
|
||||
|
||||
examples:
|
||||
- |
|
||||
admaif@702d0000 {
|
||||
compatible = "nvidia,tegra210-admaif";
|
||||
reg = <0x702d0000 0x800>;
|
||||
dmas = <&adma 1>, <&adma 1>,
|
||||
<&adma 2>, <&adma 2>,
|
||||
<&adma 3>, <&adma 3>,
|
||||
<&adma 4>, <&adma 4>,
|
||||
<&adma 5>, <&adma 5>,
|
||||
<&adma 6>, <&adma 6>,
|
||||
<&adma 7>, <&adma 7>,
|
||||
<&adma 8>, <&adma 8>,
|
||||
<&adma 9>, <&adma 9>,
|
||||
<&adma 10>, <&adma 10>;
|
||||
dma-names = "rx1", "tx1",
|
||||
"rx2", "tx2",
|
||||
"rx3", "tx3",
|
||||
"rx4", "tx4",
|
||||
"rx5", "tx5",
|
||||
"rx6", "tx6",
|
||||
"rx7", "tx7",
|
||||
"rx8", "tx8",
|
||||
"rx9", "tx9",
|
||||
"rx10", "tx10";
|
||||
};
|
||||
|
||||
...
|
||||
|
|
@ -0,0 +1,136 @@
|
|||
# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
|
||||
%YAML 1.2
|
||||
---
|
||||
$id: http://devicetree.org/schemas/sound/nvidia,tegra210-ahub.yaml#
|
||||
$schema: http://devicetree.org/meta-schemas/core.yaml#
|
||||
|
||||
title: Tegra210 AHUB Device Tree Bindings
|
||||
|
||||
description: |
|
||||
The Audio Hub (AHUB) comprises a collection of hardware accelerators
|
||||
for audio pre-processing, post-processing and a programmable full
|
||||
crossbar for routing audio data across these accelerators. It has
|
||||
external interfaces such as I2S, DMIC, DSPK. It interfaces with ADMA
|
||||
engine through ADMAIF.
|
||||
|
||||
maintainers:
|
||||
- Jon Hunter <jonathanh@nvidia.com>
|
||||
- Sameer Pujar <spujar@nvidia.com>
|
||||
|
||||
properties:
|
||||
$nodename:
|
||||
pattern: "^ahub@[0-9a-f]*$"
|
||||
|
||||
compatible:
|
||||
oneOf:
|
||||
- enum:
|
||||
- nvidia,tegra210-ahub
|
||||
- nvidia,tegra186-ahub
|
||||
- items:
|
||||
- const: nvidia,tegra194-ahub
|
||||
- const: nvidia,tegra186-ahub
|
||||
|
||||
reg:
|
||||
maxItems: 1
|
||||
|
||||
clocks:
|
||||
maxItems: 1
|
||||
|
||||
clock-names:
|
||||
const: ahub
|
||||
|
||||
assigned-clocks:
|
||||
maxItems: 1
|
||||
|
||||
assigned-clock-parents:
|
||||
maxItems: 1
|
||||
|
||||
assigned-clock-rates:
|
||||
maxItems: 1
|
||||
|
||||
"#address-cells":
|
||||
const: 1
|
||||
|
||||
"#size-cells":
|
||||
const: 1
|
||||
|
||||
ranges: true
|
||||
|
||||
required:
|
||||
- compatible
|
||||
- reg
|
||||
- clocks
|
||||
- clock-names
|
||||
- assigned-clocks
|
||||
- assigned-clock-parents
|
||||
- "#address-cells"
|
||||
- "#size-cells"
|
||||
- ranges
|
||||
|
||||
examples:
|
||||
- |
|
||||
#include<dt-bindings/clock/tegra210-car.h>
|
||||
|
||||
ahub@702d0800 {
|
||||
compatible = "nvidia,tegra210-ahub";
|
||||
reg = <0x702d0800 0x800>;
|
||||
clocks = <&tegra_car TEGRA210_CLK_D_AUDIO>;
|
||||
clock-names = "ahub";
|
||||
assigned-clocks = <&tegra_car TEGRA210_CLK_D_AUDIO>;
|
||||
assigned-clock-parents = <&tegra_car TEGRA210_CLK_PLL_A_OUT0>;
|
||||
#address-cells = <1>;
|
||||
#size-cells = <1>;
|
||||
ranges = <0x702d0000 0x702d0000 0x0000e400>;
|
||||
|
||||
// All AHUB child nodes below
|
||||
admaif@702d0000 {
|
||||
compatible = "nvidia,tegra210-admaif";
|
||||
reg = <0x702d0000 0x800>;
|
||||
dmas = <&adma 1>, <&adma 1>,
|
||||
<&adma 2>, <&adma 2>,
|
||||
<&adma 3>, <&adma 3>,
|
||||
<&adma 4>, <&adma 4>,
|
||||
<&adma 5>, <&adma 5>,
|
||||
<&adma 6>, <&adma 6>,
|
||||
<&adma 7>, <&adma 7>,
|
||||
<&adma 8>, <&adma 8>,
|
||||
<&adma 9>, <&adma 9>,
|
||||
<&adma 10>, <&adma 10>;
|
||||
dma-names = "rx1", "tx1",
|
||||
"rx2", "tx2",
|
||||
"rx3", "tx3",
|
||||
"rx4", "tx4",
|
||||
"rx5", "tx5",
|
||||
"rx6", "tx6",
|
||||
"rx7", "tx7",
|
||||
"rx8", "tx8",
|
||||
"rx9", "tx9",
|
||||
"rx10", "tx10";
|
||||
};
|
||||
|
||||
i2s@702d1000 {
|
||||
compatible = "nvidia,tegra210-i2s";
|
||||
reg = <0x702d1000 0x100>;
|
||||
clocks = <&tegra_car TEGRA210_CLK_I2S0>;
|
||||
clock-names = "i2s";
|
||||
assigned-clocks = <&tegra_car TEGRA210_CLK_I2S0>;
|
||||
assigned-clock-parents = <&tegra_car TEGRA210_CLK_PLL_A_OUT0>;
|
||||
assigned-clock-rates = <1536000>;
|
||||
sound-name-prefix = "I2S1";
|
||||
};
|
||||
|
||||
dmic@702d4000 {
|
||||
compatible = "nvidia,tegra210-dmic";
|
||||
reg = <0x702d4000 0x100>;
|
||||
clocks = <&tegra_car TEGRA210_CLK_DMIC1>;
|
||||
clock-names = "dmic";
|
||||
assigned-clocks = <&tegra_car TEGRA210_CLK_DMIC1>;
|
||||
assigned-clock-parents = <&tegra_car TEGRA210_CLK_PLL_A_OUT0>;
|
||||
assigned-clock-rates = <3072000>;
|
||||
sound-name-prefix = "DMIC1";
|
||||
};
|
||||
|
||||
// More child nodes to follow
|
||||
};
|
||||
|
||||
...
|
||||
|
|
@ -0,0 +1,83 @@
|
|||
# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
|
||||
%YAML 1.2
|
||||
---
|
||||
$id: http://devicetree.org/schemas/sound/nvidia,tegra210-dmic.yaml#
|
||||
$schema: http://devicetree.org/meta-schemas/core.yaml#
|
||||
|
||||
title: Tegra210 DMIC Controller Device Tree Bindings
|
||||
|
||||
description: |
|
||||
The Digital MIC (DMIC) Controller is used to interface with Pulse
|
||||
Density Modulation (PDM) input devices. It converts PDM signals to
|
||||
Pulse Coded Modulation (PCM) signals. DMIC can be viewed as a PDM
|
||||
receiver.
|
||||
|
||||
maintainers:
|
||||
- Jon Hunter <jonathanh@nvidia.com>
|
||||
- Sameer Pujar <spujar@nvidia.com>
|
||||
|
||||
properties:
|
||||
$nodename:
|
||||
pattern: "^dmic@[0-9a-f]*$"
|
||||
|
||||
compatible:
|
||||
oneOf:
|
||||
- const: nvidia,tegra210-dmic
|
||||
- items:
|
||||
- enum:
|
||||
- nvidia,tegra194-dmic
|
||||
- nvidia,tegra186-dmic
|
||||
- const: nvidia,tegra210-dmic
|
||||
|
||||
reg:
|
||||
maxItems: 1
|
||||
|
||||
clocks:
|
||||
maxItems: 1
|
||||
|
||||
clock-names:
|
||||
const: dmic
|
||||
|
||||
assigned-clocks:
|
||||
maxItems: 1
|
||||
|
||||
assigned-clock-parents:
|
||||
maxItems: 1
|
||||
|
||||
assigned-clock-rates:
|
||||
maxItems: 1
|
||||
|
||||
sound-name-prefix:
|
||||
pattern: "^DMIC[1-9]$"
|
||||
allOf:
|
||||
- $ref: /schemas/types.yaml#/definitions/string
|
||||
description:
|
||||
used as prefix for sink/source names of the component. Must be a
|
||||
unique string among multiple instances of the same component.
|
||||
The name can be "DMIC1" or "DMIC2" ... "DMICx", where x depends
|
||||
on the maximum available instances on a Tegra SoC.
|
||||
|
||||
required:
|
||||
- compatible
|
||||
- reg
|
||||
- clocks
|
||||
- clock-names
|
||||
- assigned-clocks
|
||||
- assigned-clock-parents
|
||||
|
||||
examples:
|
||||
- |
|
||||
#include<dt-bindings/clock/tegra210-car.h>
|
||||
|
||||
dmic@702d4000 {
|
||||
compatible = "nvidia,tegra210-dmic";
|
||||
reg = <0x702d4000 0x100>;
|
||||
clocks = <&tegra_car TEGRA210_CLK_DMIC1>;
|
||||
clock-names = "dmic";
|
||||
assigned-clocks = <&tegra_car TEGRA210_CLK_DMIC1>;
|
||||
assigned-clock-parents = <&tegra_car TEGRA210_CLK_PLL_A_OUT0>;
|
||||
assigned-clock-rates = <3072000>;
|
||||
sound-name-prefix = "DMIC1";
|
||||
};
|
||||
|
||||
...
|
||||
|
|
@ -0,0 +1,101 @@
|
|||
# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
|
||||
%YAML 1.2
|
||||
---
|
||||
$id: http://devicetree.org/schemas/sound/nvidia,tegra210-i2s.yaml#
|
||||
$schema: http://devicetree.org/meta-schemas/core.yaml#
|
||||
|
||||
title: Tegra210 I2S Controller Device Tree Bindings
|
||||
|
||||
description: |
|
||||
The Inter-IC Sound (I2S) controller implements full-duplex,
|
||||
bi-directional and single direction point-to-point serial
|
||||
interfaces. It can interface with I2S compatible devices.
|
||||
I2S controller can operate both in master and slave mode.
|
||||
|
||||
maintainers:
|
||||
- Jon Hunter <jonathanh@nvidia.com>
|
||||
- Sameer Pujar <spujar@nvidia.com>
|
||||
|
||||
properties:
|
||||
$nodename:
|
||||
pattern: "^i2s@[0-9a-f]*$"
|
||||
|
||||
compatible:
|
||||
oneOf:
|
||||
- const: nvidia,tegra210-i2s
|
||||
- items:
|
||||
- enum:
|
||||
- nvidia,tegra194-i2s
|
||||
- nvidia,tegra186-i2s
|
||||
- const: nvidia,tegra210-i2s
|
||||
|
||||
reg:
|
||||
maxItems: 1
|
||||
|
||||
clocks:
|
||||
minItems: 1
|
||||
maxItems: 2
|
||||
items:
|
||||
- description: I2S bit clock
|
||||
- description:
|
||||
Sync input clock, which can act as clock source to other I/O
|
||||
modules in AHUB. The Tegra I2S driver sets this clock rate as
|
||||
per bit clock rate. I/O module which wants to use this clock
|
||||
as source, can mention this clock as parent in the DT bindings.
|
||||
This is an optional clock entry, since it is only required when
|
||||
some other I/O wants to reference from a particular I2Sx
|
||||
instance.
|
||||
|
||||
clock-names:
|
||||
minItems: 1
|
||||
maxItems: 2
|
||||
items:
|
||||
- const: i2s
|
||||
- const: sync_input
|
||||
|
||||
assigned-clocks:
|
||||
minItems: 1
|
||||
maxItems: 2
|
||||
|
||||
assigned-clock-parents:
|
||||
minItems: 1
|
||||
maxItems: 2
|
||||
|
||||
assigned-clock-rates:
|
||||
minItems: 1
|
||||
maxItems: 2
|
||||
|
||||
sound-name-prefix:
|
||||
pattern: "^I2S[1-9]$"
|
||||
allOf:
|
||||
- $ref: /schemas/types.yaml#/definitions/string
|
||||
description:
|
||||
Used as prefix for sink/source names of the component. Must be a
|
||||
unique string among multiple instances of the same component.
|
||||
The name can be "I2S1" or "I2S2" ... "I2Sx", where x depends
|
||||
on the maximum available instances on a Tegra SoC.
|
||||
|
||||
required:
|
||||
- compatible
|
||||
- reg
|
||||
- clocks
|
||||
- clock-names
|
||||
- assigned-clocks
|
||||
- assigned-clock-parents
|
||||
|
||||
examples:
|
||||
- |
|
||||
#include<dt-bindings/clock/tegra210-car.h>
|
||||
|
||||
i2s@702d1000 {
|
||||
compatible = "nvidia,tegra210-i2s";
|
||||
reg = <0x702d1000 0x100>;
|
||||
clocks = <&tegra_car TEGRA210_CLK_I2S0>;
|
||||
clock-names = "i2s";
|
||||
assigned-clocks = <&tegra_car TEGRA210_CLK_I2S0>;
|
||||
assigned-clock-parents = <&tegra_car TEGRA210_CLK_PLL_A_OUT0>;
|
||||
assigned-clock-rates = <1536000>;
|
||||
sound-name-prefix = "I2S1";
|
||||
};
|
||||
|
||||
...
|
||||
|
|
@ -39,9 +39,9 @@ configuration of each dai. Must contain the following properties.
|
|||
Usage: Required for Compress offload dais
|
||||
Value type: <u32>
|
||||
Definition: Specifies the direction of the dai stream
|
||||
0 for both tx and rx
|
||||
1 for only tx (Capture/Encode)
|
||||
2 for only rx (Playback/Decode)
|
||||
Q6ASM_DAI_TX_RX (0) for both tx and rx
|
||||
Q6ASM_DAI_TX (1) for only tx (Capture/Encode)
|
||||
Q6ASM_DAI_RX (2) for only rx (Playback/Decode)
|
||||
|
||||
- is-compress-dai:
|
||||
Usage: Required for Compress offload dais
|
||||
|
|
@ -50,6 +50,7 @@ configuration of each dai. Must contain the following properties.
|
|||
|
||||
|
||||
= EXAMPLE
|
||||
#include <dt-bindings/sound/qcom,q6asm.h>
|
||||
|
||||
apr-service@7 {
|
||||
compatible = "qcom,q6asm";
|
||||
|
|
@ -62,7 +63,7 @@ apr-service@7 {
|
|||
|
||||
dai@0 {
|
||||
reg = <0>;
|
||||
direction = <2>;
|
||||
direction = <Q6ASM_DAI_RX>;
|
||||
is-compress-dai;
|
||||
};
|
||||
};
|
||||
|
|
|
|||
|
|
@ -43,30 +43,19 @@ properties:
|
|||
'#sound-dai-cells':
|
||||
const: 1
|
||||
|
||||
fsia,spdif-connection:
|
||||
patternProperties:
|
||||
"^fsi(a|b),spdif-connection$":
|
||||
$ref: /schemas/types.yaml#/definitions/flag
|
||||
description: FSI is connected by S/PDIF
|
||||
|
||||
fsia,stream-mode-support:
|
||||
"^fsi(a|b),stream-mode-support$":
|
||||
$ref: /schemas/types.yaml#/definitions/flag
|
||||
description: FSI supports 16bit stream mode
|
||||
|
||||
fsia,use-internal-clock:
|
||||
"^fsi(a|b),use-internal-clock$":
|
||||
$ref: /schemas/types.yaml#/definitions/flag
|
||||
description: FSI uses internal clock when master mode
|
||||
|
||||
fsib,spdif-connection:
|
||||
$ref: /schemas/types.yaml#/definitions/flag
|
||||
description: same as fsia
|
||||
|
||||
fsib,stream-mode-support:
|
||||
$ref: /schemas/types.yaml#/definitions/flag
|
||||
description: same as fsia
|
||||
|
||||
fsib,use-internal-clock:
|
||||
$ref: /schemas/types.yaml#/definitions/flag
|
||||
description: same as fsia
|
||||
|
||||
required:
|
||||
- compatible
|
||||
- reg
|
||||
|
|
|
|||
|
|
@ -271,6 +271,7 @@ Required properties:
|
|||
- "renesas,rcar_sound-r8a774a1" (RZ/G2M)
|
||||
- "renesas,rcar_sound-r8a774b1" (RZ/G2N)
|
||||
- "renesas,rcar_sound-r8a774c0" (RZ/G2E)
|
||||
- "renesas,rcar_sound-r8a774e1" (RZ/G2H)
|
||||
- "renesas,rcar_sound-r8a7778" (R-Car M1A)
|
||||
- "renesas,rcar_sound-r8a7779" (R-Car H1)
|
||||
- "renesas,rcar_sound-r8a7790" (R-Car H2)
|
||||
|
|
|
|||
|
|
@ -1,28 +0,0 @@
|
|||
* Rockchip Rk3328 internal codec
|
||||
|
||||
Required properties:
|
||||
|
||||
- compatible: "rockchip,rk3328-codec"
|
||||
- reg: physical base address of the controller and length of memory mapped
|
||||
region.
|
||||
- rockchip,grf: the phandle of the syscon node for GRF register.
|
||||
- clocks: a list of phandle + clock-specifer pairs, one for each entry in clock-names.
|
||||
- clock-names: should be "pclk".
|
||||
- spk-depop-time-ms: speak depop time msec.
|
||||
|
||||
Optional properties:
|
||||
|
||||
- mute-gpios: GPIO specifier for external line driver control (typically the
|
||||
dedicated GPIO_MUTE pin)
|
||||
|
||||
Example for rk3328 internal codec:
|
||||
|
||||
codec: codec@ff410000 {
|
||||
compatible = "rockchip,rk3328-codec";
|
||||
reg = <0x0 0xff410000 0x0 0x1000>;
|
||||
rockchip,grf = <&grf>;
|
||||
clocks = <&cru PCLK_ACODEC>;
|
||||
clock-names = "pclk";
|
||||
mute-gpios = <&grf_gpio 0 GPIO_ACTIVE_LOW>;
|
||||
spk-depop-time-ms = 100;
|
||||
};
|
||||
|
|
@ -0,0 +1,69 @@
|
|||
# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
|
||||
%YAML 1.2
|
||||
---
|
||||
$id: http://devicetree.org/schemas/sound/rockchip,rk3328-codec.yaml#
|
||||
$schema: http://devicetree.org/meta-schemas/core.yaml#
|
||||
|
||||
title: Rockchip rk3328 internal codec
|
||||
|
||||
maintainers:
|
||||
- Heiko Stuebner <heiko@sntech.de>
|
||||
|
||||
properties:
|
||||
compatible:
|
||||
const: rockchip,rk3328-codec
|
||||
|
||||
reg:
|
||||
maxItems: 1
|
||||
|
||||
clocks:
|
||||
items:
|
||||
- description: clock for audio codec
|
||||
- description: clock for I2S master clock
|
||||
|
||||
clock-names:
|
||||
items:
|
||||
- const: pclk
|
||||
- const: mclk
|
||||
|
||||
rockchip,grf:
|
||||
$ref: /schemas/types.yaml#/definitions/phandle
|
||||
description:
|
||||
The phandle of the syscon node for the GRF register.
|
||||
|
||||
spk-depop-time-ms:
|
||||
default: 200
|
||||
description:
|
||||
Speaker depop time in msec.
|
||||
|
||||
mute-gpios:
|
||||
maxItems: 1
|
||||
description:
|
||||
GPIO specifier for external line driver control (typically the
|
||||
dedicated GPIO_MUTE pin)
|
||||
|
||||
"#sound-dai-cells":
|
||||
const: 0
|
||||
|
||||
required:
|
||||
- compatible
|
||||
- reg
|
||||
- clocks
|
||||
- clock-names
|
||||
- rockchip,grf
|
||||
- "#sound-dai-cells"
|
||||
|
||||
examples:
|
||||
- |
|
||||
#include <dt-bindings/gpio/gpio.h>
|
||||
#include <dt-bindings/clock/rk3328-cru.h>
|
||||
codec: codec@ff410000 {
|
||||
compatible = "rockchip,rk3328-codec";
|
||||
reg = <0xff410000 0x1000>;
|
||||
clocks = <&cru PCLK_ACODECPHY>, <&cru SCLK_I2S1>;
|
||||
clock-names = "pclk", "mclk";
|
||||
rockchip,grf = <&grf>;
|
||||
mute-gpios = <&grf_gpio 0 GPIO_ACTIVE_LOW>;
|
||||
spk-depop-time-ms = <100>;
|
||||
#sound-dai-cells = <0>;
|
||||
};
|
||||
|
|
@ -1,29 +0,0 @@
|
|||
ROHM BD28623MUV Class D speaker amplifier for digital input
|
||||
|
||||
This codec does not have any control buses such as I2C, it detect format and
|
||||
rate of I2S signal automatically. It has two signals that can be connected
|
||||
to GPIOs: reset and mute.
|
||||
|
||||
Required properties:
|
||||
- compatible : should be "rohm,bd28623"
|
||||
- #sound-dai-cells: should be 0.
|
||||
- VCCA-supply : regulator phandle for the VCCA supply
|
||||
- VCCP1-supply : regulator phandle for the VCCP1 supply
|
||||
- VCCP2-supply : regulator phandle for the VCCP2 supply
|
||||
|
||||
Optional properties:
|
||||
- reset-gpios : GPIO specifier for the active low reset line
|
||||
- mute-gpios : GPIO specifier for the active low mute line
|
||||
|
||||
Example:
|
||||
|
||||
codec {
|
||||
compatible = "rohm,bd28623";
|
||||
#sound-dai-cells = <0>;
|
||||
|
||||
VCCA-supply = <&vcc_reg>;
|
||||
VCCP1-supply = <&vcc_reg>;
|
||||
VCCP2-supply = <&vcc_reg>;
|
||||
reset-gpios = <&gpio 0 GPIO_ACTIVE_LOW>;
|
||||
mute-gpios = <&gpio 1 GPIO_ACTIVE_LOW>;
|
||||
};
|
||||
|
|
@ -0,0 +1,67 @@
|
|||
# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
|
||||
%YAML 1.2
|
||||
---
|
||||
$id: http://devicetree.org/schemas/sound/rohm,bd28623.yaml#
|
||||
$schema: http://devicetree.org/meta-schemas/core.yaml#
|
||||
|
||||
title: ROHM BD28623MUV Class D speaker amplifier for digital input
|
||||
|
||||
description:
|
||||
This codec does not have any control buses such as I2C, it detect
|
||||
format and rate of I2S signal automatically. It has two signals
|
||||
that can be connected to GPIOs reset and mute.
|
||||
|
||||
maintainers:
|
||||
- Katsuhiro Suzuki <katsuhiro@katsuster.net>
|
||||
|
||||
properties:
|
||||
compatible:
|
||||
const: rohm,bd28623
|
||||
|
||||
"#sound-dai-cells":
|
||||
const: 0
|
||||
|
||||
VCCA-supply:
|
||||
description:
|
||||
regulator phandle for the VCCA (for analog) power supply
|
||||
|
||||
VCCP1-supply:
|
||||
description:
|
||||
regulator phandle for the VCCP1 (for ch1) power supply
|
||||
|
||||
VCCP2-supply:
|
||||
description:
|
||||
regulator phandle for the VCCP2 (for ch2) power supply
|
||||
|
||||
reset-gpios:
|
||||
maxItems: 1
|
||||
description:
|
||||
GPIO specifier for the active low reset line
|
||||
|
||||
mute-gpios:
|
||||
maxItems: 1
|
||||
description:
|
||||
GPIO specifier for the active low mute line
|
||||
|
||||
required:
|
||||
- compatible
|
||||
- VCCA-supply
|
||||
- VCCP1-supply
|
||||
- VCCP2-supply
|
||||
- "#sound-dai-cells"
|
||||
|
||||
additionalProperties: false
|
||||
|
||||
examples:
|
||||
- |
|
||||
#include <dt-bindings/gpio/gpio.h>
|
||||
codec {
|
||||
compatible = "rohm,bd28623";
|
||||
#sound-dai-cells = <0>;
|
||||
|
||||
VCCA-supply = <&vcc_reg>;
|
||||
VCCP1-supply = <&vcc_reg>;
|
||||
VCCP2-supply = <&vcc_reg>;
|
||||
reset-gpios = <&gpio 0 GPIO_ACTIVE_LOW>;
|
||||
mute-gpios = <&gpio 1 GPIO_ACTIVE_LOW>;
|
||||
};
|
||||
|
|
@ -0,0 +1,147 @@
|
|||
# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
|
||||
%YAML 1.2
|
||||
---
|
||||
$id: http://devicetree.org/schemas/sound/samsung,aries-wm8994.yaml#
|
||||
$schema: http://devicetree.org/meta-schemas/core.yaml#
|
||||
|
||||
title: Samsung Aries audio complex with WM8994 codec
|
||||
|
||||
maintainers:
|
||||
- Jonathan Bakker <xc-racer2@live.ca>
|
||||
|
||||
properties:
|
||||
compatible:
|
||||
oneOf:
|
||||
- const: samsung,aries-wm8994
|
||||
description: With FM radio and modem master
|
||||
|
||||
- const: samsung,fascinate4g-wm8994
|
||||
description: Without FM radio and modem slave
|
||||
|
||||
model:
|
||||
$ref: /schemas/types.yaml#/definitions/string
|
||||
description: The user-visible name of this sound complex.
|
||||
|
||||
cpu:
|
||||
type: object
|
||||
properties:
|
||||
sound-dai:
|
||||
minItems: 2
|
||||
maxItems: 2
|
||||
$ref: /schemas/types.yaml#/definitions/phandle-array
|
||||
description: |
|
||||
phandles to the I2S controller and bluetooth codec,
|
||||
in that order
|
||||
|
||||
codec:
|
||||
type: object
|
||||
properties:
|
||||
sound-dai:
|
||||
$ref: /schemas/types.yaml#/definitions/phandle-array
|
||||
description: phandle to the WM8994 CODEC
|
||||
|
||||
samsung,audio-routing:
|
||||
$ref: /schemas/types.yaml#/definitions/non-unique-string-array
|
||||
description: |
|
||||
List of the connections between audio
|
||||
components; each entry is a pair of strings, the first being the
|
||||
connection's sink, the second being the connection's source;
|
||||
valid names for sources and sinks are the WM8994's pins (as
|
||||
documented in its binding), and the jacks on the board -
|
||||
For samsung,aries-wm8994: HP, SPK, RCV, LINE, Main Mic, Headset Mic,
|
||||
or FM In
|
||||
For samsung,fascinate4g-wm8994: HP, SPK, RCV, LINE, Main Mic,
|
||||
or HeadsetMic
|
||||
|
||||
extcon:
|
||||
description: Extcon phandle for dock detection
|
||||
|
||||
main-micbias-supply:
|
||||
description: Supply for the micbias on the main mic
|
||||
|
||||
headset-micbias-supply:
|
||||
description: Supply for the micbias on the headset mic
|
||||
|
||||
earpath-sel-gpios:
|
||||
description: GPIO for switching between tv-out and mic paths
|
||||
|
||||
headset-detect-gpios:
|
||||
description: GPIO for detection of headset insertion
|
||||
|
||||
headset-key-gpios:
|
||||
description: GPIO for detection of headset key press
|
||||
|
||||
io-channels:
|
||||
maxItems: 1
|
||||
description: IO channel to read micbias voltage for headset detection
|
||||
|
||||
io-channel-names:
|
||||
const: headset-detect
|
||||
|
||||
required:
|
||||
- compatible
|
||||
- model
|
||||
- cpu
|
||||
- codec
|
||||
- samsung,audio-routing
|
||||
- extcon
|
||||
- main-micbias-supply
|
||||
- headset-micbias-supply
|
||||
- earpath-sel-gpios
|
||||
- headset-detect-gpios
|
||||
- headset-key-gpios
|
||||
|
||||
additionalProperties: false
|
||||
|
||||
examples:
|
||||
- |
|
||||
#include <dt-bindings/gpio/gpio.h>
|
||||
|
||||
sound {
|
||||
compatible = "samsung,fascinate4g-wm8994";
|
||||
|
||||
model = "Fascinate4G";
|
||||
|
||||
extcon = <&fsa9480>;
|
||||
|
||||
main-micbias-supply = <&main_micbias_reg>;
|
||||
headset-micbias-supply = <&headset_micbias_reg>;
|
||||
|
||||
earpath-sel-gpios = <&gpj2 6 GPIO_ACTIVE_HIGH>;
|
||||
|
||||
io-channels = <&adc 3>;
|
||||
io-channel-names = "headset-detect";
|
||||
headset-detect-gpios = <&gph0 6 GPIO_ACTIVE_HIGH>;
|
||||
headset-key-gpios = <&gph3 6 GPIO_ACTIVE_HIGH>;
|
||||
|
||||
samsung,audio-routing =
|
||||
"HP", "HPOUT1L",
|
||||
"HP", "HPOUT1R",
|
||||
|
||||
"SPK", "SPKOUTLN",
|
||||
"SPK", "SPKOUTLP",
|
||||
|
||||
"RCV", "HPOUT2N",
|
||||
"RCV", "HPOUT2P",
|
||||
|
||||
"LINE", "LINEOUT2N",
|
||||
"LINE", "LINEOUT2P",
|
||||
|
||||
"IN1LP", "Main Mic",
|
||||
"IN1LN", "Main Mic",
|
||||
|
||||
"IN1RP", "Headset Mic",
|
||||
"IN1RN", "Headset Mic";
|
||||
|
||||
pinctrl-names = "default";
|
||||
pinctrl-0 = <&headset_det &earpath_sel>;
|
||||
|
||||
cpu {
|
||||
sound-dai = <&i2s0>, <&bt_codec>;
|
||||
};
|
||||
|
||||
codec {
|
||||
sound-dai = <&wm8994>;
|
||||
};
|
||||
};
|
||||
|
||||
|
|
@ -0,0 +1,108 @@
|
|||
# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
|
||||
%YAML 1.2
|
||||
---
|
||||
$id: http://devicetree.org/schemas/sound/samsung,midas-audio.yaml#
|
||||
$schema: http://devicetree.org/meta-schemas/core.yaml#
|
||||
|
||||
title: Samsung Midas audio complex with WM1811 codec
|
||||
|
||||
maintainers:
|
||||
- Sylwester Nawrocki <s.nawrocki@samsung.com>
|
||||
|
||||
properties:
|
||||
compatible:
|
||||
const: samsung,midas-audio
|
||||
|
||||
model:
|
||||
$ref: /schemas/types.yaml#/definitions/string
|
||||
description: The user-visible name of this sound complex.
|
||||
|
||||
cpu:
|
||||
type: object
|
||||
properties:
|
||||
sound-dai:
|
||||
$ref: /schemas/types.yaml#/definitions/phandle
|
||||
description: phandle to the I2S controller
|
||||
required:
|
||||
- sound-dai
|
||||
|
||||
codec:
|
||||
type: object
|
||||
properties:
|
||||
sound-dai:
|
||||
$ref: /schemas/types.yaml#/definitions/phandle
|
||||
description: phandle to the WM1811 CODEC
|
||||
required:
|
||||
- sound-dai
|
||||
|
||||
samsung,audio-routing:
|
||||
$ref: /schemas/types.yaml#/definitions/non-unique-string-array
|
||||
description: |
|
||||
List of the connections between audio components; each entry is
|
||||
a pair of strings, the first being the connection's sink, the second
|
||||
being the connection's source; valid names for sources and sinks are
|
||||
the WM1811's pins (as documented in its binding), and the jacks
|
||||
on the board: HP, SPK, Main Mic, Sub Mic, Headset Mic.
|
||||
|
||||
mic-bias-supply:
|
||||
description: Supply for the micbias on the Main microphone
|
||||
|
||||
submic-bias-supply:
|
||||
description: Supply for the micbias on the Sub microphone
|
||||
|
||||
fm-sel-gpios:
|
||||
description: GPIO pin for FM selection
|
||||
|
||||
lineout-sel-gpios:
|
||||
description: GPIO pin for line out selection
|
||||
|
||||
required:
|
||||
- compatible
|
||||
- model
|
||||
- cpu
|
||||
- codec
|
||||
- samsung,audio-routing
|
||||
- mic-bias-supply
|
||||
- submic-bias-supply
|
||||
|
||||
additionalProperties: false
|
||||
|
||||
examples:
|
||||
- |
|
||||
#include <dt-bindings/gpio/gpio.h>
|
||||
|
||||
sound {
|
||||
compatible = "samsung,midas-audio";
|
||||
model = "Midas";
|
||||
|
||||
fm-sel-gpios = <&gpaa0 3 GPIO_ACTIVE_HIGH>;
|
||||
|
||||
mic-bias-supply = <&mic_bias_reg>;
|
||||
submic-bias-supply = <&submic_bias_reg>;
|
||||
|
||||
samsung,audio-routing =
|
||||
"HP", "HPOUT1L",
|
||||
"HP", "HPOUT1R",
|
||||
|
||||
"SPK", "SPKOUTLN",
|
||||
"SPK", "SPKOUTLP",
|
||||
"SPK", "SPKOUTRN",
|
||||
"SPK", "SPKOUTRP",
|
||||
|
||||
"RCV", "HPOUT2N",
|
||||
"RCV", "HPOUT2P",
|
||||
|
||||
"IN1LP", "Main Mic",
|
||||
"IN1LN", "Main Mic",
|
||||
"IN1RP", "Sub Mic",
|
||||
"IN1LP", "Sub Mic";
|
||||
|
||||
cpu {
|
||||
sound-dai = <&i2s0>;
|
||||
};
|
||||
|
||||
codec {
|
||||
sound-dai = <&wm1811>;
|
||||
};
|
||||
|
||||
};
|
||||
|
|
@ -1,60 +0,0 @@
|
|||
* Freescale SGTL5000 Stereo Codec
|
||||
|
||||
Required properties:
|
||||
- compatible : "fsl,sgtl5000".
|
||||
|
||||
- reg : the I2C address of the device
|
||||
|
||||
- #sound-dai-cells: must be equal to 0
|
||||
|
||||
- clocks : the clock provider of SYS_MCLK
|
||||
|
||||
- VDDA-supply : the regulator provider of VDDA
|
||||
|
||||
- VDDIO-supply: the regulator provider of VDDIO
|
||||
|
||||
Optional properties:
|
||||
|
||||
- VDDD-supply : the regulator provider of VDDD
|
||||
|
||||
- micbias-resistor-k-ohms : the bias resistor to be used in kOhms
|
||||
The resistor can take values of 2k, 4k or 8k.
|
||||
If set to 0 it will be off.
|
||||
If this node is not mentioned or if the value is unknown, then
|
||||
micbias resistor is set to 4K.
|
||||
|
||||
- micbias-voltage-m-volts : the bias voltage to be used in mVolts
|
||||
The voltage can take values from 1.25V to 3V by 250mV steps
|
||||
If this node is not mentioned or the value is unknown, then
|
||||
the value is set to 1.25V.
|
||||
|
||||
- lrclk-strength: the LRCLK pad strength. Possible values are:
|
||||
0, 1, 2 and 3 as per the table below:
|
||||
|
||||
VDDIO 1.8V 2.5V 3.3V
|
||||
0 = Disable
|
||||
1 = 1.66 mA 2.87 mA 4.02 mA
|
||||
2 = 3.33 mA 5.74 mA 8.03 mA
|
||||
3 = 4.99 mA 8.61 mA 12.05 mA
|
||||
|
||||
- sclk-strength: the SCLK pad strength. Possible values are:
|
||||
0, 1, 2 and 3 as per the table below:
|
||||
|
||||
VDDIO 1.8V 2.5V 3.3V
|
||||
0 = Disable
|
||||
1 = 1.66 mA 2.87 mA 4.02 mA
|
||||
2 = 3.33 mA 5.74 mA 8.03 mA
|
||||
3 = 4.99 mA 8.61 mA 12.05 mA
|
||||
|
||||
Example:
|
||||
|
||||
sgtl5000: codec@a {
|
||||
compatible = "fsl,sgtl5000";
|
||||
reg = <0x0a>;
|
||||
#sound-dai-cells = <0>;
|
||||
clocks = <&clks 150>;
|
||||
micbias-resistor-k-ohms = <2>;
|
||||
micbias-voltage-m-volts = <2250>;
|
||||
VDDA-supply = <®_3p3v>;
|
||||
VDDIO-supply = <®_3p3v>;
|
||||
};
|
||||
|
|
@ -0,0 +1,103 @@
|
|||
# SPDX-License-Identifier: GPL-2.0-only
|
||||
%YAML 1.2
|
||||
---
|
||||
$id: http://devicetree.org/schemas/sound/sgtl5000.yaml#
|
||||
$schema: http://devicetree.org/meta-schemas/core.yaml#
|
||||
|
||||
title: Freescale SGTL5000 Stereo Codec
|
||||
|
||||
maintainers:
|
||||
- Fabio Estevam <festevam@gmail.com>
|
||||
|
||||
properties:
|
||||
compatible:
|
||||
const: fsl,sgtl5000
|
||||
|
||||
reg:
|
||||
maxItems: 1
|
||||
|
||||
"#sound-dai-cells":
|
||||
const: 0
|
||||
|
||||
clocks:
|
||||
items:
|
||||
- description: the clock provider of SYS_MCLK
|
||||
|
||||
VDDA-supply:
|
||||
description: the regulator provider of VDDA
|
||||
|
||||
VDDIO-supply:
|
||||
description: the regulator provider of VDDIO
|
||||
|
||||
VDDD-supply:
|
||||
description: the regulator provider of VDDD
|
||||
|
||||
micbias-resistor-k-ohms:
|
||||
description: The bias resistor to be used in kOhms. The resistor can take
|
||||
values of 2k, 4k or 8k. If set to 0 it will be off. If this node is not
|
||||
mentioned or if the value is unknown, then micbias resistor is set to
|
||||
4k.
|
||||
$ref: "/schemas/types.yaml#/definitions/uint32"
|
||||
enum: [ 0, 2, 4, 8 ]
|
||||
|
||||
micbias-voltage-m-volts:
|
||||
description: The bias voltage to be used in mVolts. The voltage can take
|
||||
values from 1.25V to 3V by 250mV steps. If this node is not mentioned
|
||||
or the value is unknown, then the value is set to 1.25V.
|
||||
$ref: "/schemas/types.yaml#/definitions/uint32"
|
||||
enum: [ 1250, 1500, 1750, 2000, 2250, 2500, 2750, 3000 ]
|
||||
|
||||
lrclk-strength:
|
||||
description: |
|
||||
The LRCLK pad strength. Possible values are: 0, 1, 2 and 3 as per the
|
||||
table below:
|
||||
|
||||
VDDIO 1.8V 2.5V 3.3V
|
||||
0 = Disable
|
||||
1 = 1.66 mA 2.87 mA 4.02 mA
|
||||
2 = 3.33 mA 5.74 mA 8.03 mA
|
||||
3 = 4.99 mA 8.61 mA 12.05 mA
|
||||
$ref: "/schemas/types.yaml#/definitions/uint32"
|
||||
enum: [ 0, 1, 2, 3 ]
|
||||
|
||||
sclk-strength:
|
||||
description: |
|
||||
The SCLK pad strength. Possible values are: 0, 1, 2 and 3 as per the
|
||||
table below:
|
||||
|
||||
VDDIO 1.8V 2.5V 3.3V
|
||||
0 = Disable
|
||||
1 = 1.66 mA 2.87 mA 4.02 mA
|
||||
2 = 3.33 mA 5.74 mA 8.03 mA
|
||||
3 = 4.99 mA 8.61 mA 12.05 mA
|
||||
$ref: "/schemas/types.yaml#/definitions/uint32"
|
||||
enum: [ 0, 1, 2, 3 ]
|
||||
|
||||
required:
|
||||
- compatible
|
||||
- reg
|
||||
- "#sound-dai-cells"
|
||||
- clocks
|
||||
- VDDA-supply
|
||||
- VDDIO-supply
|
||||
|
||||
additionalProperties: false
|
||||
|
||||
examples:
|
||||
- |
|
||||
i2c {
|
||||
#address-cells = <1>;
|
||||
#size-cells = <0>;
|
||||
|
||||
codec@a {
|
||||
compatible = "fsl,sgtl5000";
|
||||
reg = <0x0a>;
|
||||
#sound-dai-cells = <0>;
|
||||
clocks = <&clks 150>;
|
||||
micbias-resistor-k-ohms = <2>;
|
||||
micbias-voltage-m-volts = <2250>;
|
||||
VDDA-supply = <®_3p3v>;
|
||||
VDDIO-supply = <®_3p3v>;
|
||||
};
|
||||
};
|
||||
...
|
||||
|
|
@ -0,0 +1,81 @@
|
|||
# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause
|
||||
%YAML 1.2
|
||||
---
|
||||
$id: http://devicetree.org/schemas/sound/socionext,uniphier-aio.yaml#
|
||||
$schema: http://devicetree.org/meta-schemas/core.yaml#
|
||||
|
||||
title: UniPhier AIO audio system
|
||||
|
||||
maintainers:
|
||||
- <alsa-devel@alsa-project.org>
|
||||
|
||||
properties:
|
||||
compatible:
|
||||
enum:
|
||||
- socionext,uniphier-ld11-aio
|
||||
- socionext,uniphier-ld20-aio
|
||||
- socionext,uniphier-pxs2-aio
|
||||
|
||||
reg:
|
||||
maxItems: 1
|
||||
|
||||
interrupts:
|
||||
maxItems: 1
|
||||
|
||||
clock-names:
|
||||
const: aio
|
||||
|
||||
clocks:
|
||||
maxItems: 1
|
||||
|
||||
reset-names:
|
||||
const: aio
|
||||
|
||||
resets:
|
||||
maxItems: 1
|
||||
|
||||
socionext,syscon:
|
||||
description: |
|
||||
Specifies a phandle to soc-glue, which is used for changing mode of S/PDIF
|
||||
signal pin to output from Hi-Z. This property is optional if you use I2S
|
||||
signal pins only.
|
||||
$ref: "/schemas/types.yaml#/definitions/phandle"
|
||||
|
||||
"#sound-dai-cells":
|
||||
const: 1
|
||||
|
||||
patternProperties:
|
||||
"^port@[0-9]$":
|
||||
type: object
|
||||
properties:
|
||||
endpoint: true
|
||||
required:
|
||||
- endpoint
|
||||
|
||||
additionalProperties: false
|
||||
|
||||
required:
|
||||
- compatible
|
||||
- reg
|
||||
- interrupts
|
||||
- clock-names
|
||||
- clocks
|
||||
- reset-names
|
||||
- resets
|
||||
- "#sound-dai-cells"
|
||||
|
||||
examples:
|
||||
- |
|
||||
audio@56000000 {
|
||||
compatible = "socionext,uniphier-ld20-aio";
|
||||
reg = <0x56000000 0x80000>;
|
||||
interrupts = <0 144 4>;
|
||||
pinctrl-names = "default";
|
||||
pinctrl-0 = <&pinctrl_aout>;
|
||||
clock-names = "aio";
|
||||
clocks = <&sys_clk 40>;
|
||||
reset-names = "aio";
|
||||
resets = <&sys_rst 40>;
|
||||
#sound-dai-cells = <1>;
|
||||
socionext,syscon = <&soc_glue>;
|
||||
};
|
||||
|
|
@ -0,0 +1,70 @@
|
|||
# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause
|
||||
%YAML 1.2
|
||||
---
|
||||
$id: http://devicetree.org/schemas/sound/socionext,uniphier-evea.yaml#
|
||||
$schema: http://devicetree.org/meta-schemas/core.yaml#
|
||||
|
||||
title: UniPhier EVEA SoC-internal sound codec
|
||||
|
||||
maintainers:
|
||||
- <alsa-devel@alsa-project.org>
|
||||
|
||||
properties:
|
||||
compatible:
|
||||
const: socionext,uniphier-evea
|
||||
|
||||
reg:
|
||||
maxItems: 1
|
||||
|
||||
clock-names:
|
||||
items:
|
||||
- const: evea
|
||||
- const: exiv
|
||||
|
||||
clocks:
|
||||
minItems: 2
|
||||
maxItems: 2
|
||||
|
||||
reset-names:
|
||||
items:
|
||||
- const: evea
|
||||
- const: exiv
|
||||
- const: adamv
|
||||
|
||||
resets:
|
||||
minItems: 3
|
||||
maxItems: 3
|
||||
|
||||
"#sound-dai-cells":
|
||||
const: 1
|
||||
|
||||
patternProperties:
|
||||
"^port@[0-9]$":
|
||||
type: object
|
||||
properties:
|
||||
endpoint: true
|
||||
required:
|
||||
- endpoint
|
||||
|
||||
additionalProperties: false
|
||||
|
||||
required:
|
||||
- compatible
|
||||
- reg
|
||||
- clock-names
|
||||
- clocks
|
||||
- reset-names
|
||||
- resets
|
||||
- "#sound-dai-cells"
|
||||
|
||||
examples:
|
||||
- |
|
||||
codec@57900000 {
|
||||
compatible = "socionext,uniphier-evea";
|
||||
reg = <0x57900000 0x1000>;
|
||||
clock-names = "evea", "exiv";
|
||||
clocks = <&sys_clk 41>, <&sys_clk 42>;
|
||||
reset-names = "evea", "exiv", "adamv";
|
||||
resets = <&sys_rst 41>, <&sys_rst 42>, <&adamv_rst 0>;
|
||||
#sound-dai-cells = <1>;
|
||||
};
|
||||
|
|
@ -33,4 +33,4 @@ tas2552: tas2552@41 {
|
|||
};
|
||||
|
||||
For more product information please see the link below:
|
||||
http://www.ti.com/product/TAS2552
|
||||
https://www.ti.com/product/TAS2552
|
||||
|
|
|
|||
|
|
@ -11,12 +11,14 @@ Required properties:
|
|||
- compatible: - Should contain "ti,tas2562", "ti,tas2563".
|
||||
- reg: - The i2c address. Should be 0x4c, 0x4d, 0x4e or 0x4f.
|
||||
- ti,imon-slot-no:- TDM TX current sense time slot.
|
||||
- ti,vmon-slot-no:- TDM TX voltage sense time slot. This slot must always be
|
||||
greater then ti,imon-slot-no.
|
||||
|
||||
Optional properties:
|
||||
- interrupt-parent: phandle to the interrupt controller which provides
|
||||
the interrupt.
|
||||
- interrupts: (GPIO) interrupt to which the chip is connected.
|
||||
- shut-down: GPIO used to control the state of the device.
|
||||
- shut-down-gpio: GPIO used to control the state of the device.
|
||||
|
||||
Examples:
|
||||
tas2562@4c {
|
||||
|
|
@ -28,7 +30,8 @@ tas2562@4c {
|
|||
interrupt-parent = <&gpio1>;
|
||||
interrupts = <14>;
|
||||
|
||||
shut-down = <&gpio1 15 0>;
|
||||
shut-down-gpio = <&gpio1 15 0>;
|
||||
ti,imon-slot-no = <0>;
|
||||
ti,vmon-slot-no = <1>;
|
||||
};
|
||||
|
||||
|
|
|
|||
|
|
@ -0,0 +1,69 @@
|
|||
# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
|
||||
# Copyright (C) 2019 Texas Instruments Incorporated
|
||||
%YAML 1.2
|
||||
---
|
||||
$id: "http://devicetree.org/schemas/sound/tas2562.yaml#"
|
||||
$schema: "http://devicetree.org/meta-schemas/core.yaml#"
|
||||
|
||||
title: Texas Instruments TAS2562 Smart PA
|
||||
|
||||
maintainers:
|
||||
- Dan Murphy <dmurphy@ti.com>
|
||||
|
||||
description: |
|
||||
The TAS2562 is a mono, digital input Class-D audio amplifier optimized for
|
||||
efficiently driving high peak power into small loudspeakers.
|
||||
Integrated speaker voltage and current sense provides for
|
||||
real time monitoring of loudspeaker behavior.
|
||||
|
||||
properties:
|
||||
compatible:
|
||||
enum:
|
||||
- ti,tas2562
|
||||
- ti,tas2563
|
||||
|
||||
reg:
|
||||
maxItems: 1
|
||||
description: |
|
||||
I2C address of the device can be one of these 0x4c, 0x4d, 0x4e or 0x4f
|
||||
|
||||
shut-down-gpios:
|
||||
description: GPIO used to control the state of the device.
|
||||
deprecated: true
|
||||
|
||||
shutdown-gpios:
|
||||
description: GPIO used to control the state of the device.
|
||||
|
||||
interrupts:
|
||||
maxItems: 1
|
||||
|
||||
ti,imon-slot-no:
|
||||
$ref: /schemas/types.yaml#/definitions/uint32
|
||||
description: TDM TX current sense time slot.
|
||||
|
||||
'#sound-dai-cells':
|
||||
const: 1
|
||||
|
||||
required:
|
||||
- compatible
|
||||
- reg
|
||||
|
||||
additionalProperties: false
|
||||
|
||||
examples:
|
||||
- |
|
||||
#include <dt-bindings/gpio/gpio.h>
|
||||
i2c0 {
|
||||
#address-cells = <1>;
|
||||
#size-cells = <0>;
|
||||
codec: codec@4c {
|
||||
compatible = "ti,tas2562";
|
||||
reg = <0x4c>;
|
||||
#sound-dai-cells = <1>;
|
||||
interrupt-parent = <&gpio1>;
|
||||
interrupts = <14>;
|
||||
shutdown-gpios = <&gpio1 15 0>;
|
||||
ti,imon-slot-no = <0>;
|
||||
};
|
||||
};
|
||||
|
||||
|
|
@ -1,37 +0,0 @@
|
|||
Texas Instruments TAS2770 Smart PA
|
||||
|
||||
The TAS2770 is a mono, digital input Class-D audio amplifier optimized for
|
||||
efficiently driving high peak power into small loudspeakers.
|
||||
Integrated speaker voltage and current sense provides for
|
||||
real time monitoring of loudspeaker behavior.
|
||||
|
||||
Required properties:
|
||||
|
||||
- compatible: - Should contain "ti,tas2770".
|
||||
- reg: - The i2c address. Should contain <0x4c>, <0x4d>,<0x4e>, or <0x4f>.
|
||||
- #address-cells - Should be <1>.
|
||||
- #size-cells - Should be <0>.
|
||||
- ti,asi-format: - Sets TDM RX capture edge. 0->Rising; 1->Falling.
|
||||
- ti,imon-slot-no:- TDM TX current sense time slot.
|
||||
- ti,vmon-slot-no:- TDM TX voltage sense time slot.
|
||||
|
||||
Optional properties:
|
||||
|
||||
- interrupt-parent: the phandle to the interrupt controller which provides
|
||||
the interrupt.
|
||||
- interrupts: interrupt specification for data-ready.
|
||||
|
||||
Examples:
|
||||
|
||||
tas2770@4c {
|
||||
compatible = "ti,tas2770";
|
||||
reg = <0x4c>;
|
||||
#address-cells = <1>;
|
||||
#size-cells = <0>;
|
||||
interrupt-parent = <&msm_gpio>;
|
||||
interrupts = <97 0>;
|
||||
ti,asi-format = <0>;
|
||||
ti,imon-slot-no = <0>;
|
||||
ti,vmon-slot-no = <2>;
|
||||
};
|
||||
|
||||
|
|
@ -0,0 +1,76 @@
|
|||
# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
|
||||
# Copyright (C) 2019-20 Texas Instruments Incorporated
|
||||
%YAML 1.2
|
||||
---
|
||||
$id: "http://devicetree.org/schemas/sound/tas2770.yaml#"
|
||||
$schema: "http://devicetree.org/meta-schemas/core.yaml#"
|
||||
|
||||
title: Texas Instruments TAS2770 Smart PA
|
||||
|
||||
maintainers:
|
||||
- Shi Fu <shifu0704@thundersoft.com>
|
||||
|
||||
description: |
|
||||
The TAS2770 is a mono, digital input Class-D audio amplifier optimized for
|
||||
efficiently driving high peak power into small loudspeakers.
|
||||
Integrated speaker voltage and current sense provides for
|
||||
real time monitoring of loudspeaker behavior.
|
||||
|
||||
properties:
|
||||
compatible:
|
||||
enum:
|
||||
- ti,tas2770
|
||||
|
||||
reg:
|
||||
maxItems: 1
|
||||
description: |
|
||||
I2C address of the device can be one of these 0x4c, 0x4d, 0x4e or 0x4f
|
||||
|
||||
reset-gpio:
|
||||
description: GPIO used to reset the device.
|
||||
|
||||
interrupts:
|
||||
maxItems: 1
|
||||
|
||||
ti,imon-slot-no:
|
||||
$ref: /schemas/types.yaml#/definitions/uint32
|
||||
description: TDM TX current sense time slot.
|
||||
|
||||
ti,vmon-slot-no:
|
||||
$ref: /schemas/types.yaml#/definitions/uint32
|
||||
description: TDM TX voltage sense time slot.
|
||||
|
||||
ti,asi-format:
|
||||
$ref: /schemas/types.yaml#/definitions/uint32
|
||||
description: Sets TDM RX capture edge.
|
||||
enum:
|
||||
- 0 # Rising edge
|
||||
- 1 # Falling edge
|
||||
|
||||
'#sound-dai-cells':
|
||||
const: 1
|
||||
|
||||
required:
|
||||
- compatible
|
||||
- reg
|
||||
|
||||
additionalProperties: false
|
||||
|
||||
examples:
|
||||
- |
|
||||
#include <dt-bindings/gpio/gpio.h>
|
||||
i2c0 {
|
||||
#address-cells = <1>;
|
||||
#size-cells = <0>;
|
||||
codec: codec@4c {
|
||||
compatible = "ti,tas2770";
|
||||
reg = <0x4c>;
|
||||
#sound-dai-cells = <1>;
|
||||
interrupt-parent = <&gpio1>;
|
||||
interrupts = <14>;
|
||||
reset-gpio = <&gpio1 15 0>;
|
||||
ti,imon-slot-no = <0>;
|
||||
ti,vmon-slot-no = <2>;
|
||||
};
|
||||
};
|
||||
|
||||
|
|
@ -4,9 +4,9 @@ The TAS5720 serial control bus communicates through the I2C protocol only. The
|
|||
serial bus is also used for periodic codec fault checking/reporting during
|
||||
audio playback. For more product information please see the links below:
|
||||
|
||||
http://www.ti.com/product/TAS5720L
|
||||
http://www.ti.com/product/TAS5720M
|
||||
http://www.ti.com/product/TAS5722L
|
||||
https://www.ti.com/product/TAS5720L
|
||||
https://www.ti.com/product/TAS5720M
|
||||
https://www.ti.com/product/TAS5722L
|
||||
|
||||
Required properties:
|
||||
|
||||
|
|
|
|||
|
|
@ -0,0 +1,95 @@
|
|||
# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
|
||||
%YAML 1.2
|
||||
---
|
||||
$id: http://devicetree.org/schemas/sound/ti,j721e-cpb-audio.yaml#
|
||||
$schema: http://devicetree.org/meta-schemas/core.yaml#
|
||||
|
||||
title: Texas Instruments J721e Common Processor Board Audio Support
|
||||
|
||||
maintainers:
|
||||
- Peter Ujfalusi <peter.ujfalusi@ti.com>
|
||||
|
||||
description: |
|
||||
The audio support on the board is using pcm3168a codec connected to McASP10
|
||||
serializers in parallel setup.
|
||||
The pcm3168a SCKI clock is sourced from j721e AUDIO_REFCLK2 pin.
|
||||
In order to support 48KHz and 44.1KHz family of sampling rates the parent
|
||||
clock for AUDIO_REFCLK2 needs to be changed between PLL4 (for 48KHz) and
|
||||
PLL15 (for 44.1KHz). The same PLLs are used for McASP10's AUXCLK clock via
|
||||
different HSDIVIDER.
|
||||
|
||||
Clocking setup for 48KHz family:
|
||||
PLL4 ---> PLL4_HSDIV0 ---> MCASP10_AUXCLK ---> McASP10.auxclk
|
||||
|-> PLL4_HSDIV2 ---> AUDIO_REFCLK2 ---> pcm3168a.SCKI
|
||||
|
||||
Clocking setup for 44.1KHz family:
|
||||
PLL15 ---> PLL15_HSDIV0 ---> MCASP10_AUXCLK ---> McASP10.auxclk
|
||||
|-> PLL15_HSDIV2 ---> AUDIO_REFCLK2 ---> pcm3168a.SCKI
|
||||
|
||||
properties:
|
||||
compatible:
|
||||
items:
|
||||
- const: ti,j721e-cpb-audio
|
||||
|
||||
model:
|
||||
$ref: /schemas/types.yaml#/definitions/string
|
||||
description: User specified audio sound card name
|
||||
|
||||
ti,cpb-mcasp:
|
||||
description: phandle to McASP used on CPB
|
||||
allOf:
|
||||
- $ref: /schemas/types.yaml#/definitions/phandle
|
||||
|
||||
ti,cpb-codec:
|
||||
description: phandle to the pcm3168a codec used on the CPB
|
||||
allOf:
|
||||
- $ref: /schemas/types.yaml#/definitions/phandle
|
||||
|
||||
clocks:
|
||||
items:
|
||||
- description: AUXCLK clock for McASP used by CPB audio
|
||||
- description: Parent for CPB_McASP auxclk (for 48KHz)
|
||||
- description: Parent for CPB_McASP auxclk (for 44.1KHz)
|
||||
- description: SCKI clock for the pcm3168a codec on CPB
|
||||
- description: Parent for CPB_SCKI clock (for 48KHz)
|
||||
- description: Parent for CPB_SCKI clock (for 44.1KHz)
|
||||
|
||||
clock-names:
|
||||
items:
|
||||
- const: cpb-mcasp-auxclk
|
||||
- const: cpb-mcasp-auxclk-48000
|
||||
- const: cpb-mcasp-auxclk-44100
|
||||
- const: cpb-codec-scki
|
||||
- const: cpb-codec-scki-48000
|
||||
- const: cpb-codec-scki-44100
|
||||
|
||||
required:
|
||||
- compatible
|
||||
- model
|
||||
- ti,cpb-mcasp
|
||||
- ti,cpb-codec
|
||||
- clocks
|
||||
- clock-names
|
||||
|
||||
additionalProperties: false
|
||||
|
||||
examples:
|
||||
- |+
|
||||
sound {
|
||||
compatible = "ti,j721e-cpb-audio";
|
||||
model = "j721e-cpb";
|
||||
|
||||
status = "okay";
|
||||
|
||||
ti,cpb-mcasp = <&mcasp10>;
|
||||
ti,cpb-codec = <&pcm3168a_1>;
|
||||
|
||||
clocks = <&k3_clks 184 1>,
|
||||
<&k3_clks 184 2>, <&k3_clks 184 4>,
|
||||
<&k3_clks 157 371>,
|
||||
<&k3_clks 157 400>, <&k3_clks 157 401>;
|
||||
clock-names = "cpb-mcasp-auxclk",
|
||||
"cpb-mcasp-auxclk-48000", "cpb-mcasp-auxclk-44100",
|
||||
"cpb-codec-scki",
|
||||
"cpb-codec-scki-48000", "cpb-codec-scki-44100";
|
||||
};
|
||||
|
|
@ -0,0 +1,150 @@
|
|||
# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
|
||||
%YAML 1.2
|
||||
---
|
||||
$id: http://devicetree.org/schemas/sound/ti,j721e-cpb-ivi-audio.yaml#
|
||||
$schema: http://devicetree.org/meta-schemas/core.yaml#
|
||||
|
||||
title: Texas Instruments J721e Common Processor Board Audio Support
|
||||
|
||||
maintainers:
|
||||
- Peter Ujfalusi <peter.ujfalusi@ti.com>
|
||||
|
||||
description: |
|
||||
The Infotainment board plugs into the Common Processor Board, the support of the
|
||||
extension board is extending the CPB audio support, decribed in:
|
||||
sound/ti,j721e-cpb-audio.txt
|
||||
|
||||
The audio support on the Infotainment Expansion Board consists of McASP0
|
||||
connected to two pcm3168a codecs with dedicated set of serializers to each.
|
||||
The SCKI for pcm3168a is sourced from j721e AUDIO_REFCLK0 pin.
|
||||
|
||||
In order to support 48KHz and 44.1KHz family of sampling rates the parent clock
|
||||
for AUDIO_REFCLK0 needs to be changed between PLL4 (for 48KHz) and PLL15 (for
|
||||
44.1KHz). The same PLLs are used for McASP0's AUXCLK clock via different
|
||||
HSDIVIDER.
|
||||
|
||||
Note: the same PLL4 and PLL15 is used by the audio support on the CPB!
|
||||
|
||||
Clocking setup for 48KHz family:
|
||||
PLL4 ---> PLL4_HSDIV0 ---> MCASP10_AUXCLK ---> McASP10.auxclk
|
||||
| |-> MCASP0_AUXCLK ---> McASP0.auxclk
|
||||
|
|
||||
|-> PLL4_HSDIV2 ---> AUDIO_REFCLK2 ---> pcm3168a.SCKI
|
||||
|-> AUDIO_REFCLK0 ---> pcm3168a_a/b.SCKI
|
||||
|
||||
Clocking setup for 44.1KHz family:
|
||||
PLL15 ---> PLL15_HSDIV0 ---> MCASP10_AUXCLK ---> McASP10.auxclk
|
||||
| |-> MCASP0_AUXCLK ---> McASP0.auxclk
|
||||
|
|
||||
|-> PLL15_HSDIV2 ---> AUDIO_REFCLK2 ---> pcm3168a.SCKI
|
||||
|-> AUDIO_REFCLK0 ---> pcm3168a_a/b.SCKI
|
||||
|
||||
properties:
|
||||
compatible:
|
||||
items:
|
||||
- const: ti,j721e-cpb-ivi-audio
|
||||
|
||||
model:
|
||||
$ref: /schemas/types.yaml#/definitions/string
|
||||
description: User specified audio sound card name
|
||||
|
||||
ti,cpb-mcasp:
|
||||
description: phandle to McASP used on CPB
|
||||
allOf:
|
||||
- $ref: /schemas/types.yaml#/definitions/phandle
|
||||
|
||||
ti,cpb-codec:
|
||||
description: phandle to the pcm3168a codec used on the CPB
|
||||
allOf:
|
||||
- $ref: /schemas/types.yaml#/definitions/phandle
|
||||
|
||||
ti,ivi-mcasp:
|
||||
description: phandle to McASP used on IVI
|
||||
allOf:
|
||||
- $ref: /schemas/types.yaml#/definitions/phandle
|
||||
|
||||
ti,ivi-codec-a:
|
||||
description: phandle to the pcm3168a-A codec on the expansion board
|
||||
allOf:
|
||||
- $ref: /schemas/types.yaml#/definitions/phandle
|
||||
|
||||
ti,ivi-codec-b:
|
||||
description: phandle to the pcm3168a-B codec on the expansion board
|
||||
allOf:
|
||||
- $ref: /schemas/types.yaml#/definitions/phandle
|
||||
|
||||
clocks:
|
||||
items:
|
||||
- description: AUXCLK clock for McASP used by CPB audio
|
||||
- description: Parent for CPB_McASP auxclk (for 48KHz)
|
||||
- description: Parent for CPB_McASP auxclk (for 44.1KHz)
|
||||
- description: SCKI clock for the pcm3168a codec on CPB
|
||||
- description: Parent for CPB_SCKI clock (for 48KHz)
|
||||
- description: Parent for CPB_SCKI clock (for 44.1KHz)
|
||||
- description: AUXCLK clock for McASP used by IVI audio
|
||||
- description: Parent for IVI_McASP auxclk (for 48KHz)
|
||||
- description: Parent for IVI_McASP auxclk (for 44.1KHz)
|
||||
- description: SCKI clock for the pcm3168a codec on IVI
|
||||
- description: Parent for IVI_SCKI clock (for 48KHz)
|
||||
- description: Parent for IVI_SCKI clock (for 44.1KHz)
|
||||
|
||||
clock-names:
|
||||
items:
|
||||
- const: cpb-mcasp-auxclk
|
||||
- const: cpb-mcasp-auxclk-48000
|
||||
- const: cpb-mcasp-auxclk-44100
|
||||
- const: cpb-codec-scki
|
||||
- const: cpb-codec-scki-48000
|
||||
- const: cpb-codec-scki-44100
|
||||
- const: ivi-mcasp-auxclk
|
||||
- const: ivi-mcasp-auxclk-48000
|
||||
- const: ivi-mcasp-auxclk-44100
|
||||
- const: ivi-codec-scki
|
||||
- const: ivi-codec-scki-48000
|
||||
- const: ivi-codec-scki-44100
|
||||
|
||||
required:
|
||||
- compatible
|
||||
- model
|
||||
- ti,cpb-mcasp
|
||||
- ti,cpb-codec
|
||||
- ti,ivi-mcasp
|
||||
- ti,ivi-codec-a
|
||||
- ti,ivi-codec-b
|
||||
- clocks
|
||||
- clock-names
|
||||
|
||||
additionalProperties: false
|
||||
|
||||
examples:
|
||||
- |+
|
||||
sound {
|
||||
compatible = "ti,j721e-cpb-ivi-audio";
|
||||
model = "j721e-cpb-ivi";
|
||||
|
||||
status = "okay";
|
||||
|
||||
ti,cpb-mcasp = <&mcasp10>;
|
||||
ti,cpb-codec = <&pcm3168a_1>;
|
||||
|
||||
ti,ivi-mcasp = <&mcasp0>;
|
||||
ti,ivi-codec-a = <&pcm3168a_a>;
|
||||
ti,ivi-codec-b = <&pcm3168a_b>;
|
||||
|
||||
clocks = <&k3_clks 184 1>,
|
||||
<&k3_clks 184 2>, <&k3_clks 184 4>,
|
||||
<&k3_clks 157 371>,
|
||||
<&k3_clks 157 400>, <&k3_clks 157 401>,
|
||||
<&k3_clks 174 1>,
|
||||
<&k3_clks 174 2>, <&k3_clks 174 4>,
|
||||
<&k3_clks 157 301>,
|
||||
<&k3_clks 157 330>, <&k3_clks 157 331>;
|
||||
clock-names = "cpb-mcasp-auxclk",
|
||||
"cpb-mcasp-auxclk-48000", "cpb-mcasp-auxclk-44100",
|
||||
"cpb-codec-scki",
|
||||
"cpb-codec-scki-48000", "cpb-codec-scki-44100",
|
||||
"ivi-mcasp-auxclk",
|
||||
"ivi-mcasp-auxclk-48000", "ivi-mcasp-auxclk-44100",
|
||||
"ivi-codec-scki",
|
||||
"ivi-codec-scki-48000", "ivi-codec-scki-44100";
|
||||
};
|
||||
|
|
@ -19,4 +19,4 @@ tas6424: tas6424@6a {
|
|||
};
|
||||
|
||||
For more product information please see the link below:
|
||||
http://www.ti.com/product/TAS6424-Q1
|
||||
https://www.ti.com/product/TAS6424-Q1
|
||||
|
|
|
|||
|
|
@ -18,9 +18,9 @@ description: |
|
|||
microphone bias or supply voltage generation.
|
||||
|
||||
Specifications can be found at:
|
||||
http://www.ti.com/lit/ds/symlink/tlv320adc3140.pdf
|
||||
http://www.ti.com/lit/ds/symlink/tlv320adc5140.pdf
|
||||
http://www.ti.com/lit/ds/symlink/tlv320adc6140.pdf
|
||||
https://www.ti.com/lit/ds/symlink/tlv320adc3140.pdf
|
||||
https://www.ti.com/lit/ds/symlink/tlv320adc5140.pdf
|
||||
https://www.ti.com/lit/ds/symlink/tlv320adc6140.pdf
|
||||
|
||||
properties:
|
||||
compatible:
|
||||
|
|
@ -108,6 +108,32 @@ properties:
|
|||
maximum: 7
|
||||
default: [0, 0, 0, 0]
|
||||
|
||||
patternProperties:
|
||||
'^ti,gpo-config-[1-4]$':
|
||||
$ref: /schemas/types.yaml#/definitions/uint32-array
|
||||
description: |
|
||||
Defines the configuration and output driver for the general purpose
|
||||
output pins (GPO). These values are pairs, the first value is for the
|
||||
configuration type and the second value is for the output drive type.
|
||||
The array is defined as <GPO_CFG GPO_DRV>
|
||||
|
||||
GPO output configuration can be one of the following:
|
||||
|
||||
0 - (default) disabled
|
||||
1 - GPOX is configured as a general-purpose output (GPO)
|
||||
2 - GPOX is configured as a device interrupt output (IRQ)
|
||||
3 - GPOX is configured as a secondary ASI output (SDOUT2)
|
||||
4 - GPOX is configured as a PDM clock output (PDMCLK)
|
||||
|
||||
GPO output drive configuration for the GPO pins can be one of the following:
|
||||
|
||||
0d - (default) Hi-Z output
|
||||
1d - Drive active low and active high
|
||||
2d - Drive active low and weak high
|
||||
3d - Drive active low and Hi-Z
|
||||
4d - Drive weak low and active high
|
||||
5d - Drive Hi-Z and active high
|
||||
|
||||
required:
|
||||
- compatible
|
||||
- reg
|
||||
|
|
@ -124,6 +150,8 @@ examples:
|
|||
ti,mic-bias-source = <6>;
|
||||
ti,pdm-edge-select = <0 1 0 1>;
|
||||
ti,gpi-config = <4 5 6 7>;
|
||||
ti,gpo-config-1 = <0 0>;
|
||||
ti,gpo-config-2 = <0 0>;
|
||||
reset-gpios = <&gpio0 14 GPIO_ACTIVE_HIGH>;
|
||||
};
|
||||
};
|
||||
|
|
|
|||
|
|
@ -1,45 +0,0 @@
|
|||
Socionext UniPhier SoC audio driver
|
||||
|
||||
The Socionext UniPhier audio subsystem consists of I2S and S/PDIF blocks in
|
||||
the same register space.
|
||||
|
||||
Required properties:
|
||||
- compatible : should be one of the following:
|
||||
"socionext,uniphier-ld11-aio"
|
||||
"socionext,uniphier-ld20-aio"
|
||||
"socionext,uniphier-pxs2-aio"
|
||||
- reg : offset and length of the register set for the device.
|
||||
- interrupts : should contain I2S or S/PDIF interrupt.
|
||||
- pinctrl-names : should be "default".
|
||||
- pinctrl-0 : defined I2S signal pins for an external codec chip.
|
||||
- clock-names : should include following entries:
|
||||
"aio"
|
||||
- clocks : a list of phandle, should contain an entry for each
|
||||
entry in clock-names.
|
||||
- reset-names : should include following entries:
|
||||
"aio"
|
||||
- resets : a list of phandle, should contain an entry for each
|
||||
entry in reset-names.
|
||||
- #sound-dai-cells: should be 1.
|
||||
|
||||
Optional properties:
|
||||
- socionext,syscon: a phandle, should contain soc-glue.
|
||||
The soc-glue is used for changing mode of S/PDIF signal pin
|
||||
to Output from Hi-Z. This property is optional if you use
|
||||
I2S signal pins only.
|
||||
|
||||
Example:
|
||||
audio {
|
||||
compatible = "socionext,uniphier-ld20-aio";
|
||||
reg = <0x56000000 0x80000>;
|
||||
interrupts = <0 144 4>;
|
||||
pinctrl-names = "default";
|
||||
pinctrl-0 = <&pinctrl_aout>;
|
||||
clock-names = "aio";
|
||||
clocks = <&sys_clk 40>;
|
||||
reset-names = "aio";
|
||||
resets = <&sys_rst 40>;
|
||||
#sound-dai-cells = <1>;
|
||||
|
||||
socionext,syscon = <&sg>;
|
||||
};
|
||||
|
|
@ -1,26 +0,0 @@
|
|||
Socionext EVEA - UniPhier SoC internal codec driver
|
||||
|
||||
Required properties:
|
||||
- compatible : should be "socionext,uniphier-evea".
|
||||
- reg : offset and length of the register set for the device.
|
||||
- clock-names : should include following entries:
|
||||
"evea", "exiv"
|
||||
- clocks : a list of phandle, should contain an entry for each
|
||||
entries in clock-names.
|
||||
- reset-names : should include following entries:
|
||||
"evea", "exiv", "adamv"
|
||||
- resets : a list of phandle, should contain reset entries of
|
||||
reset-names.
|
||||
- #sound-dai-cells: should be 1.
|
||||
|
||||
Example:
|
||||
|
||||
codec {
|
||||
compatible = "socionext,uniphier-evea";
|
||||
reg = <0x57900000 0x1000>;
|
||||
clock-names = "evea", "exiv";
|
||||
clocks = <&sys_clk 41>, <&sys_clk 42>;
|
||||
reset-names = "evea", "exiv", "adamv";
|
||||
resets = <&sys_rst 41>, <&sys_rst 42>, <&adamv_rst 0>;
|
||||
#sound-dai-cells = <1>;
|
||||
};
|
||||
|
|
@ -21,6 +21,17 @@ Optional properties:
|
|||
enabled and disabled together with HP_L and HP_R pins in response to jack
|
||||
detect events.
|
||||
|
||||
- wlf,hp-cfg: A list of headphone jack detect configuration register values.
|
||||
The list must be 3 entries long.
|
||||
hp-cfg[0]: HPSEL[1:0] of R48 (Additional Control 4).
|
||||
hp-cfg[1]: {HPSWEN:HPSWPOL} of R24 (Additional Control 2).
|
||||
hp-cfg[2]: {TOCLKSEL:TOEN} of R23 (Additional Control 1).
|
||||
|
||||
- wlf,gpio-cfg: A list of GPIO configuration register values.
|
||||
The list must be 2 entries long.
|
||||
gpio-cfg[0]: ALRCGPIO of R9 (Audio interface)
|
||||
gpio-cfg[1]: {GPIOPOL:GPIOSEL[2:0]} of R48 (Additional Control 4).
|
||||
|
||||
Example:
|
||||
|
||||
wm8960: codec@1a {
|
||||
|
|
|
|||
|
|
@ -68,6 +68,29 @@ Optional properties:
|
|||
- wlf,csnaddr-pd : If present enable the internal pull-down resistor on
|
||||
the CS/ADDR pin.
|
||||
|
||||
Pins on the device (for linking into audio routes):
|
||||
|
||||
* IN1LN
|
||||
* IN1LP
|
||||
* IN2LN
|
||||
* IN2LP:VXRN
|
||||
* IN1RN
|
||||
* IN1RP
|
||||
* IN2RN
|
||||
* IN2RP:VXRP
|
||||
* SPKOUTLP
|
||||
* SPKOUTLN
|
||||
* SPKOUTRP
|
||||
* SPKOUTRN
|
||||
* HPOUT1L
|
||||
* HPOUT1R
|
||||
* HPOUT2P
|
||||
* HPOUT2N
|
||||
* LINEOUT1P
|
||||
* LINEOUT1N
|
||||
* LINEOUT2P
|
||||
* LINEOUT2N
|
||||
|
||||
Example:
|
||||
|
||||
wm8994: codec@1a {
|
||||
|
|
|
|||
|
|
@ -80,8 +80,6 @@ properties:
|
|||
- fsl,mpl3115
|
||||
# MPR121: Proximity Capacitive Touch Sensor Controller
|
||||
- fsl,mpr121
|
||||
# SGTL5000: Ultra Low-Power Audio Codec
|
||||
- fsl,sgtl5000
|
||||
# G751: Digital Temperature Sensor and Thermal Watchdog with Two-Wire Interface
|
||||
- gmt,g751
|
||||
# Infineon IR38064 Voltage Regulator
|
||||
|
|
|
|||
|
|
@ -20,7 +20,7 @@ patternProperties:
|
|||
"^(keypad|m25p|max8952|max8997|max8998|mpmc),.*": true
|
||||
"^(pinctrl-single|#pinctrl-single|PowerPC),.*": true
|
||||
"^(pl022|pxa-mmc|rcar_sound|rotary-encoder|s5m8767|sdhci),.*": true
|
||||
"^(simple-audio-card|simple-graph-card|st-plgpio|st-spics|ts),.*": true
|
||||
"^(simple-audio-card|st-plgpio|st-spics|ts),.*": true
|
||||
|
||||
# Keep list in alphabetical order.
|
||||
"^70mai,.*":
|
||||
|
|
|
|||
|
|
@ -309,7 +309,7 @@ pcifix
|
|||
This module supports all ADB PCM channels, ac97 mixer, SPDIF, hardware
|
||||
EQ, mpu401, gameport. A3D and wavetable support are still in development.
|
||||
Development and reverse engineering work is being coordinated at
|
||||
http://savannah.nongnu.org/projects/openvortex/
|
||||
https://savannah.nongnu.org/projects/openvortex/
|
||||
SPDIF output has a copy of the AC97 codec output, unless you use the
|
||||
``spdif`` pcm device, which allows raw data passthru.
|
||||
The hardware EQ hardware and SPDIF is only present in the Vortex2 and
|
||||
|
|
@ -1575,7 +1575,7 @@ See Documentation/sound/cards/multisound.sh for important information
|
|||
about this driver. Note that it has been discontinued, but the
|
||||
Voyetra Turtle Beach knowledge base entry for it is still available
|
||||
at
|
||||
http://www.turtlebeach.com
|
||||
https://www.turtlebeach.com
|
||||
|
||||
Module snd-msnd-pinnacle
|
||||
------------------------
|
||||
|
|
@ -2703,4 +2703,4 @@ Kernel Bugzilla
|
|||
ALSA Developers ML
|
||||
mailto:alsa-devel@alsa-project.org
|
||||
alsa-info.sh script
|
||||
http://www.alsa-project.org/alsa-info.sh
|
||||
https://www.alsa-project.org/alsa-info.sh
|
||||
|
|
|
|||
|
|
@ -331,7 +331,7 @@ WO 9901953 (A1)
|
|||
Execution and Audio Data Sequencing (Jan. 14, 1999)
|
||||
|
||||
|
||||
US Patents (http://www.uspto.gov/)
|
||||
US Patents (https://www.uspto.gov/)
|
||||
----------------------------------
|
||||
|
||||
US 5925841
|
||||
|
|
|
|||
|
|
@ -336,7 +336,7 @@ WO 9901953 (A1)
|
|||
Execution and Audio Data Sequencing (Jan. 14, 1999)
|
||||
|
||||
|
||||
US Patents (http://www.uspto.gov/)
|
||||
US Patents (https://www.uspto.gov/)
|
||||
----------------------------------
|
||||
|
||||
US 5925841
|
||||
|
|
|
|||
|
|
@ -151,6 +151,57 @@ Modifications include:
|
|||
- Addition of encoding options when required (derived from OpenMAX IL)
|
||||
- Addition of rateControlSupported (missing in OpenMAX AL)
|
||||
|
||||
State Machine
|
||||
=============
|
||||
|
||||
The compressed audio stream state machine is described below ::
|
||||
|
||||
+----------+
|
||||
| |
|
||||
| OPEN |
|
||||
| |
|
||||
+----------+
|
||||
|
|
||||
|
|
||||
| compr_set_params()
|
||||
|
|
||||
v
|
||||
compr_free() +----------+
|
||||
+------------------------------------| |
|
||||
| | SETUP |
|
||||
| +-------------------------| |<-------------------------+
|
||||
| | compr_write() +----------+ |
|
||||
| | ^ |
|
||||
| | | compr_drain_notify() |
|
||||
| | | or |
|
||||
| | | compr_stop() |
|
||||
| | | |
|
||||
| | +----------+ |
|
||||
| | | | |
|
||||
| | | DRAIN | |
|
||||
| | | | |
|
||||
| | +----------+ |
|
||||
| | ^ |
|
||||
| | | |
|
||||
| | | compr_drain() |
|
||||
| | | |
|
||||
| v | |
|
||||
| +----------+ +----------+ |
|
||||
| | | compr_start() | | compr_stop() |
|
||||
| | PREPARE |------------------->| RUNNING |--------------------------+
|
||||
| | | | | |
|
||||
| +----------+ +----------+ |
|
||||
| | | ^ |
|
||||
| |compr_free() | | |
|
||||
| | compr_pause() | | compr_resume() |
|
||||
| | | | |
|
||||
| v v | |
|
||||
| +----------+ +----------+ |
|
||||
| | | | | compr_stop() |
|
||||
+--->| FREE | | PAUSE |---------------------------+
|
||||
| | | |
|
||||
+----------+ +----------+
|
||||
|
||||
|
||||
Gapless Playback
|
||||
================
|
||||
|
|
@ -199,6 +250,38 @@ Sequence flow for gapless would be:
|
|||
|
||||
(note: order for partial_drain and write for next track can be reversed as well)
|
||||
|
||||
Gapless Playback SM
|
||||
===================
|
||||
|
||||
For Gapless, we move from running state to partial drain and back, along
|
||||
with setting of meta_data and signalling for next track ::
|
||||
|
||||
|
||||
+----------+
|
||||
compr_drain_notify() | |
|
||||
+------------------------>| RUNNING |
|
||||
| | |
|
||||
| +----------+
|
||||
| |
|
||||
| |
|
||||
| | compr_next_track()
|
||||
| |
|
||||
| V
|
||||
| +----------+
|
||||
| | |
|
||||
| |NEXT_TRACK|
|
||||
| | |
|
||||
| +----------+
|
||||
| |
|
||||
| |
|
||||
| | compr_partial_drain()
|
||||
| |
|
||||
| V
|
||||
| +----------+
|
||||
| | |
|
||||
+------------------------ | PARTIAL_ |
|
||||
| DRAIN |
|
||||
+----------+
|
||||
|
||||
Not supported
|
||||
=============
|
||||
|
|
|
|||
|
|
@ -91,7 +91,7 @@ PCM Proc Files
|
|||
|
||||
``card*/pcm*/xrun_debug``
|
||||
This file appears when ``CONFIG_SND_DEBUG=y`` and
|
||||
``CONFIG_PCM_XRUN_DEBUG=y``.
|
||||
``CONFIG_SND_PCM_XRUN_DEBUG=y``.
|
||||
This shows the status of xrun (= buffer overrun/xrun) and
|
||||
invalid PCM position debug/check of ALSA PCM middle layer.
|
||||
It takes an integer value, can be changed by writing to this
|
||||
|
|
|
|||
|
|
@ -42,7 +42,7 @@ If you are interested in the deep debugging of HD-audio, read the
|
|||
HD-audio specification at first. The specification is found on
|
||||
Intel's web page, for example:
|
||||
|
||||
* http://www.intel.com/standards/hdaudio/
|
||||
* https://www.intel.com/standards/hdaudio/
|
||||
|
||||
|
||||
HD-Audio Controller
|
||||
|
|
@ -728,7 +728,7 @@ version can be found on git repository:
|
|||
|
||||
The script can be fetched directly from the following URL, too:
|
||||
|
||||
* http://www.alsa-project.org/alsa-info.sh
|
||||
* https://www.alsa-project.org/alsa-info.sh
|
||||
|
||||
Run this script as root, and it will gather the important information
|
||||
such as the module lists, module parameters, proc file contents
|
||||
|
|
@ -818,7 +818,7 @@ proc-compatible output.
|
|||
|
||||
The hda-analyzer:
|
||||
|
||||
* http://git.alsa-project.org/?p=alsa.git;a=tree;f=hda-analyzer
|
||||
* https://git.alsa-project.org/?p=alsa.git;a=tree;f=hda-analyzer
|
||||
|
||||
is a part of alsa.git repository in alsa-project.org:
|
||||
|
||||
|
|
|
|||
|
|
@ -99,7 +99,7 @@ ASoC Core API
|
|||
.. kernel-doc:: include/sound/soc.h
|
||||
.. kernel-doc:: sound/soc/soc-core.c
|
||||
.. kernel-doc:: sound/soc/soc-devres.c
|
||||
.. kernel-doc:: sound/soc/soc-io.c
|
||||
.. kernel-doc:: sound/soc/soc-component.c
|
||||
.. kernel-doc:: sound/soc/soc-pcm.c
|
||||
.. kernel-doc:: sound/soc/soc-ops.c
|
||||
.. kernel-doc:: sound/soc/soc-compress.c
|
||||
|
|
|
|||
|
|
@ -3579,7 +3579,7 @@ dependent on the bus. For normal devices, pass the device pointer
|
|||
``SNDRV_DMA_TYPE_DEV`` type. For the continuous buffer unrelated to the
|
||||
bus can be pre-allocated with ``SNDRV_DMA_TYPE_CONTINUOUS`` type.
|
||||
You can pass NULL to the device pointer in that case, which is the
|
||||
default mode implying to allocate with ``GFP_KRENEL`` flag.
|
||||
default mode implying to allocate with ``GFP_KERNEL`` flag.
|
||||
If you need a different GFP flag, you can pass it by encoding the flag
|
||||
into the device pointer via a special macro
|
||||
:c:func:`snd_dma_continuous_data()`.
|
||||
|
|
|
|||
|
|
@ -17,7 +17,7 @@ frame (FRAME) (usually 48kHz) is always driven by the controller. Each AC97
|
|||
frame is 21uS long and is divided into 13 time slots.
|
||||
|
||||
The AC97 specification can be found at :
|
||||
http://www.intel.com/p/en_US/business/design
|
||||
https://www.intel.com/p/en_US/business/design
|
||||
|
||||
|
||||
I2S
|
||||
|
|
|
|||
|
|
@ -140,13 +140,13 @@
|
|||
compatible = "audio-graph-card";
|
||||
label = "Droid 4 Audio";
|
||||
|
||||
simple-graph-card,widgets =
|
||||
widgets =
|
||||
"Speaker", "Earpiece",
|
||||
"Speaker", "Loudspeaker",
|
||||
"Headphone", "Headphone Jack",
|
||||
"Microphone", "Internal Mic";
|
||||
|
||||
simple-graph-card,routing =
|
||||
routing =
|
||||
"Earpiece", "EP",
|
||||
"Loudspeaker", "SPKR",
|
||||
"Headphone Jack", "HSL",
|
||||
|
|
|
|||
|
|
@ -672,8 +672,8 @@ static void sii902x_audio_shutdown(struct device *dev, void *data)
|
|||
clk_disable_unprepare(sii902x->audio.mclk);
|
||||
}
|
||||
|
||||
static int sii902x_audio_digital_mute(struct device *dev,
|
||||
void *data, bool enable)
|
||||
static int sii902x_audio_mute(struct device *dev, void *data,
|
||||
bool enable, int direction)
|
||||
{
|
||||
struct sii902x *sii902x = dev_get_drvdata(dev);
|
||||
|
||||
|
|
@ -724,9 +724,10 @@ static int sii902x_audio_get_dai_id(struct snd_soc_component *component,
|
|||
static const struct hdmi_codec_ops sii902x_audio_codec_ops = {
|
||||
.hw_params = sii902x_audio_hw_params,
|
||||
.audio_shutdown = sii902x_audio_shutdown,
|
||||
.digital_mute = sii902x_audio_digital_mute,
|
||||
.mute_stream = sii902x_audio_mute,
|
||||
.get_eld = sii902x_audio_get_eld,
|
||||
.get_dai_id = sii902x_audio_get_dai_id,
|
||||
.no_capture_mute = 1,
|
||||
};
|
||||
|
||||
static int sii902x_audio_codec_init(struct sii902x *sii902x,
|
||||
|
|
|
|||
|
|
@ -1605,7 +1605,8 @@ static int hdmi_audio_hw_params(struct device *dev, void *data,
|
|||
return 0;
|
||||
}
|
||||
|
||||
static int hdmi_audio_digital_mute(struct device *dev, void *data, bool mute)
|
||||
static int hdmi_audio_mute(struct device *dev, void *data,
|
||||
bool mute, int direction)
|
||||
{
|
||||
struct hdmi_context *hdata = dev_get_drvdata(dev);
|
||||
|
||||
|
|
@ -1635,8 +1636,9 @@ static int hdmi_audio_get_eld(struct device *dev, void *data, uint8_t *buf,
|
|||
static const struct hdmi_codec_ops audio_codec_ops = {
|
||||
.hw_params = hdmi_audio_hw_params,
|
||||
.audio_shutdown = hdmi_audio_shutdown,
|
||||
.digital_mute = hdmi_audio_digital_mute,
|
||||
.mute_stream = hdmi_audio_mute,
|
||||
.get_eld = hdmi_audio_get_eld,
|
||||
.no_capture_mute = 1,
|
||||
};
|
||||
|
||||
static int hdmi_register_audio_device(struct hdmi_context *hdata)
|
||||
|
|
|
|||
|
|
@ -1133,8 +1133,8 @@ static void tda998x_audio_shutdown(struct device *dev, void *data)
|
|||
mutex_unlock(&priv->audio_mutex);
|
||||
}
|
||||
|
||||
static int tda998x_audio_digital_mute(struct device *dev, void *data,
|
||||
bool enable)
|
||||
static int tda998x_audio_mute_stream(struct device *dev, void *data,
|
||||
bool enable, int direction)
|
||||
{
|
||||
struct tda998x_priv *priv = dev_get_drvdata(dev);
|
||||
|
||||
|
|
@ -1162,8 +1162,9 @@ static int tda998x_audio_get_eld(struct device *dev, void *data,
|
|||
static const struct hdmi_codec_ops audio_codec_ops = {
|
||||
.hw_params = tda998x_audio_hw_params,
|
||||
.audio_shutdown = tda998x_audio_shutdown,
|
||||
.digital_mute = tda998x_audio_digital_mute,
|
||||
.mute_stream = tda998x_audio_mute_stream,
|
||||
.get_eld = tda998x_audio_get_eld,
|
||||
.no_capture_mute = 1,
|
||||
};
|
||||
|
||||
static int tda998x_audio_codec_init(struct tda998x_priv *priv,
|
||||
|
|
|
|||
|
|
@ -1643,7 +1643,8 @@ static void mtk_hdmi_audio_shutdown(struct device *dev, void *data)
|
|||
}
|
||||
|
||||
static int
|
||||
mtk_hdmi_audio_digital_mute(struct device *dev, void *data, bool enable)
|
||||
mtk_hdmi_audio_mute(struct device *dev, void *data,
|
||||
bool enable, int direction)
|
||||
{
|
||||
struct mtk_hdmi *hdmi = dev_get_drvdata(dev);
|
||||
|
||||
|
|
@ -1684,9 +1685,10 @@ static const struct hdmi_codec_ops mtk_hdmi_audio_codec_ops = {
|
|||
.hw_params = mtk_hdmi_audio_hw_params,
|
||||
.audio_startup = mtk_hdmi_audio_startup,
|
||||
.audio_shutdown = mtk_hdmi_audio_shutdown,
|
||||
.digital_mute = mtk_hdmi_audio_digital_mute,
|
||||
.mute_stream = mtk_hdmi_audio_mute,
|
||||
.get_eld = mtk_hdmi_audio_get_eld,
|
||||
.hook_plugged_cb = mtk_hdmi_audio_hook_plugged_cb,
|
||||
.no_capture_mute = 1,
|
||||
};
|
||||
|
||||
static int mtk_hdmi_register_audio_driver(struct device *dev)
|
||||
|
|
|
|||
|
|
@ -817,8 +817,8 @@ out:
|
|||
mutex_unlock(&dp->lock);
|
||||
}
|
||||
|
||||
static int cdn_dp_audio_digital_mute(struct device *dev, void *data,
|
||||
bool enable)
|
||||
static int cdn_dp_audio_mute_stream(struct device *dev, void *data,
|
||||
bool enable, int direction)
|
||||
{
|
||||
struct cdn_dp_device *dp = dev_get_drvdata(dev);
|
||||
int ret;
|
||||
|
|
@ -849,8 +849,9 @@ static int cdn_dp_audio_get_eld(struct device *dev, void *data,
|
|||
static const struct hdmi_codec_ops audio_codec_ops = {
|
||||
.hw_params = cdn_dp_audio_hw_params,
|
||||
.audio_shutdown = cdn_dp_audio_shutdown,
|
||||
.digital_mute = cdn_dp_audio_digital_mute,
|
||||
.mute_stream = cdn_dp_audio_mute_stream,
|
||||
.get_eld = cdn_dp_audio_get_eld,
|
||||
.no_capture_mute = 1,
|
||||
};
|
||||
|
||||
static int cdn_dp_audio_codec_init(struct cdn_dp_device *dp,
|
||||
|
|
|
|||
|
|
@ -1191,7 +1191,8 @@ static int hdmi_audio_hw_params(struct device *dev,
|
|||
return 0;
|
||||
}
|
||||
|
||||
static int hdmi_audio_digital_mute(struct device *dev, void *data, bool enable)
|
||||
static int hdmi_audio_mute(struct device *dev, void *data,
|
||||
bool enable, int direction)
|
||||
{
|
||||
struct sti_hdmi *hdmi = dev_get_drvdata(dev);
|
||||
|
||||
|
|
@ -1219,8 +1220,9 @@ static int hdmi_audio_get_eld(struct device *dev, void *data, uint8_t *buf, size
|
|||
static const struct hdmi_codec_ops audio_codec_ops = {
|
||||
.hw_params = hdmi_audio_hw_params,
|
||||
.audio_shutdown = hdmi_audio_shutdown,
|
||||
.digital_mute = hdmi_audio_digital_mute,
|
||||
.mute_stream = hdmi_audio_mute,
|
||||
.get_eld = hdmi_audio_get_eld,
|
||||
.no_capture_mute = 1,
|
||||
};
|
||||
|
||||
static int sti_hdmi_register_audio_driver(struct device *dev,
|
||||
|
|
|
|||
|
|
@ -439,8 +439,8 @@ static int zx_hdmi_audio_hw_params(struct device *dev,
|
|||
return zx_hdmi_infoframe_trans(hdmi, &frame, FSEL_AUDIO);
|
||||
}
|
||||
|
||||
static int zx_hdmi_audio_digital_mute(struct device *dev, void *data,
|
||||
bool enable)
|
||||
static int zx_hdmi_audio_mute(struct device *dev, void *data,
|
||||
bool enable, int direction)
|
||||
{
|
||||
struct zx_hdmi *hdmi = dev_get_drvdata(dev);
|
||||
|
||||
|
|
@ -468,8 +468,9 @@ static const struct hdmi_codec_ops zx_hdmi_codec_ops = {
|
|||
.audio_startup = zx_hdmi_audio_startup,
|
||||
.hw_params = zx_hdmi_audio_hw_params,
|
||||
.audio_shutdown = zx_hdmi_audio_shutdown,
|
||||
.digital_mute = zx_hdmi_audio_digital_mute,
|
||||
.mute_stream = zx_hdmi_audio_mute,
|
||||
.get_eld = zx_hdmi_audio_get_eld,
|
||||
.no_capture_mute = 1,
|
||||
};
|
||||
|
||||
static struct hdmi_codec_pdata zx_hdmi_codec_pdata = {
|
||||
|
|
|
|||
|
|
@ -986,7 +986,7 @@ static int lantiq_ssc_probe(struct platform_device *pdev)
|
|||
master->bits_per_word_mask = SPI_BPW_RANGE_MASK(2, 8) |
|
||||
SPI_BPW_MASK(16) | SPI_BPW_MASK(32);
|
||||
|
||||
spi->wq = alloc_ordered_workqueue(dev_name(dev), 0);
|
||||
spi->wq = alloc_ordered_workqueue(dev_name(dev), WQ_MEM_RECLAIM);
|
||||
if (!spi->wq) {
|
||||
err = -ENOMEM;
|
||||
goto err_clk_put;
|
||||
|
|
|
|||
|
|
@ -19,4 +19,8 @@
|
|||
#define MSM_FRONTEND_DAI_MULTIMEDIA15 14
|
||||
#define MSM_FRONTEND_DAI_MULTIMEDIA16 15
|
||||
|
||||
#define Q6ASM_DAI_TX_RX 0
|
||||
#define Q6ASM_DAI_TX 1
|
||||
#define Q6ASM_DAI_RX 2
|
||||
|
||||
#endif /* __DT_BINDINGS_Q6_ASM_H__ */
|
||||
|
|
|
|||
|
|
@ -188,20 +188,21 @@ int snd_ctl_enum_info(struct snd_ctl_elem_info *info, unsigned int channels,
|
|||
*/
|
||||
struct snd_kcontrol *snd_ctl_make_virtual_master(char *name,
|
||||
const unsigned int *tlv);
|
||||
int _snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave,
|
||||
unsigned int flags);
|
||||
/* optional flags for slave */
|
||||
#define SND_CTL_SLAVE_NEED_UPDATE (1 << 0)
|
||||
int _snd_ctl_add_follower(struct snd_kcontrol *master,
|
||||
struct snd_kcontrol *follower,
|
||||
unsigned int flags);
|
||||
/* optional flags for follower */
|
||||
#define SND_CTL_FOLLOWER_NEED_UPDATE (1 << 0)
|
||||
|
||||
/**
|
||||
* snd_ctl_add_slave - Add a virtual slave control
|
||||
* snd_ctl_add_follower - Add a virtual follower control
|
||||
* @master: vmaster element
|
||||
* @slave: slave element to add
|
||||
* @follower: follower element to add
|
||||
*
|
||||
* Add a virtual slave control to the given master element created via
|
||||
* Add a virtual follower control to the given master element created via
|
||||
* snd_ctl_create_virtual_master() beforehand.
|
||||
*
|
||||
* All slaves must be the same type (returning the same information
|
||||
* All followers must be the same type (returning the same information
|
||||
* via info callback). The function doesn't check it, so it's your
|
||||
* responsibility.
|
||||
*
|
||||
|
|
@ -213,18 +214,18 @@ int _snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave,
|
|||
* Return: Zero if successful or a negative error code.
|
||||
*/
|
||||
static inline int
|
||||
snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave)
|
||||
snd_ctl_add_follower(struct snd_kcontrol *master, struct snd_kcontrol *follower)
|
||||
{
|
||||
return _snd_ctl_add_slave(master, slave, 0);
|
||||
return _snd_ctl_add_follower(master, follower, 0);
|
||||
}
|
||||
|
||||
/**
|
||||
* snd_ctl_add_slave_uncached - Add a virtual slave control
|
||||
* snd_ctl_add_follower_uncached - Add a virtual follower control
|
||||
* @master: vmaster element
|
||||
* @slave: slave element to add
|
||||
* @follower: follower element to add
|
||||
*
|
||||
* Add a virtual slave control to the given master.
|
||||
* Unlike snd_ctl_add_slave(), the element added via this function
|
||||
* Add a virtual follower control to the given master.
|
||||
* Unlike snd_ctl_add_follower(), the element added via this function
|
||||
* is supposed to have volatile values, and get callback is called
|
||||
* at each time queried from the master.
|
||||
*
|
||||
|
|
@ -235,10 +236,10 @@ snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave)
|
|||
* Return: Zero if successful or a negative error code.
|
||||
*/
|
||||
static inline int
|
||||
snd_ctl_add_slave_uncached(struct snd_kcontrol *master,
|
||||
struct snd_kcontrol *slave)
|
||||
snd_ctl_add_follower_uncached(struct snd_kcontrol *master,
|
||||
struct snd_kcontrol *follower)
|
||||
{
|
||||
return _snd_ctl_add_slave(master, slave, SND_CTL_SLAVE_NEED_UPDATE);
|
||||
return _snd_ctl_add_follower(master, follower, SND_CTL_FOLLOWER_NEED_UPDATE);
|
||||
}
|
||||
|
||||
int snd_ctl_add_vmaster_hook(struct snd_kcontrol *kctl,
|
||||
|
|
@ -246,11 +247,11 @@ int snd_ctl_add_vmaster_hook(struct snd_kcontrol *kctl,
|
|||
void *private_data);
|
||||
void snd_ctl_sync_vmaster(struct snd_kcontrol *kctl, bool hook_only);
|
||||
#define snd_ctl_sync_vmaster_hook(kctl) snd_ctl_sync_vmaster(kctl, true)
|
||||
int snd_ctl_apply_vmaster_slaves(struct snd_kcontrol *kctl,
|
||||
int (*func)(struct snd_kcontrol *vslave,
|
||||
struct snd_kcontrol *slave,
|
||||
void *arg),
|
||||
void *arg);
|
||||
int snd_ctl_apply_vmaster_followers(struct snd_kcontrol *kctl,
|
||||
int (*func)(struct snd_kcontrol *vfollower,
|
||||
struct snd_kcontrol *follower,
|
||||
void *arg),
|
||||
void *arg);
|
||||
|
||||
/*
|
||||
* Helper functions for jack-detection controls
|
||||
|
|
|
|||
|
|
@ -613,4 +613,8 @@ int snd_gus_dram_write(struct snd_gus_card *gus, char __user *ptr,
|
|||
int snd_gus_dram_read(struct snd_gus_card *gus, char __user *ptr,
|
||||
unsigned int addr, unsigned int size, int rom);
|
||||
|
||||
/* gus_timer.c */
|
||||
void snd_gf1_timers_init(struct snd_gus_card *gus);
|
||||
void snd_gf1_timers_done(struct snd_gus_card *gus);
|
||||
|
||||
#endif /* __SOUND_GUS_H */
|
||||
|
|
|
|||
|
|
@ -208,7 +208,7 @@ struct hda_codec {
|
|||
struct mutex control_mutex;
|
||||
struct snd_array spdif_out;
|
||||
unsigned int spdif_in_enable; /* SPDIF input enable? */
|
||||
const hda_nid_t *slave_dig_outs; /* optional digital out slave widgets */
|
||||
const hda_nid_t *follower_dig_outs; /* optional digital out follower widgets */
|
||||
struct snd_array init_pins; /* initial (BIOS) pin configurations */
|
||||
struct snd_array driver_pins; /* pin configs set by codec parser */
|
||||
struct snd_array cvt_setups; /* audio convert setups */
|
||||
|
|
@ -415,6 +415,8 @@ __printf(2, 3)
|
|||
struct hda_pcm *snd_hda_codec_pcm_new(struct hda_codec *codec,
|
||||
const char *fmt, ...);
|
||||
|
||||
void snd_hda_codec_cleanup_for_unbind(struct hda_codec *codec);
|
||||
|
||||
static inline void snd_hda_codec_pcm_get(struct hda_pcm *pcm)
|
||||
{
|
||||
kref_get(&pcm->kref);
|
||||
|
|
|
|||
|
|
@ -347,6 +347,9 @@ struct hdac_bus {
|
|||
|
||||
int bdl_pos_adj; /* BDL position adjustment */
|
||||
|
||||
/* delay time in us for dma stop */
|
||||
unsigned int dma_stop_delay;
|
||||
|
||||
/* locks */
|
||||
spinlock_t reg_lock;
|
||||
struct mutex cmd_mutex;
|
||||
|
|
|
|||
|
|
@ -2,7 +2,7 @@
|
|||
/*
|
||||
* hdmi-codec.h - HDMI Codec driver API
|
||||
*
|
||||
* Copyright (C) 2014 Texas Instruments Incorporated - http://www.ti.com
|
||||
* Copyright (C) 2014 Texas Instruments Incorporated - https://www.ti.com
|
||||
*
|
||||
* Author: Jyri Sarha <jsarha@ti.com>
|
||||
*/
|
||||
|
|
@ -76,7 +76,8 @@ struct hdmi_codec_ops {
|
|||
* Mute/unmute HDMI audio stream.
|
||||
* Optional
|
||||
*/
|
||||
int (*digital_mute)(struct device *dev, void *data, bool enable);
|
||||
int (*mute_stream)(struct device *dev, void *data,
|
||||
bool enable, int direction);
|
||||
|
||||
/*
|
||||
* Provides EDID-Like-Data from connected HDMI device.
|
||||
|
|
@ -99,6 +100,9 @@ struct hdmi_codec_ops {
|
|||
int (*hook_plugged_cb)(struct device *dev, void *data,
|
||||
hdmi_codec_plugged_cb fn,
|
||||
struct device *codec_dev);
|
||||
|
||||
/* bit field */
|
||||
unsigned int no_capture_mute:1;
|
||||
};
|
||||
|
||||
/* HDMI codec initalization data */
|
||||
|
|
|
|||
|
|
@ -94,7 +94,11 @@ static inline dma_addr_t snd_sgbuf_get_addr(struct snd_dma_buffer *dmab,
|
|||
size_t offset)
|
||||
{
|
||||
struct snd_sg_buf *sgbuf = dmab->private_data;
|
||||
dma_addr_t addr = sgbuf->table[offset >> PAGE_SHIFT].addr;
|
||||
dma_addr_t addr;
|
||||
|
||||
if (!sgbuf)
|
||||
return dmab->addr + offset;
|
||||
addr = sgbuf->table[offset >> PAGE_SHIFT].addr;
|
||||
addr &= ~((dma_addr_t)PAGE_SIZE - 1);
|
||||
return addr + offset % PAGE_SIZE;
|
||||
}
|
||||
|
|
@ -106,6 +110,9 @@ static inline void *snd_sgbuf_get_ptr(struct snd_dma_buffer *dmab,
|
|||
size_t offset)
|
||||
{
|
||||
struct snd_sg_buf *sgbuf = dmab->private_data;
|
||||
|
||||
if (!sgbuf)
|
||||
return dmab->area + offset;
|
||||
return sgbuf->table[offset >> PAGE_SHIFT].buf + offset % PAGE_SIZE;
|
||||
}
|
||||
|
||||
|
|
|
|||
|
|
@ -2,7 +2,7 @@
|
|||
/*
|
||||
* hdmi-audio.c -- OMAP4+ DSS HDMI audio support library
|
||||
*
|
||||
* Copyright (C) 2014 Texas Instruments Incorporated - http://www.ti.com
|
||||
* Copyright (C) 2014 Texas Instruments Incorporated - https://www.ti.com
|
||||
*
|
||||
* Author: Jyri Sarha <jsarha@ti.com>
|
||||
*/
|
||||
|
|
|
|||
|
|
@ -1,26 +0,0 @@
|
|||
/* SPDX-License-Identifier: GPL-2.0-only */
|
||||
/*
|
||||
* linux/sound/rt5670.h -- Platform data for RT5670
|
||||
*
|
||||
* Copyright 2014 Realtek Microelectronics
|
||||
*/
|
||||
|
||||
#ifndef __LINUX_SND_RT5670_H
|
||||
#define __LINUX_SND_RT5670_H
|
||||
|
||||
struct rt5670_platform_data {
|
||||
int jd_mode;
|
||||
bool in2_diff;
|
||||
bool dev_gpio;
|
||||
bool gpio1_is_ext_spk_en;
|
||||
|
||||
bool dmic_en;
|
||||
unsigned int dmic1_data_pin;
|
||||
/* 0 = GPIO6; 1 = IN2P; 3 = GPIO7*/
|
||||
unsigned int dmic2_data_pin;
|
||||
/* 0 = GPIO8; 1 = IN3N; */
|
||||
unsigned int dmic3_data_pin;
|
||||
/* 0 = GPIO9; 1 = GPIO10; 2 = GPIO5*/
|
||||
};
|
||||
|
||||
#endif
|
||||
|
|
@ -12,9 +12,9 @@
|
|||
#include <sound/soc.h>
|
||||
|
||||
#define asoc_simple_init_hp(card, sjack, prefix) \
|
||||
asoc_simple_init_jack(card, sjack, 1, prefix)
|
||||
asoc_simple_init_jack(card, sjack, 1, prefix, NULL)
|
||||
#define asoc_simple_init_mic(card, sjack, prefix) \
|
||||
asoc_simple_init_jack(card, sjack, 0, prefix)
|
||||
asoc_simple_init_jack(card, sjack, 0, prefix, NULL)
|
||||
|
||||
struct asoc_simple_dai {
|
||||
const char *name;
|
||||
|
|
@ -131,7 +131,7 @@ int asoc_simple_parse_pin_switches(struct snd_soc_card *card,
|
|||
|
||||
int asoc_simple_init_jack(struct snd_soc_card *card,
|
||||
struct asoc_simple_jack *sjack,
|
||||
int is_hp, char *prefix);
|
||||
int is_hp, char *prefix, char *pin);
|
||||
int asoc_simple_init_priv(struct asoc_simple_priv *priv,
|
||||
struct link_info *li);
|
||||
|
||||
|
|
|
|||
|
|
@ -2,7 +2,8 @@
|
|||
*
|
||||
* soc-component.h
|
||||
*
|
||||
* Copyright (c) 2019 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
|
||||
* Copyright (C) 2019 Renesas Electronics Corp.
|
||||
* Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
|
||||
*/
|
||||
#ifndef __SOC_COMPONENT_H
|
||||
#define __SOC_COMPONENT_H
|
||||
|
|
@ -324,10 +325,12 @@ static inline int snd_soc_component_cache_sync(
|
|||
return regcache_sync(component->regmap);
|
||||
}
|
||||
|
||||
void snd_soc_component_set_aux(struct snd_soc_component *component,
|
||||
struct snd_soc_aux_dev *aux);
|
||||
int snd_soc_component_init(struct snd_soc_component *component);
|
||||
|
||||
/* component IO */
|
||||
int snd_soc_component_read(struct snd_soc_component *component,
|
||||
unsigned int reg, unsigned int *val);
|
||||
unsigned int snd_soc_component_read32(struct snd_soc_component *component,
|
||||
unsigned int snd_soc_component_read(struct snd_soc_component *component,
|
||||
unsigned int reg);
|
||||
int snd_soc_component_write(struct snd_soc_component *component,
|
||||
unsigned int reg, unsigned int val);
|
||||
|
|
@ -359,6 +362,7 @@ int snd_soc_component_stream_event(struct snd_soc_component *component,
|
|||
int snd_soc_component_set_bias_level(struct snd_soc_component *component,
|
||||
enum snd_soc_bias_level level);
|
||||
|
||||
void snd_soc_component_setup_regmap(struct snd_soc_component *component);
|
||||
#ifdef CONFIG_REGMAP
|
||||
void snd_soc_component_init_regmap(struct snd_soc_component *component,
|
||||
struct regmap *regmap);
|
||||
|
|
@ -421,16 +425,6 @@ int snd_soc_component_open(struct snd_soc_component *component,
|
|||
struct snd_pcm_substream *substream);
|
||||
int snd_soc_component_close(struct snd_soc_component *component,
|
||||
struct snd_pcm_substream *substream);
|
||||
int snd_soc_component_prepare(struct snd_soc_component *component,
|
||||
struct snd_pcm_substream *substream);
|
||||
int snd_soc_component_hw_params(struct snd_soc_component *component,
|
||||
struct snd_pcm_substream *substream,
|
||||
struct snd_pcm_hw_params *params);
|
||||
int snd_soc_component_hw_free(struct snd_soc_component *component,
|
||||
struct snd_pcm_substream *substream);
|
||||
int snd_soc_component_trigger(struct snd_soc_component *component,
|
||||
struct snd_pcm_substream *substream,
|
||||
int cmd);
|
||||
void snd_soc_component_suspend(struct snd_soc_component *component);
|
||||
void snd_soc_component_resume(struct snd_soc_component *component);
|
||||
int snd_soc_component_is_suspended(struct snd_soc_component *component);
|
||||
|
|
@ -455,5 +449,13 @@ int snd_soc_pcm_component_mmap(struct snd_pcm_substream *substream,
|
|||
struct vm_area_struct *vma);
|
||||
int snd_soc_pcm_component_new(struct snd_soc_pcm_runtime *rtd);
|
||||
void snd_soc_pcm_component_free(struct snd_soc_pcm_runtime *rtd);
|
||||
int snd_soc_pcm_component_prepare(struct snd_pcm_substream *substream);
|
||||
int snd_soc_pcm_component_hw_params(struct snd_pcm_substream *substream,
|
||||
struct snd_pcm_hw_params *params,
|
||||
struct snd_soc_component **last);
|
||||
void snd_soc_pcm_component_hw_free(struct snd_pcm_substream *substream,
|
||||
struct snd_soc_component *last);
|
||||
int snd_soc_pcm_component_trigger(struct snd_pcm_substream *substream,
|
||||
int cmd);
|
||||
|
||||
#endif /* __SOC_COMPONENT_H */
|
||||
|
|
|
|||
|
|
@ -39,7 +39,7 @@ struct snd_compr_stream;
|
|||
/*
|
||||
* DAI Clock gating.
|
||||
*
|
||||
* DAI bit clocks can be be gated (disabled) when the DAI is not
|
||||
* DAI bit clocks can be gated (disabled) when the DAI is not
|
||||
* sending or receiving PCM data in a frame. This can be used to save power.
|
||||
*/
|
||||
#define SND_SOC_DAIFMT_CONT (1 << 4) /* continuous clock */
|
||||
|
|
@ -76,12 +76,12 @@ struct snd_compr_stream;
|
|||
*
|
||||
* This is wrt the codec, the inverse is true for the interface
|
||||
* i.e. if the codec is clk and FRM master then the interface is
|
||||
* clk and frame slave.
|
||||
* clk and frame secondary.
|
||||
*/
|
||||
#define SND_SOC_DAIFMT_CBM_CFM (1 << 12) /* codec clk & FRM master */
|
||||
#define SND_SOC_DAIFMT_CBS_CFM (2 << 12) /* codec clk slave & FRM master */
|
||||
#define SND_SOC_DAIFMT_CBM_CFS (3 << 12) /* codec clk master & frame slave */
|
||||
#define SND_SOC_DAIFMT_CBS_CFS (4 << 12) /* codec clk & FRM slave */
|
||||
#define SND_SOC_DAIFMT_CBS_CFM (2 << 12) /* codec clk secondary & FRM master */
|
||||
#define SND_SOC_DAIFMT_CBM_CFS (3 << 12) /* codec clk master & frame secondary */
|
||||
#define SND_SOC_DAIFMT_CBS_CFS (4 << 12) /* codec clk & FRM secondary */
|
||||
|
||||
#define SND_SOC_DAIFMT_FORMAT_MASK 0x000f
|
||||
#define SND_SOC_DAIFMT_CLOCK_MASK 0x00f0
|
||||
|
|
@ -247,7 +247,6 @@ struct snd_soc_dai_ops {
|
|||
* DAI digital mute - optional.
|
||||
* Called by soc-core to minimise any pops.
|
||||
*/
|
||||
int (*digital_mute)(struct snd_soc_dai *dai, int mute);
|
||||
int (*mute_stream)(struct snd_soc_dai *dai, int mute, int stream);
|
||||
|
||||
/*
|
||||
|
|
@ -281,6 +280,9 @@ struct snd_soc_dai_ops {
|
|||
*/
|
||||
snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *,
|
||||
struct snd_soc_dai *);
|
||||
|
||||
/* bit field */
|
||||
unsigned int no_capture_mute:1;
|
||||
};
|
||||
|
||||
struct snd_soc_cdai_ops {
|
||||
|
|
|
|||
|
|
@ -16,6 +16,8 @@
|
|||
#include <sound/asoc.h>
|
||||
|
||||
struct device;
|
||||
struct snd_soc_pcm_runtime;
|
||||
struct soc_enum;
|
||||
|
||||
/* widget has no PM register bit */
|
||||
#define SND_SOC_NOPM -1
|
||||
|
|
@ -376,6 +378,24 @@ struct snd_soc_dapm_widget_list;
|
|||
struct snd_soc_dapm_update;
|
||||
enum snd_soc_dapm_direction;
|
||||
|
||||
/*
|
||||
* Bias levels
|
||||
*
|
||||
* @ON: Bias is fully on for audio playback and capture operations.
|
||||
* @PREPARE: Prepare for audio operations. Called before DAPM switching for
|
||||
* stream start and stop operations.
|
||||
* @STANDBY: Low power standby state when no playback/capture operations are
|
||||
* in progress. NOTE: The transition time between STANDBY and ON
|
||||
* should be as fast as possible and no longer than 10ms.
|
||||
* @OFF: Power Off. No restrictions on transition times.
|
||||
*/
|
||||
enum snd_soc_bias_level {
|
||||
SND_SOC_BIAS_OFF = 0,
|
||||
SND_SOC_BIAS_STANDBY = 1,
|
||||
SND_SOC_BIAS_PREPARE = 2,
|
||||
SND_SOC_BIAS_ON = 3,
|
||||
};
|
||||
|
||||
int dapm_regulator_event(struct snd_soc_dapm_widget *w,
|
||||
struct snd_kcontrol *kcontrol, int event);
|
||||
int dapm_clock_event(struct snd_soc_dapm_widget *w,
|
||||
|
|
|
|||
|
|
@ -9,6 +9,7 @@
|
|||
#define __SOC_LINK_H
|
||||
|
||||
int snd_soc_link_init(struct snd_soc_pcm_runtime *rtd);
|
||||
void snd_soc_link_exit(struct snd_soc_pcm_runtime *rtd);
|
||||
int snd_soc_link_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
|
||||
struct snd_pcm_hw_params *params);
|
||||
|
||||
|
|
|
|||
|
|
@ -368,24 +368,6 @@
|
|||
#define SOC_ENUM_SINGLE_VIRT_DECL(name, xtexts) \
|
||||
const struct soc_enum name = SOC_ENUM_SINGLE_VIRT(ARRAY_SIZE(xtexts), xtexts)
|
||||
|
||||
/*
|
||||
* Bias levels
|
||||
*
|
||||
* @ON: Bias is fully on for audio playback and capture operations.
|
||||
* @PREPARE: Prepare for audio operations. Called before DAPM switching for
|
||||
* stream start and stop operations.
|
||||
* @STANDBY: Low power standby state when no playback/capture operations are
|
||||
* in progress. NOTE: The transition time between STANDBY and ON
|
||||
* should be as fast as possible and no longer than 10ms.
|
||||
* @OFF: Power Off. No restrictions on transition times.
|
||||
*/
|
||||
enum snd_soc_bias_level {
|
||||
SND_SOC_BIAS_OFF = 0,
|
||||
SND_SOC_BIAS_STANDBY = 1,
|
||||
SND_SOC_BIAS_PREPARE = 2,
|
||||
SND_SOC_BIAS_ON = 3,
|
||||
};
|
||||
|
||||
struct device_node;
|
||||
struct snd_jack;
|
||||
struct snd_soc_card;
|
||||
|
|
@ -432,11 +414,12 @@ static inline int snd_soc_resume(struct device *dev)
|
|||
}
|
||||
#endif
|
||||
int snd_soc_poweroff(struct device *dev);
|
||||
int snd_soc_add_component(struct device *dev,
|
||||
struct snd_soc_component *component,
|
||||
const struct snd_soc_component_driver *component_driver,
|
||||
struct snd_soc_dai_driver *dai_drv,
|
||||
int num_dai);
|
||||
int snd_soc_component_initialize(struct snd_soc_component *component,
|
||||
const struct snd_soc_component_driver *driver,
|
||||
struct device *dev);
|
||||
int snd_soc_add_component(struct snd_soc_component *component,
|
||||
struct snd_soc_dai_driver *dai_drv,
|
||||
int num_dai);
|
||||
int snd_soc_register_component(struct device *dev,
|
||||
const struct snd_soc_component_driver *component_driver,
|
||||
struct snd_soc_dai_driver *dai_drv, int num_dai);
|
||||
|
|
@ -801,6 +784,9 @@ struct snd_soc_dai_link {
|
|||
/* codec/machine specific init - e.g. add machine controls */
|
||||
int (*init)(struct snd_soc_pcm_runtime *rtd);
|
||||
|
||||
/* codec/machine specific exit - dual of init() */
|
||||
void (*exit)(struct snd_soc_pcm_runtime *rtd);
|
||||
|
||||
/* optional hw_params re-writing for BE and FE sync */
|
||||
int (*be_hw_params_fixup)(struct snd_soc_pcm_runtime *rtd,
|
||||
struct snd_pcm_hw_params *params);
|
||||
|
|
@ -1183,6 +1169,8 @@ struct snd_soc_pcm_runtime {
|
|||
/* see soc_new_pcm_runtime() */
|
||||
#define asoc_rtd_to_cpu(rtd, n) (rtd)->dais[n]
|
||||
#define asoc_rtd_to_codec(rtd, n) (rtd)->dais[n + (rtd)->num_cpus]
|
||||
#define asoc_substream_to_rtd(substream) \
|
||||
(struct snd_soc_pcm_runtime *)snd_pcm_substream_chip(substream)
|
||||
|
||||
#define for_each_rtd_components(rtd, i, component) \
|
||||
for ((i) = 0, component = NULL; \
|
||||
|
|
|
|||
|
|
@ -16,6 +16,23 @@ struct wm8960_data {
|
|||
bool capless; /* Headphone outputs configured in capless mode */
|
||||
|
||||
bool shared_lrclk; /* DAC and ADC LRCLKs are wired together */
|
||||
|
||||
/*
|
||||
* Setup for headphone detection
|
||||
*
|
||||
* hp_cfg[0]: HPSEL[1:0] of R48 (Additional Control 4)
|
||||
* hp_cfg[1]: {HPSWEN:HPSWPOL} of R24 (Additional Control 2).
|
||||
* hp_cfg[2]: {TOCLKSEL:TOEN} of R23 (Additional Control 1).
|
||||
*/
|
||||
u32 hp_cfg[3];
|
||||
|
||||
/*
|
||||
* Setup for gpio configuration
|
||||
*
|
||||
* gpio_cfg[0]: ALRCGPIO of R9 (Audio interface)
|
||||
* gpio_cfg[1]: {GPIOPOL:GPIOSEL[2:0]} of R48 (Additional Control 4).
|
||||
*/
|
||||
u32 gpio_cfg[2];
|
||||
};
|
||||
|
||||
#endif
|
||||
|
|
|
|||
|
|
@ -219,7 +219,7 @@ static int atmel_ac97c_playback_prepare(struct snd_pcm_substream *substream)
|
|||
switch (runtime->format) {
|
||||
case SNDRV_PCM_FORMAT_S16_LE:
|
||||
break;
|
||||
case SNDRV_PCM_FORMAT_S16_BE: /* fall through */
|
||||
case SNDRV_PCM_FORMAT_S16_BE:
|
||||
word &= ~(AC97C_CMR_CEM_LITTLE);
|
||||
break;
|
||||
default:
|
||||
|
|
@ -301,7 +301,7 @@ static int atmel_ac97c_capture_prepare(struct snd_pcm_substream *substream)
|
|||
switch (runtime->format) {
|
||||
case SNDRV_PCM_FORMAT_S16_LE:
|
||||
break;
|
||||
case SNDRV_PCM_FORMAT_S16_BE: /* fall through */
|
||||
case SNDRV_PCM_FORMAT_S16_BE:
|
||||
word &= ~(AC97C_CMR_CEM_LITTLE);
|
||||
break;
|
||||
default:
|
||||
|
|
@ -356,14 +356,14 @@ atmel_ac97c_playback_trigger(struct snd_pcm_substream *substream, int cmd)
|
|||
camr = ac97c_readl(chip, CAMR);
|
||||
|
||||
switch (cmd) {
|
||||
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: /* fall through */
|
||||
case SNDRV_PCM_TRIGGER_RESUME: /* fall through */
|
||||
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
|
||||
case SNDRV_PCM_TRIGGER_RESUME:
|
||||
case SNDRV_PCM_TRIGGER_START:
|
||||
ptcr = ATMEL_PDC_TXTEN;
|
||||
camr |= AC97C_CMR_CENA | AC97C_CSR_ENDTX;
|
||||
break;
|
||||
case SNDRV_PCM_TRIGGER_PAUSE_PUSH: /* fall through */
|
||||
case SNDRV_PCM_TRIGGER_SUSPEND: /* fall through */
|
||||
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
|
||||
case SNDRV_PCM_TRIGGER_SUSPEND:
|
||||
case SNDRV_PCM_TRIGGER_STOP:
|
||||
ptcr |= ATMEL_PDC_TXTDIS;
|
||||
if (chip->opened <= 1)
|
||||
|
|
@ -388,14 +388,14 @@ atmel_ac97c_capture_trigger(struct snd_pcm_substream *substream, int cmd)
|
|||
ptcr = readl(chip->regs + ATMEL_PDC_PTSR);
|
||||
|
||||
switch (cmd) {
|
||||
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: /* fall through */
|
||||
case SNDRV_PCM_TRIGGER_RESUME: /* fall through */
|
||||
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
|
||||
case SNDRV_PCM_TRIGGER_RESUME:
|
||||
case SNDRV_PCM_TRIGGER_START:
|
||||
ptcr = ATMEL_PDC_RXTEN;
|
||||
camr |= AC97C_CMR_CENA | AC97C_CSR_ENDRX;
|
||||
break;
|
||||
case SNDRV_PCM_TRIGGER_PAUSE_PUSH: /* fall through */
|
||||
case SNDRV_PCM_TRIGGER_SUSPEND: /* fall through */
|
||||
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
|
||||
case SNDRV_PCM_TRIGGER_SUSPEND:
|
||||
case SNDRV_PCM_TRIGGER_STOP:
|
||||
ptcr |= ATMEL_PDC_RXTDIS;
|
||||
if (chip->opened <= 1)
|
||||
|
|
|
|||
|
|
@ -203,7 +203,10 @@ int snd_card_new(struct device *parent, int idx, const char *xid,
|
|||
mutex_unlock(&snd_card_mutex);
|
||||
card->dev = parent;
|
||||
card->number = idx;
|
||||
#ifdef MODULE
|
||||
WARN_ON(!module);
|
||||
card->module = module;
|
||||
#endif
|
||||
INIT_LIST_HEAD(&card->devices);
|
||||
init_rwsem(&card->controls_rwsem);
|
||||
rwlock_init(&card->ctl_files_rwlock);
|
||||
|
|
|
|||
|
|
@ -135,16 +135,17 @@ int snd_dma_alloc_pages(int type, struct device *device, size_t size,
|
|||
dmab->dev.type = type;
|
||||
dmab->dev.dev = device;
|
||||
dmab->bytes = 0;
|
||||
dmab->area = NULL;
|
||||
dmab->addr = 0;
|
||||
dmab->private_data = NULL;
|
||||
switch (type) {
|
||||
case SNDRV_DMA_TYPE_CONTINUOUS:
|
||||
gfp = snd_mem_get_gfp_flags(device, GFP_KERNEL);
|
||||
dmab->area = alloc_pages_exact(size, gfp);
|
||||
dmab->addr = 0;
|
||||
break;
|
||||
case SNDRV_DMA_TYPE_VMALLOC:
|
||||
gfp = snd_mem_get_gfp_flags(device, GFP_KERNEL | __GFP_HIGHMEM);
|
||||
dmab->area = __vmalloc(size, gfp);
|
||||
dmab->addr = 0;
|
||||
break;
|
||||
#ifdef CONFIG_HAS_DMA
|
||||
#ifdef CONFIG_GENERIC_ALLOCATOR
|
||||
|
|
@ -157,7 +158,7 @@ int snd_dma_alloc_pages(int type, struct device *device, size_t size,
|
|||
*/
|
||||
dmab->dev.type = SNDRV_DMA_TYPE_DEV;
|
||||
#endif /* CONFIG_GENERIC_ALLOCATOR */
|
||||
/* fall through */
|
||||
fallthrough;
|
||||
case SNDRV_DMA_TYPE_DEV:
|
||||
case SNDRV_DMA_TYPE_DEV_UC:
|
||||
snd_malloc_dev_pages(dmab, size);
|
||||
|
|
@ -171,8 +172,6 @@ int snd_dma_alloc_pages(int type, struct device *device, size_t size,
|
|||
#endif
|
||||
default:
|
||||
pr_err("snd-malloc: invalid device type %d\n", type);
|
||||
dmab->area = NULL;
|
||||
dmab->addr = 0;
|
||||
return -ENXIO;
|
||||
}
|
||||
if (! dmab->area)
|
||||
|
|
|
|||
|
|
@ -2851,7 +2851,7 @@ static int snd_pcm_oss_mmap(struct file *file, struct vm_area_struct *area)
|
|||
substream = pcm_oss_file->streams[SNDRV_PCM_STREAM_PLAYBACK];
|
||||
if (substream)
|
||||
break;
|
||||
/* Fall through */
|
||||
fallthrough;
|
||||
case VM_READ:
|
||||
substream = pcm_oss_file->streams[SNDRV_PCM_STREAM_CAPTURE];
|
||||
break;
|
||||
|
|
|
|||
|
|
@ -357,7 +357,7 @@ snd_pcm_format_t snd_pcm_plug_slave_format(snd_pcm_format_t format,
|
|||
if (snd_mask_test(format_mask, (__force int)format1))
|
||||
return format1;
|
||||
}
|
||||
/* fall through */
|
||||
fallthrough;
|
||||
default:
|
||||
return (__force snd_pcm_format_t)-EINVAL;
|
||||
}
|
||||
|
|
|
|||
|
|
@ -103,7 +103,7 @@ EXPORT_SYMBOL(snd_pcm_create_iec958_consumer);
|
|||
|
||||
/**
|
||||
* snd_pcm_create_iec958_consumer_hw_params - create IEC958 channel status
|
||||
* @hw_params: the hw_params instance for extracting rate and sample format
|
||||
* @params: the hw_params instance for extracting rate and sample format
|
||||
* @cs: channel status buffer, at least four bytes
|
||||
* @len: length of channel status buffer
|
||||
*
|
||||
|
|
|
|||
|
|
@ -39,6 +39,7 @@ static int do_alloc_pages(struct snd_card *card, int type, struct device *dev,
|
|||
if (max_alloc_per_card &&
|
||||
card->total_pcm_alloc_bytes + size > max_alloc_per_card)
|
||||
return -ENOMEM;
|
||||
|
||||
err = snd_dma_alloc_pages(type, dev, size, dmab);
|
||||
if (!err) {
|
||||
mutex_lock(&card->memory_mutex);
|
||||
|
|
|
|||
|
|
@ -1903,7 +1903,7 @@ static int snd_pcm_prepare(struct snd_pcm_substream *substream,
|
|||
switch (substream->runtime->status->state) {
|
||||
case SNDRV_PCM_STATE_PAUSED:
|
||||
snd_pcm_pause(substream, false);
|
||||
/* fallthru */
|
||||
fallthrough;
|
||||
case SNDRV_PCM_STATE_SUSPENDED:
|
||||
snd_pcm_stop(substream, SNDRV_PCM_STATE_SETUP);
|
||||
break;
|
||||
|
|
@ -2811,7 +2811,7 @@ static int do_pcm_hwsync(struct snd_pcm_substream *substream)
|
|||
case SNDRV_PCM_STATE_DRAINING:
|
||||
if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
|
||||
return -EBADFD;
|
||||
/* Fall through */
|
||||
fallthrough;
|
||||
case SNDRV_PCM_STATE_RUNNING:
|
||||
return snd_pcm_update_hw_ptr(substream);
|
||||
case SNDRV_PCM_STATE_PREPARED:
|
||||
|
|
@ -3713,7 +3713,6 @@ int snd_pcm_lib_default_mmap(struct snd_pcm_substream *substream,
|
|||
area->vm_end - area->vm_start, area->vm_page_prot);
|
||||
}
|
||||
#endif /* CONFIG_GENERIC_ALLOCATOR */
|
||||
#ifndef CONFIG_X86 /* for avoiding warnings arch/x86/mm/pat.c */
|
||||
if (IS_ENABLED(CONFIG_HAS_DMA) && !substream->ops->page &&
|
||||
(substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV ||
|
||||
substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV_UC))
|
||||
|
|
@ -3722,7 +3721,6 @@ int snd_pcm_lib_default_mmap(struct snd_pcm_substream *substream,
|
|||
substream->runtime->dma_area,
|
||||
substream->runtime->dma_addr,
|
||||
substream->runtime->dma_bytes);
|
||||
#endif /* CONFIG_X86 */
|
||||
/* mmap with fault handler */
|
||||
area->vm_ops = &snd_pcm_vm_ops_data_fault;
|
||||
return 0;
|
||||
|
|
@ -3816,7 +3814,7 @@ static int snd_pcm_mmap(struct file *file, struct vm_area_struct *area)
|
|||
case SNDRV_PCM_MMAP_OFFSET_STATUS_OLD:
|
||||
if (pcm_file->no_compat_mmap || !IS_ENABLED(CONFIG_64BIT))
|
||||
return -ENXIO;
|
||||
/* fallthrough */
|
||||
fallthrough;
|
||||
case SNDRV_PCM_MMAP_OFFSET_STATUS_NEW:
|
||||
if (!pcm_status_mmap_allowed(pcm_file))
|
||||
return -ENXIO;
|
||||
|
|
@ -3824,7 +3822,7 @@ static int snd_pcm_mmap(struct file *file, struct vm_area_struct *area)
|
|||
case SNDRV_PCM_MMAP_OFFSET_CONTROL_OLD:
|
||||
if (pcm_file->no_compat_mmap || !IS_ENABLED(CONFIG_64BIT))
|
||||
return -ENXIO;
|
||||
/* fallthrough */
|
||||
fallthrough;
|
||||
case SNDRV_PCM_MMAP_OFFSET_CONTROL_NEW:
|
||||
if (!pcm_control_mmap_allowed(pcm_file))
|
||||
return -ENXIO;
|
||||
|
|
|
|||
|
|
@ -168,10 +168,16 @@ static long
|
|||
odev_ioctl(struct file *file, unsigned int cmd, unsigned long arg)
|
||||
{
|
||||
struct seq_oss_devinfo *dp;
|
||||
long rc;
|
||||
|
||||
dp = file->private_data;
|
||||
if (snd_BUG_ON(!dp))
|
||||
return -ENXIO;
|
||||
return snd_seq_oss_ioctl(dp, cmd, arg);
|
||||
|
||||
mutex_lock(®ister_mutex);
|
||||
rc = snd_seq_oss_ioctl(dp, cmd, arg);
|
||||
mutex_unlock(®ister_mutex);
|
||||
return rc;
|
||||
}
|
||||
|
||||
#ifdef CONFIG_COMPAT
|
||||
|
|
|
|||
|
|
@ -79,7 +79,7 @@ snd_seq_oss_process_timer_event(struct seq_oss_timer *rec, union evrec *ev)
|
|||
case TMR_WAIT_REL:
|
||||
parm += rec->cur_tick;
|
||||
rec->realtime = 0;
|
||||
/* fall through */
|
||||
fallthrough;
|
||||
case TMR_WAIT_ABS:
|
||||
if (parm == 0) {
|
||||
rec->realtime = 1;
|
||||
|
|
|
|||
|
|
@ -309,7 +309,7 @@ do_control(const struct snd_midi_op *ops, void *drv,
|
|||
break;
|
||||
case MIDI_CTL_MSB_DATA_ENTRY:
|
||||
chan->control[MIDI_CTL_LSB_DATA_ENTRY] = 0;
|
||||
/* fall through */
|
||||
fallthrough;
|
||||
case MIDI_CTL_LSB_DATA_ENTRY:
|
||||
if (chan->param_type == SNDRV_MIDI_PARAM_TYPE_REGISTERED)
|
||||
rpn(ops, drv, chan, chset);
|
||||
|
|
|
|||
|
|
@ -142,6 +142,9 @@ unsigned int snd_sgbuf_get_chunk_size(struct snd_dma_buffer *dmab,
|
|||
struct snd_sg_buf *sg = dmab->private_data;
|
||||
unsigned int start, end, pg;
|
||||
|
||||
if (!sg)
|
||||
return size;
|
||||
|
||||
start = ofs >> PAGE_SHIFT;
|
||||
end = (ofs + size - 1) >> PAGE_SHIFT;
|
||||
/* check page continuity */
|
||||
|
|
|
|||
|
|
@ -1,6 +1,6 @@
|
|||
// SPDX-License-Identifier: GPL-2.0-only
|
||||
/*
|
||||
* Virtual master and slave controls
|
||||
* Virtual master and follower controls
|
||||
*
|
||||
* Copyright (c) 2008 by Takashi Iwai <tiwai@suse.de>
|
||||
*/
|
||||
|
|
@ -21,15 +21,15 @@ struct link_ctl_info {
|
|||
};
|
||||
|
||||
/*
|
||||
* link master - this contains a list of slave controls that are
|
||||
* link master - this contains a list of follower controls that are
|
||||
* identical types, i.e. info returns the same value type and value
|
||||
* ranges, but may have different number of counts.
|
||||
*
|
||||
* The master control is so far only mono volume/switch for simplicity.
|
||||
* The same value will be applied to all slaves.
|
||||
* The same value will be applied to all followers.
|
||||
*/
|
||||
struct link_master {
|
||||
struct list_head slaves;
|
||||
struct list_head followers;
|
||||
struct link_ctl_info info;
|
||||
int val; /* the master value */
|
||||
unsigned int tlv[4];
|
||||
|
|
@ -38,23 +38,23 @@ struct link_master {
|
|||
};
|
||||
|
||||
/*
|
||||
* link slave - this contains a slave control element
|
||||
* link follower - this contains a follower control element
|
||||
*
|
||||
* It fakes the control callbacsk with additional attenuation by the
|
||||
* master control. A slave may have either one or two channels.
|
||||
* It fakes the control callbacks with additional attenuation by the
|
||||
* master control. A follower may have either one or two channels.
|
||||
*/
|
||||
|
||||
struct link_slave {
|
||||
struct link_follower {
|
||||
struct list_head list;
|
||||
struct link_master *master;
|
||||
struct link_ctl_info info;
|
||||
int vals[2]; /* current values */
|
||||
unsigned int flags;
|
||||
struct snd_kcontrol *kctl; /* original kcontrol pointer */
|
||||
struct snd_kcontrol slave; /* the copy of original control entry */
|
||||
struct snd_kcontrol follower; /* the copy of original control entry */
|
||||
};
|
||||
|
||||
static int slave_update(struct link_slave *slave)
|
||||
static int follower_update(struct link_follower *follower)
|
||||
{
|
||||
struct snd_ctl_elem_value *uctl;
|
||||
int err, ch;
|
||||
|
|
@ -62,68 +62,68 @@ static int slave_update(struct link_slave *slave)
|
|||
uctl = kzalloc(sizeof(*uctl), GFP_KERNEL);
|
||||
if (!uctl)
|
||||
return -ENOMEM;
|
||||
uctl->id = slave->slave.id;
|
||||
err = slave->slave.get(&slave->slave, uctl);
|
||||
uctl->id = follower->follower.id;
|
||||
err = follower->follower.get(&follower->follower, uctl);
|
||||
if (err < 0)
|
||||
goto error;
|
||||
for (ch = 0; ch < slave->info.count; ch++)
|
||||
slave->vals[ch] = uctl->value.integer.value[ch];
|
||||
for (ch = 0; ch < follower->info.count; ch++)
|
||||
follower->vals[ch] = uctl->value.integer.value[ch];
|
||||
error:
|
||||
kfree(uctl);
|
||||
return err < 0 ? err : 0;
|
||||
}
|
||||
|
||||
/* get the slave ctl info and save the initial values */
|
||||
static int slave_init(struct link_slave *slave)
|
||||
/* get the follower ctl info and save the initial values */
|
||||
static int follower_init(struct link_follower *follower)
|
||||
{
|
||||
struct snd_ctl_elem_info *uinfo;
|
||||
int err;
|
||||
|
||||
if (slave->info.count) {
|
||||
if (follower->info.count) {
|
||||
/* already initialized */
|
||||
if (slave->flags & SND_CTL_SLAVE_NEED_UPDATE)
|
||||
return slave_update(slave);
|
||||
if (follower->flags & SND_CTL_FOLLOWER_NEED_UPDATE)
|
||||
return follower_update(follower);
|
||||
return 0;
|
||||
}
|
||||
|
||||
uinfo = kmalloc(sizeof(*uinfo), GFP_KERNEL);
|
||||
if (!uinfo)
|
||||
return -ENOMEM;
|
||||
uinfo->id = slave->slave.id;
|
||||
err = slave->slave.info(&slave->slave, uinfo);
|
||||
uinfo->id = follower->follower.id;
|
||||
err = follower->follower.info(&follower->follower, uinfo);
|
||||
if (err < 0) {
|
||||
kfree(uinfo);
|
||||
return err;
|
||||
}
|
||||
slave->info.type = uinfo->type;
|
||||
slave->info.count = uinfo->count;
|
||||
if (slave->info.count > 2 ||
|
||||
(slave->info.type != SNDRV_CTL_ELEM_TYPE_INTEGER &&
|
||||
slave->info.type != SNDRV_CTL_ELEM_TYPE_BOOLEAN)) {
|
||||
pr_err("ALSA: vmaster: invalid slave element\n");
|
||||
follower->info.type = uinfo->type;
|
||||
follower->info.count = uinfo->count;
|
||||
if (follower->info.count > 2 ||
|
||||
(follower->info.type != SNDRV_CTL_ELEM_TYPE_INTEGER &&
|
||||
follower->info.type != SNDRV_CTL_ELEM_TYPE_BOOLEAN)) {
|
||||
pr_err("ALSA: vmaster: invalid follower element\n");
|
||||
kfree(uinfo);
|
||||
return -EINVAL;
|
||||
}
|
||||
slave->info.min_val = uinfo->value.integer.min;
|
||||
slave->info.max_val = uinfo->value.integer.max;
|
||||
follower->info.min_val = uinfo->value.integer.min;
|
||||
follower->info.max_val = uinfo->value.integer.max;
|
||||
kfree(uinfo);
|
||||
|
||||
return slave_update(slave);
|
||||
return follower_update(follower);
|
||||
}
|
||||
|
||||
/* initialize master volume */
|
||||
static int master_init(struct link_master *master)
|
||||
{
|
||||
struct link_slave *slave;
|
||||
struct link_follower *follower;
|
||||
|
||||
if (master->info.count)
|
||||
return 0; /* already initialized */
|
||||
|
||||
list_for_each_entry(slave, &master->slaves, list) {
|
||||
int err = slave_init(slave);
|
||||
list_for_each_entry(follower, &master->followers, list) {
|
||||
int err = follower_init(follower);
|
||||
if (err < 0)
|
||||
return err;
|
||||
master->info = slave->info;
|
||||
master->info = follower->info;
|
||||
master->info.count = 1; /* always mono */
|
||||
/* set full volume as default (= no attenuation) */
|
||||
master->val = master->info.max_val;
|
||||
|
|
@ -134,113 +134,113 @@ static int master_init(struct link_master *master)
|
|||
return -ENOENT;
|
||||
}
|
||||
|
||||
static int slave_get_val(struct link_slave *slave,
|
||||
struct snd_ctl_elem_value *ucontrol)
|
||||
static int follower_get_val(struct link_follower *follower,
|
||||
struct snd_ctl_elem_value *ucontrol)
|
||||
{
|
||||
int err, ch;
|
||||
|
||||
err = slave_init(slave);
|
||||
err = follower_init(follower);
|
||||
if (err < 0)
|
||||
return err;
|
||||
for (ch = 0; ch < slave->info.count; ch++)
|
||||
ucontrol->value.integer.value[ch] = slave->vals[ch];
|
||||
for (ch = 0; ch < follower->info.count; ch++)
|
||||
ucontrol->value.integer.value[ch] = follower->vals[ch];
|
||||
return 0;
|
||||
}
|
||||
|
||||
static int slave_put_val(struct link_slave *slave,
|
||||
struct snd_ctl_elem_value *ucontrol)
|
||||
static int follower_put_val(struct link_follower *follower,
|
||||
struct snd_ctl_elem_value *ucontrol)
|
||||
{
|
||||
int err, ch, vol;
|
||||
|
||||
err = master_init(slave->master);
|
||||
err = master_init(follower->master);
|
||||
if (err < 0)
|
||||
return err;
|
||||
|
||||
switch (slave->info.type) {
|
||||
switch (follower->info.type) {
|
||||
case SNDRV_CTL_ELEM_TYPE_BOOLEAN:
|
||||
for (ch = 0; ch < slave->info.count; ch++)
|
||||
for (ch = 0; ch < follower->info.count; ch++)
|
||||
ucontrol->value.integer.value[ch] &=
|
||||
!!slave->master->val;
|
||||
!!follower->master->val;
|
||||
break;
|
||||
case SNDRV_CTL_ELEM_TYPE_INTEGER:
|
||||
for (ch = 0; ch < slave->info.count; ch++) {
|
||||
for (ch = 0; ch < follower->info.count; ch++) {
|
||||
/* max master volume is supposed to be 0 dB */
|
||||
vol = ucontrol->value.integer.value[ch];
|
||||
vol += slave->master->val - slave->master->info.max_val;
|
||||
if (vol < slave->info.min_val)
|
||||
vol = slave->info.min_val;
|
||||
else if (vol > slave->info.max_val)
|
||||
vol = slave->info.max_val;
|
||||
vol += follower->master->val - follower->master->info.max_val;
|
||||
if (vol < follower->info.min_val)
|
||||
vol = follower->info.min_val;
|
||||
else if (vol > follower->info.max_val)
|
||||
vol = follower->info.max_val;
|
||||
ucontrol->value.integer.value[ch] = vol;
|
||||
}
|
||||
break;
|
||||
}
|
||||
return slave->slave.put(&slave->slave, ucontrol);
|
||||
return follower->follower.put(&follower->follower, ucontrol);
|
||||
}
|
||||
|
||||
/*
|
||||
* ctl callbacks for slaves
|
||||
* ctl callbacks for followers
|
||||
*/
|
||||
static int slave_info(struct snd_kcontrol *kcontrol,
|
||||
struct snd_ctl_elem_info *uinfo)
|
||||
static int follower_info(struct snd_kcontrol *kcontrol,
|
||||
struct snd_ctl_elem_info *uinfo)
|
||||
{
|
||||
struct link_slave *slave = snd_kcontrol_chip(kcontrol);
|
||||
return slave->slave.info(&slave->slave, uinfo);
|
||||
struct link_follower *follower = snd_kcontrol_chip(kcontrol);
|
||||
return follower->follower.info(&follower->follower, uinfo);
|
||||
}
|
||||
|
||||
static int slave_get(struct snd_kcontrol *kcontrol,
|
||||
struct snd_ctl_elem_value *ucontrol)
|
||||
static int follower_get(struct snd_kcontrol *kcontrol,
|
||||
struct snd_ctl_elem_value *ucontrol)
|
||||
{
|
||||
struct link_slave *slave = snd_kcontrol_chip(kcontrol);
|
||||
return slave_get_val(slave, ucontrol);
|
||||
struct link_follower *follower = snd_kcontrol_chip(kcontrol);
|
||||
return follower_get_val(follower, ucontrol);
|
||||
}
|
||||
|
||||
static int slave_put(struct snd_kcontrol *kcontrol,
|
||||
struct snd_ctl_elem_value *ucontrol)
|
||||
static int follower_put(struct snd_kcontrol *kcontrol,
|
||||
struct snd_ctl_elem_value *ucontrol)
|
||||
{
|
||||
struct link_slave *slave = snd_kcontrol_chip(kcontrol);
|
||||
struct link_follower *follower = snd_kcontrol_chip(kcontrol);
|
||||
int err, ch, changed = 0;
|
||||
|
||||
err = slave_init(slave);
|
||||
err = follower_init(follower);
|
||||
if (err < 0)
|
||||
return err;
|
||||
for (ch = 0; ch < slave->info.count; ch++) {
|
||||
if (slave->vals[ch] != ucontrol->value.integer.value[ch]) {
|
||||
for (ch = 0; ch < follower->info.count; ch++) {
|
||||
if (follower->vals[ch] != ucontrol->value.integer.value[ch]) {
|
||||
changed = 1;
|
||||
slave->vals[ch] = ucontrol->value.integer.value[ch];
|
||||
follower->vals[ch] = ucontrol->value.integer.value[ch];
|
||||
}
|
||||
}
|
||||
if (!changed)
|
||||
return 0;
|
||||
err = slave_put_val(slave, ucontrol);
|
||||
err = follower_put_val(follower, ucontrol);
|
||||
if (err < 0)
|
||||
return err;
|
||||
return 1;
|
||||
}
|
||||
|
||||
static int slave_tlv_cmd(struct snd_kcontrol *kcontrol,
|
||||
int op_flag, unsigned int size,
|
||||
unsigned int __user *tlv)
|
||||
static int follower_tlv_cmd(struct snd_kcontrol *kcontrol,
|
||||
int op_flag, unsigned int size,
|
||||
unsigned int __user *tlv)
|
||||
{
|
||||
struct link_slave *slave = snd_kcontrol_chip(kcontrol);
|
||||
struct link_follower *follower = snd_kcontrol_chip(kcontrol);
|
||||
/* FIXME: this assumes that the max volume is 0 dB */
|
||||
return slave->slave.tlv.c(&slave->slave, op_flag, size, tlv);
|
||||
return follower->follower.tlv.c(&follower->follower, op_flag, size, tlv);
|
||||
}
|
||||
|
||||
static void slave_free(struct snd_kcontrol *kcontrol)
|
||||
static void follower_free(struct snd_kcontrol *kcontrol)
|
||||
{
|
||||
struct link_slave *slave = snd_kcontrol_chip(kcontrol);
|
||||
if (slave->slave.private_free)
|
||||
slave->slave.private_free(&slave->slave);
|
||||
if (slave->master)
|
||||
list_del(&slave->list);
|
||||
kfree(slave);
|
||||
struct link_follower *follower = snd_kcontrol_chip(kcontrol);
|
||||
if (follower->follower.private_free)
|
||||
follower->follower.private_free(&follower->follower);
|
||||
if (follower->master)
|
||||
list_del(&follower->list);
|
||||
kfree(follower);
|
||||
}
|
||||
|
||||
/*
|
||||
* Add a slave control to the group with the given master control
|
||||
* Add a follower control to the group with the given master control
|
||||
*
|
||||
* All slaves must be the same type (returning the same information
|
||||
* All followers must be the same type (returning the same information
|
||||
* via info callback). The function doesn't check it, so it's your
|
||||
* responsibility.
|
||||
*
|
||||
|
|
@ -249,35 +249,36 @@ static void slave_free(struct snd_kcontrol *kcontrol)
|
|||
* - logarithmic volume control (dB level), no linear volume
|
||||
* - master can only attenuate the volume, no gain
|
||||
*/
|
||||
int _snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave,
|
||||
unsigned int flags)
|
||||
int _snd_ctl_add_follower(struct snd_kcontrol *master,
|
||||
struct snd_kcontrol *follower,
|
||||
unsigned int flags)
|
||||
{
|
||||
struct link_master *master_link = snd_kcontrol_chip(master);
|
||||
struct link_slave *srec;
|
||||
struct link_follower *srec;
|
||||
|
||||
srec = kzalloc(struct_size(srec, slave.vd, slave->count),
|
||||
srec = kzalloc(struct_size(srec, follower.vd, follower->count),
|
||||
GFP_KERNEL);
|
||||
if (!srec)
|
||||
return -ENOMEM;
|
||||
srec->kctl = slave;
|
||||
srec->slave = *slave;
|
||||
memcpy(srec->slave.vd, slave->vd, slave->count * sizeof(*slave->vd));
|
||||
srec->kctl = follower;
|
||||
srec->follower = *follower;
|
||||
memcpy(srec->follower.vd, follower->vd, follower->count * sizeof(*follower->vd));
|
||||
srec->master = master_link;
|
||||
srec->flags = flags;
|
||||
|
||||
/* override callbacks */
|
||||
slave->info = slave_info;
|
||||
slave->get = slave_get;
|
||||
slave->put = slave_put;
|
||||
if (slave->vd[0].access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK)
|
||||
slave->tlv.c = slave_tlv_cmd;
|
||||
slave->private_data = srec;
|
||||
slave->private_free = slave_free;
|
||||
follower->info = follower_info;
|
||||
follower->get = follower_get;
|
||||
follower->put = follower_put;
|
||||
if (follower->vd[0].access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK)
|
||||
follower->tlv.c = follower_tlv_cmd;
|
||||
follower->private_data = srec;
|
||||
follower->private_free = follower_free;
|
||||
|
||||
list_add_tail(&srec->list, &master_link->slaves);
|
||||
list_add_tail(&srec->list, &master_link->followers);
|
||||
return 0;
|
||||
}
|
||||
EXPORT_SYMBOL(_snd_ctl_add_slave);
|
||||
EXPORT_SYMBOL(_snd_ctl_add_follower);
|
||||
|
||||
/*
|
||||
* ctl callbacks for master controls
|
||||
|
|
@ -309,20 +310,20 @@ static int master_get(struct snd_kcontrol *kcontrol,
|
|||
return 0;
|
||||
}
|
||||
|
||||
static int sync_slaves(struct link_master *master, int old_val, int new_val)
|
||||
static int sync_followers(struct link_master *master, int old_val, int new_val)
|
||||
{
|
||||
struct link_slave *slave;
|
||||
struct link_follower *follower;
|
||||
struct snd_ctl_elem_value *uval;
|
||||
|
||||
uval = kmalloc(sizeof(*uval), GFP_KERNEL);
|
||||
if (!uval)
|
||||
return -ENOMEM;
|
||||
list_for_each_entry(slave, &master->slaves, list) {
|
||||
list_for_each_entry(follower, &master->followers, list) {
|
||||
master->val = old_val;
|
||||
uval->id = slave->slave.id;
|
||||
slave_get_val(slave, uval);
|
||||
uval->id = follower->follower.id;
|
||||
follower_get_val(follower, uval);
|
||||
master->val = new_val;
|
||||
slave_put_val(slave, uval);
|
||||
follower_put_val(follower, uval);
|
||||
}
|
||||
kfree(uval);
|
||||
return 0;
|
||||
|
|
@ -344,7 +345,7 @@ static int master_put(struct snd_kcontrol *kcontrol,
|
|||
if (new_val == old_val)
|
||||
return 0;
|
||||
|
||||
err = sync_slaves(master, old_val, new_val);
|
||||
err = sync_followers(master, old_val, new_val);
|
||||
if (err < 0)
|
||||
return err;
|
||||
if (master->hook && !first_init)
|
||||
|
|
@ -355,17 +356,17 @@ static int master_put(struct snd_kcontrol *kcontrol,
|
|||
static void master_free(struct snd_kcontrol *kcontrol)
|
||||
{
|
||||
struct link_master *master = snd_kcontrol_chip(kcontrol);
|
||||
struct link_slave *slave, *n;
|
||||
struct link_follower *follower, *n;
|
||||
|
||||
/* free all slave links and retore the original slave kctls */
|
||||
list_for_each_entry_safe(slave, n, &master->slaves, list) {
|
||||
struct snd_kcontrol *sctl = slave->kctl;
|
||||
/* free all follower links and retore the original follower kctls */
|
||||
list_for_each_entry_safe(follower, n, &master->followers, list) {
|
||||
struct snd_kcontrol *sctl = follower->kctl;
|
||||
struct list_head olist = sctl->list;
|
||||
memcpy(sctl, &slave->slave, sizeof(*sctl));
|
||||
memcpy(sctl->vd, slave->slave.vd,
|
||||
memcpy(sctl, &follower->follower, sizeof(*sctl));
|
||||
memcpy(sctl->vd, follower->follower.vd,
|
||||
sctl->count * sizeof(*sctl->vd));
|
||||
sctl->list = olist; /* keep the current linked-list */
|
||||
kfree(slave);
|
||||
kfree(follower);
|
||||
}
|
||||
kfree(master);
|
||||
}
|
||||
|
|
@ -378,8 +379,8 @@ static void master_free(struct snd_kcontrol *kcontrol)
|
|||
*
|
||||
* Creates a virtual master control with the given name string.
|
||||
*
|
||||
* After creating a vmaster element, you can add the slave controls
|
||||
* via snd_ctl_add_slave() or snd_ctl_add_slave_uncached().
|
||||
* After creating a vmaster element, you can add the follower controls
|
||||
* via snd_ctl_add_follower() or snd_ctl_add_follower_uncached().
|
||||
*
|
||||
* The optional argument @tlv can be used to specify the TLV information
|
||||
* for dB scale of the master control. It should be a single element
|
||||
|
|
@ -403,7 +404,7 @@ struct snd_kcontrol *snd_ctl_make_virtual_master(char *name,
|
|||
master = kzalloc(sizeof(*master), GFP_KERNEL);
|
||||
if (!master)
|
||||
return NULL;
|
||||
INIT_LIST_HEAD(&master->slaves);
|
||||
INIT_LIST_HEAD(&master->followers);
|
||||
|
||||
kctl = snd_ctl_new1(&knew, master);
|
||||
if (!kctl) {
|
||||
|
|
@ -455,11 +456,11 @@ int snd_ctl_add_vmaster_hook(struct snd_kcontrol *kcontrol,
|
|||
EXPORT_SYMBOL_GPL(snd_ctl_add_vmaster_hook);
|
||||
|
||||
/**
|
||||
* snd_ctl_sync_vmaster - Sync the vmaster slaves and hook
|
||||
* snd_ctl_sync_vmaster - Sync the vmaster followers and hook
|
||||
* @kcontrol: vmaster kctl element
|
||||
* @hook_only: sync only the hook
|
||||
*
|
||||
* Forcibly call the put callback of each slave and call the hook function
|
||||
* Forcibly call the put callback of each follower and call the hook function
|
||||
* to synchronize with the current value of the given vmaster element.
|
||||
* NOP when NULL is passed to @kcontrol.
|
||||
*/
|
||||
|
|
@ -476,7 +477,7 @@ void snd_ctl_sync_vmaster(struct snd_kcontrol *kcontrol, bool hook_only)
|
|||
if (err < 0)
|
||||
return;
|
||||
first_init = err;
|
||||
err = sync_slaves(master, master->val, master->val);
|
||||
err = sync_followers(master, master->val, master->val);
|
||||
if (err < 0)
|
||||
return;
|
||||
}
|
||||
|
|
@ -487,34 +488,34 @@ void snd_ctl_sync_vmaster(struct snd_kcontrol *kcontrol, bool hook_only)
|
|||
EXPORT_SYMBOL_GPL(snd_ctl_sync_vmaster);
|
||||
|
||||
/**
|
||||
* snd_ctl_apply_vmaster_slaves - Apply function to each vmaster slave
|
||||
* snd_ctl_apply_vmaster_followers - Apply function to each vmaster follower
|
||||
* @kctl: vmaster kctl element
|
||||
* @func: function to apply
|
||||
* @arg: optional function argument
|
||||
*
|
||||
* Apply the function @func to each slave kctl of the given vmaster kctl.
|
||||
* Apply the function @func to each follower kctl of the given vmaster kctl.
|
||||
* Returns 0 if successful, or a negative error code.
|
||||
*/
|
||||
int snd_ctl_apply_vmaster_slaves(struct snd_kcontrol *kctl,
|
||||
int (*func)(struct snd_kcontrol *vslave,
|
||||
struct snd_kcontrol *slave,
|
||||
void *arg),
|
||||
void *arg)
|
||||
int snd_ctl_apply_vmaster_followers(struct snd_kcontrol *kctl,
|
||||
int (*func)(struct snd_kcontrol *vfollower,
|
||||
struct snd_kcontrol *follower,
|
||||
void *arg),
|
||||
void *arg)
|
||||
{
|
||||
struct link_master *master;
|
||||
struct link_slave *slave;
|
||||
struct link_follower *follower;
|
||||
int err;
|
||||
|
||||
master = snd_kcontrol_chip(kctl);
|
||||
err = master_init(master);
|
||||
if (err < 0)
|
||||
return err;
|
||||
list_for_each_entry(slave, &master->slaves, list) {
|
||||
err = func(slave->kctl, &slave->slave, arg);
|
||||
list_for_each_entry(follower, &master->followers, list) {
|
||||
err = func(follower->kctl, &follower->follower, arg);
|
||||
if (err < 0)
|
||||
return err;
|
||||
}
|
||||
|
||||
return 0;
|
||||
}
|
||||
EXPORT_SYMBOL_GPL(snd_ctl_apply_vmaster_slaves);
|
||||
EXPORT_SYMBOL_GPL(snd_ctl_apply_vmaster_followers);
|
||||
|
|
|
|||
|
|
@ -354,7 +354,7 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan)
|
|||
instr_4op = 1;
|
||||
break;
|
||||
}
|
||||
/* fall through */
|
||||
fallthrough;
|
||||
default:
|
||||
spin_unlock_irqrestore(&opl3->voice_lock, flags);
|
||||
return;
|
||||
|
|
@ -443,7 +443,7 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan)
|
|||
switch (connection) {
|
||||
case 0x03:
|
||||
snd_opl3_calc_volume(&vol_op[2], vel, chan);
|
||||
/* fallthru */
|
||||
fallthrough;
|
||||
case 0x02:
|
||||
snd_opl3_calc_volume(&vol_op[0], vel, chan);
|
||||
break;
|
||||
|
|
|
|||
|
|
@ -511,8 +511,9 @@ irqreturn_t snd_vx_threaded_irq_handler(int irq, void *dev)
|
|||
/* The start on time code conditions are filled (ie the time code
|
||||
* received by the board is equal to one of those given to it).
|
||||
*/
|
||||
if (events & TIME_CODE_EVENT_PENDING)
|
||||
if (events & TIME_CODE_EVENT_PENDING) {
|
||||
; /* so far, nothing to do yet */
|
||||
}
|
||||
|
||||
/* The frequency has changed on the board (UER mode). */
|
||||
if (events & FREQUENCY_CHANGE_EVENT_PENDING)
|
||||
|
|
|
|||
|
|
@ -293,7 +293,6 @@ static int pcr_set_check(struct cmp_connection *c, __be32 pcr)
|
|||
/**
|
||||
* cmp_connection_establish - establish a connection to the target
|
||||
* @c: the connection manager
|
||||
* @max_payload_bytes: the amount of data (including CIP headers) per packet
|
||||
*
|
||||
* This function establishes a point-to-point connection from the local
|
||||
* computer to the target by allocating isochronous resources (channel and
|
||||
|
|
|
|||
|
|
@ -24,6 +24,9 @@
|
|||
#define V3_NO_ADAT_OPT_OUT_IFACE_A 0x00040000
|
||||
#define V3_NO_ADAT_OPT_OUT_IFACE_B 0x00400000
|
||||
|
||||
#define V3_MSG_FLAG_CLK_CHANGED 0x00000002
|
||||
#define V3_CLK_WAIT_MSEC 4000
|
||||
|
||||
int snd_motu_protocol_v3_get_clock_rate(struct snd_motu *motu,
|
||||
unsigned int *rate)
|
||||
{
|
||||
|
|
@ -79,9 +82,16 @@ int snd_motu_protocol_v3_set_clock_rate(struct snd_motu *motu,
|
|||
return err;
|
||||
|
||||
if (need_to_wait) {
|
||||
/* Cost expensive. */
|
||||
if (msleep_interruptible(4000) > 0)
|
||||
return -EINTR;
|
||||
int result;
|
||||
|
||||
motu->msg = 0;
|
||||
result = wait_event_interruptible_timeout(motu->hwdep_wait,
|
||||
motu->msg & V3_MSG_FLAG_CLK_CHANGED,
|
||||
msecs_to_jiffies(V3_CLK_WAIT_MSEC));
|
||||
if (result < 0)
|
||||
return result;
|
||||
if (result == 0)
|
||||
return -ETIMEDOUT;
|
||||
}
|
||||
|
||||
return 0;
|
||||
|
|
|
|||
Some files were not shown because too many files have changed in this diff Show More
Loading…
Reference in New Issue