sound updates for 6.10-rc1
This one became bigger than usual, not in the total size but rather
containing lots of small changes all over the places. The majority of
changes are about ASoC, especially SOF / Intel stuff, and we see an
interesting work for ASoC DAPM graph visualization, while there are
many other code cleanup and refactoring, too.
Core:
- A deadlock fix at device disconnection
- A new tool dapm-graph for visualising the DAPM state
ASoC:
- Large updates throughout the Intel audio drivers
- Fixes and clarifications for the DAPM documentation
- Cleanups of accessors for driver data, module labelling, and for
constification
- Modernsation and cleanup work in the Mediatek drivers
- Several fixes and features for the DaVinci I2S driver
- New drivers for several AMD and Intel platforms, Nuvoton NAU8325,
Rockchip RK3308 and Texas Instruments PCM6240
HD-audio:
- Cleanup for CONFIG_PM dependencies
- Cirrus HD-audio codec fixes and quirks
Others:
- Series of tree-wide fixes in Makefiles to use *-y
- Additions of missing module descriptions
- Scarlett2 USB mixer enhancements
- A series of legacy emu10k1 fixes and improvements
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Merge tag 'sound-6.10-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"This one became bigger than usual, not in the total size but rather
containing lots of small changes all over the places.
The majority of changes are about ASoC, especially SOF / Intel stuff,
and we see an interesting work for ASoC DAPM graph visualization,
while there are many other code cleanup and refactoring, too.
Core:
- A deadlock fix at device disconnection
- A new tool dapm-graph for visualising the DAPM state
ASoC:
- Large updates throughout the Intel audio drivers
- Fixes and clarifications for the DAPM documentation
- Cleanups of accessors for driver data, module labelling, and for
constification
- Modernsation and cleanup work in the Mediatek drivers
- Several fixes and features for the DaVinci I2S driver
- New drivers for several AMD and Intel platforms, Nuvoton NAU8325,
Rockchip RK3308 and Texas Instruments PCM6240
HD-audio:
- Cleanup for CONFIG_PM dependencies
- Cirrus HD-audio codec fixes and quirks
Others:
- Series of tree-wide fixes in Makefiles to use *-y
- Additions of missing module descriptions
- Scarlett2 USB mixer enhancements
- A series of legacy emu10k1 fixes and improvements"
* tag 'sound-6.10-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (603 commits)
ALSA: hda/realtek: Drop doubly quirk entry for 103c:8a2e
ALSA: hda/realtek - fixed headset Mic not show
ASoC: SOF: amd: Fix build error with built-in config
ALSA: scarlett2: Increase mixer range to +12dB
ALSA: scarlett2: Add S/PDIF source selection controls
ALSA: core: Remove superfluous CONFIG_PM
ALSA: Fix deadlocks with kctl removals at disconnection
ASoC: audio-graph-card2: call of_node_get() before of_get_next_child()
ASoC: SOF: amd: Correct spaces in Makefile
ASoC: rt715-sdca-sdw: Fix wrong complete waiting in rt715_dev_resume()
ASoC: Intel: sof_sdw_rt_amp: use dai parameter
ASoC: Intel: sof_sdw: add dai parameter to rtd_init callback
ASoC: Intel: sof_sdw: use .controls/.widgets to add controls/widgets
ASoC: Intel: sof_sdw: add controls and dapm widgets in codec_info
ASoC: Intel: sof_sdw: use generic name for controls/widgets
ASoC: Intel: sof_sdw_cs_amp: rename Speakers to Speaker
ASoC: Intel: maxim-common: change max98373 data to static
ASoC: Intel: sof_sdw: add max98373 dapm routes
ASoC: Intel: sof_rt5682: use max_98373_dai_link function
ASoC: Intel: sof_nau8825: use max_98373_dai_link function
...
This commit is contained in:
Linus Torvalds
commit
33e02dc69a
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@ -1,50 +0,0 @@
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Texas Instruments DaVinci McBSP module
|
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~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
|
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|
||||
This binding describes the "Multi-channel Buffered Serial Port" (McBSP)
|
||||
audio interface found in some TI DaVinci processors like the OMAP-L138 or AM180x.
|
||||
|
||||
|
||||
Required properties:
|
||||
~~~~~~~~~~~~~~~~~~~~
|
||||
- compatible :
|
||||
"ti,da850-mcbsp" : for DA850, AM180x and OPAM-L138 platforms
|
||||
|
||||
- reg : physical base address and length of the controller memory mapped
|
||||
region(s).
|
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- reg-names : Should contain:
|
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* "mpu" for the main registers (required).
|
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* "dat" for the data FIFO (optional).
|
||||
|
||||
- dmas: three element list of DMA controller phandles, DMA request line and
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TC channel ordered triplets.
|
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- dma-names: identifier string for each DMA request line in the dmas property.
|
||||
These strings correspond 1:1 with the ordered pairs in dmas. The dma
|
||||
identifiers must be "rx" and "tx".
|
||||
|
||||
Optional properties:
|
||||
~~~~~~~~~~~~~~~~~~~~
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||||
- interrupts : Interrupt numbers for McBSP
|
||||
- interrupt-names : Known interrupt names are "rx" and "tx"
|
||||
|
||||
- pinctrl-0: Should specify pin control group used for this controller.
|
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- pinctrl-names: Should contain only one value - "default", for more details
|
||||
please refer to pinctrl-bindings.txt
|
||||
|
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Example (AM1808):
|
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~~~~~~~~~~~~~~~~~
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|
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mcbsp0: mcbsp@1d10000 {
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compatible = "ti,da850-mcbsp";
|
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pinctrl-names = "default";
|
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pinctrl-0 = <&mcbsp0_pins>;
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|
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reg = <0x00110000 0x1000>,
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<0x00310000 0x1000>;
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reg-names = "mpu", "dat";
|
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interrupts = <97 98>;
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interrupt-names = "rx", "tx";
|
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dmas = <&edma0 3 1
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&edma0 2 1>;
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dma-names = "tx", "rx";
|
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};
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@ -0,0 +1,113 @@
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|||
# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
|
||||
%YAML 1.2
|
||||
---
|
||||
$id: http://devicetree.org/schemas/sound/davinci-mcbsp.yaml#
|
||||
$schema: http://devicetree.org/meta-schemas/core.yaml#
|
||||
|
||||
title: McBSP Controller for TI SoCs
|
||||
|
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maintainers:
|
||||
- Bastien Curutchet <bastien.curutchet@bootlin.com>
|
||||
|
||||
allOf:
|
||||
- $ref: dai-common.yaml#
|
||||
|
||||
properties:
|
||||
compatible:
|
||||
enum:
|
||||
- ti,da850-mcbsp
|
||||
|
||||
reg:
|
||||
minItems: 1
|
||||
items:
|
||||
- description: CFG registers
|
||||
- description: data registers
|
||||
|
||||
reg-names:
|
||||
minItems: 1
|
||||
items:
|
||||
- const: mpu
|
||||
- const: dat
|
||||
|
||||
dmas:
|
||||
items:
|
||||
- description: transmission DMA channel
|
||||
- description: reception DMA channel
|
||||
|
||||
dma-names:
|
||||
items:
|
||||
- const: tx
|
||||
- const: rx
|
||||
|
||||
interrupts:
|
||||
items:
|
||||
- description: RX interrupt
|
||||
- description: TX interrupt
|
||||
|
||||
interrupt-names:
|
||||
items:
|
||||
- const: rx
|
||||
- const: tx
|
||||
|
||||
clocks:
|
||||
minItems: 1
|
||||
items:
|
||||
- description: functional clock
|
||||
- description: external input clock for sample rate generator.
|
||||
|
||||
clock-names:
|
||||
minItems: 1
|
||||
items:
|
||||
- const: fck
|
||||
- const: clks
|
||||
|
||||
power-domains:
|
||||
maxItems: 1
|
||||
|
||||
"#sound-dai-cells":
|
||||
const: 0
|
||||
|
||||
ti,T1-framing-tx:
|
||||
$ref: /schemas/types.yaml#/definitions/flag
|
||||
description:
|
||||
If the property is present, tx data delay is set to 2 bit clock periods.
|
||||
McBSP will insert a blank period (high-impedance period) before the first
|
||||
data bit. This can be used to interface to T1-framing devices.
|
||||
|
||||
ti,T1-framing-rx:
|
||||
$ref: /schemas/types.yaml#/definitions/flag
|
||||
description:
|
||||
If the property is present, rx data delay is set to 2 bit clock periods.
|
||||
McBSP will discard the bit preceding the data stream (called framing bit).
|
||||
This can be used to interface to T1-framing devices.
|
||||
|
||||
required:
|
||||
- "#sound-dai-cells"
|
||||
- compatible
|
||||
- reg
|
||||
- reg-names
|
||||
- dmas
|
||||
- dma-names
|
||||
- clocks
|
||||
|
||||
unevaluatedProperties: false
|
||||
|
||||
examples:
|
||||
- |
|
||||
mcbsp0@1d10000 {
|
||||
#sound-dai-cells = <0>;
|
||||
compatible = "ti,da850-mcbsp";
|
||||
pinctrl-names = "default";
|
||||
pinctrl-0 = <&mcbsp0_pins>;
|
||||
|
||||
reg = <0x111000 0x1000>,
|
||||
<0x311000 0x1000>;
|
||||
reg-names = "mpu", "dat";
|
||||
interrupts = <97>, <98>;
|
||||
interrupt-names = "rx", "tx";
|
||||
dmas = <&edma0 3 1>,
|
||||
<&edma0 2 1>;
|
||||
dma-names = "tx", "rx";
|
||||
|
||||
clocks = <&psc1 14>;
|
||||
};
|
||||
|
|
@ -1,50 +0,0 @@
|
|||
NXP Audio Mixer (AUDMIX).
|
||||
|
||||
The Audio Mixer is a on-chip functional module that allows mixing of two
|
||||
audio streams into a single audio stream. Audio Mixer has two input serial
|
||||
audio interfaces. These are driven by two Synchronous Audio interface
|
||||
modules (SAI). Each input serial interface carries 8 audio channels in its
|
||||
frame in TDM manner. Mixer mixes audio samples of corresponding channels
|
||||
from two interfaces into a single sample. Before mixing, audio samples of
|
||||
two inputs can be attenuated based on configuration. The output of the
|
||||
Audio Mixer is also a serial audio interface. Like input interfaces it has
|
||||
the same TDM frame format. This output is used to drive the serial DAC TDM
|
||||
interface of audio codec and also sent to the external pins along with the
|
||||
receive path of normal audio SAI module for readback by the CPU.
|
||||
|
||||
The output of Audio Mixer can be selected from any of the three streams
|
||||
- serial audio input 1
|
||||
- serial audio input 2
|
||||
- mixed audio
|
||||
|
||||
Mixing operation is independent of audio sample rate but the two audio
|
||||
input streams must have same audio sample rate with same number of channels
|
||||
in TDM frame to be eligible for mixing.
|
||||
|
||||
Device driver required properties:
|
||||
=================================
|
||||
- compatible : Compatible list, contains "fsl,imx8qm-audmix"
|
||||
|
||||
- reg : Offset and length of the register set for the device.
|
||||
|
||||
- clocks : Must contain an entry for each entry in clock-names.
|
||||
|
||||
- clock-names : Must include the "ipg" for register access.
|
||||
|
||||
- power-domains : Must contain the phandle to AUDMIX power domain node
|
||||
|
||||
- dais : Must contain a list of phandles to AUDMIX connected
|
||||
DAIs. The current implementation requires two phandles
|
||||
to SAI interfaces to be provided, the first SAI in the
|
||||
list being used to route the AUDMIX output.
|
||||
|
||||
Device driver configuration example:
|
||||
======================================
|
||||
audmix: audmix@59840000 {
|
||||
compatible = "fsl,imx8qm-audmix";
|
||||
reg = <0x0 0x59840000 0x0 0x10000>;
|
||||
clocks = <&clk IMX8QXP_AUD_AUDMIX_IPG>;
|
||||
clock-names = "ipg";
|
||||
power-domains = <&pd_audmix>;
|
||||
dais = <&sai4>, <&sai5>;
|
||||
};
|
||||
|
|
@ -0,0 +1,83 @@
|
|||
# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
|
||||
%YAML 1.2
|
||||
---
|
||||
$id: http://devicetree.org/schemas/sound/fsl,audmix.yaml#
|
||||
$schema: http://devicetree.org/meta-schemas/core.yaml#
|
||||
|
||||
title: NXP Audio Mixer (AUDMIX).
|
||||
|
||||
maintainers:
|
||||
- Shengjiu Wang <shengjiu.wang@nxp.com>
|
||||
- Frank Li <Frank.Li@nxp.com>
|
||||
|
||||
description: |
|
||||
The Audio Mixer is a on-chip functional module that allows mixing of two
|
||||
audio streams into a single audio stream. Audio Mixer has two input serial
|
||||
audio interfaces. These are driven by two Synchronous Audio interface
|
||||
modules (SAI). Each input serial interface carries 8 audio channels in its
|
||||
frame in TDM manner. Mixer mixes audio samples of corresponding channels
|
||||
from two interfaces into a single sample. Before mixing, audio samples of
|
||||
two inputs can be attenuated based on configuration. The output of the
|
||||
Audio Mixer is also a serial audio interface. Like input interfaces it has
|
||||
the same TDM frame format. This output is used to drive the serial DAC TDM
|
||||
interface of audio codec and also sent to the external pins along with the
|
||||
receive path of normal audio SAI module for readback by the CPU.
|
||||
|
||||
The output of Audio Mixer can be selected from any of the three streams
|
||||
- serial audio input 1
|
||||
- serial audio input 2
|
||||
- mixed audio
|
||||
|
||||
Mixing operation is independent of audio sample rate but the two audio
|
||||
input streams must have same audio sample rate with same number of channels
|
||||
in TDM frame to be eligible for mixing.
|
||||
|
||||
properties:
|
||||
compatible:
|
||||
const: fsl,imx8qm-audmix
|
||||
|
||||
reg:
|
||||
maxItems: 1
|
||||
|
||||
clocks:
|
||||
maxItems: 1
|
||||
|
||||
clock-names:
|
||||
items:
|
||||
- const: ipg
|
||||
|
||||
power-domains:
|
||||
maxItems: 1
|
||||
|
||||
dais:
|
||||
description: contain a list of phandles to AUDMIX connected DAIs.
|
||||
$ref: /schemas/types.yaml#/definitions/phandle-array
|
||||
minItems: 2
|
||||
items:
|
||||
- description: the AUDMIX output
|
||||
maxItems: 1
|
||||
- description: serial audio input 1
|
||||
maxItems: 1
|
||||
- description: serial audio input 2
|
||||
maxItems: 1
|
||||
|
||||
required:
|
||||
- compatible
|
||||
- reg
|
||||
- clocks
|
||||
- clock-names
|
||||
- power-domains
|
||||
- dais
|
||||
|
||||
unevaluatedProperties: false
|
||||
|
||||
examples:
|
||||
- |
|
||||
audmix@59840000 {
|
||||
compatible = "fsl,imx8qm-audmix";
|
||||
reg = <0x59840000 0x10000>;
|
||||
clocks = <&amix_lpcg 0>;
|
||||
clock-names = "ipg";
|
||||
power-domains = <&pd_audmix>;
|
||||
dais = <&sai4>, <&sai5>;
|
||||
};
|
||||
|
|
@ -1,68 +0,0 @@
|
|||
Freescale Enhanced Serial Audio Interface (ESAI) Controller
|
||||
|
||||
The Enhanced Serial Audio Interface (ESAI) provides a full-duplex serial port
|
||||
for serial communication with a variety of serial devices, including industry
|
||||
standard codecs, Sony/Phillips Digital Interface (S/PDIF) transceivers, and
|
||||
other DSPs. It has up to six transmitters and four receivers.
|
||||
|
||||
Required properties:
|
||||
|
||||
- compatible : Compatible list, should contain one of the following
|
||||
compatibles:
|
||||
"fsl,imx35-esai",
|
||||
"fsl,vf610-esai",
|
||||
"fsl,imx6ull-esai",
|
||||
"fsl,imx8qm-esai",
|
||||
|
||||
- reg : Offset and length of the register set for the device.
|
||||
|
||||
- interrupts : Contains the spdif interrupt.
|
||||
|
||||
- dmas : Generic dma devicetree binding as described in
|
||||
Documentation/devicetree/bindings/dma/dma.txt.
|
||||
|
||||
- dma-names : Two dmas have to be defined, "tx" and "rx".
|
||||
|
||||
- clocks : Contains an entry for each entry in clock-names.
|
||||
|
||||
- clock-names : Includes the following entries:
|
||||
"core" The core clock used to access registers
|
||||
"extal" The esai baud clock for esai controller used to
|
||||
derive HCK, SCK and FS.
|
||||
"fsys" The system clock derived from ahb clock used to
|
||||
derive HCK, SCK and FS.
|
||||
"spba" The spba clock is required when ESAI is placed as a
|
||||
bus slave of the Shared Peripheral Bus and when two
|
||||
or more bus masters (CPU, DMA or DSP) try to access
|
||||
it. This property is optional depending on the SoC
|
||||
design.
|
||||
|
||||
- fsl,fifo-depth : The number of elements in the transmit and receive
|
||||
FIFOs. This number is the maximum allowed value for
|
||||
TFCR[TFWM] or RFCR[RFWM].
|
||||
|
||||
- fsl,esai-synchronous: This is a boolean property. If present, indicating
|
||||
that ESAI would work in the synchronous mode, which
|
||||
means all the settings for Receiving would be
|
||||
duplicated from Transmission related registers.
|
||||
|
||||
Optional properties:
|
||||
|
||||
- big-endian : If this property is absent, the native endian mode
|
||||
will be in use as default, or the big endian mode
|
||||
will be in use for all the device registers.
|
||||
|
||||
Example:
|
||||
|
||||
esai: esai@2024000 {
|
||||
compatible = "fsl,imx35-esai";
|
||||
reg = <0x02024000 0x4000>;
|
||||
interrupts = <0 51 0x04>;
|
||||
clocks = <&clks 208>, <&clks 118>, <&clks 208>;
|
||||
clock-names = "core", "extal", "fsys";
|
||||
dmas = <&sdma 23 21 0>, <&sdma 24 21 0>;
|
||||
dma-names = "rx", "tx";
|
||||
fsl,fifo-depth = <128>;
|
||||
fsl,esai-synchronous;
|
||||
big-endian;
|
||||
};
|
||||
|
|
@ -0,0 +1,118 @@
|
|||
# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
|
||||
%YAML 1.2
|
||||
---
|
||||
$id: http://devicetree.org/schemas/sound/fsl,esai.yaml#
|
||||
$schema: http://devicetree.org/meta-schemas/core.yaml#
|
||||
|
||||
title: Freescale Enhanced Serial Audio Interface (ESAI) Controller
|
||||
|
||||
maintainers:
|
||||
- Shengjiu Wang <shengjiu.wang@nxp.com>
|
||||
- Frank Li <Frank.Li@nxp.com>
|
||||
|
||||
description:
|
||||
The Enhanced Serial Audio Interface (ESAI) provides a full-duplex serial port
|
||||
for serial communication with a variety of serial devices, including industry
|
||||
standard codecs, Sony/Phillips Digital Interface (S/PDIF) transceivers, and
|
||||
other DSPs. It has up to six transmitters and four receivers.
|
||||
|
||||
properties:
|
||||
compatible:
|
||||
enum:
|
||||
- fsl,imx35-esai
|
||||
- fsl,imx6ull-esai
|
||||
- fsl,imx8qm-esai
|
||||
- fsl,vf610-esai
|
||||
|
||||
reg:
|
||||
maxItems: 1
|
||||
|
||||
interrupts:
|
||||
maxItems: 1
|
||||
|
||||
clocks:
|
||||
minItems: 3
|
||||
items:
|
||||
- description:
|
||||
The core clock used to access registers.
|
||||
- description:
|
||||
The esai baud clock for esai controller used to
|
||||
derive HCK, SCK and FS.
|
||||
- description:
|
||||
The system clock derived from ahb clock used to
|
||||
derive HCK, SCK and FS.
|
||||
- description:
|
||||
The spba clock is required when ESAI is placed as a
|
||||
bus slave of the Shared Peripheral Bus and when two
|
||||
or more bus masters (CPU, DMA or DSP) try to access
|
||||
it. This property is optional depending on the SoC
|
||||
design.
|
||||
|
||||
clock-names:
|
||||
minItems: 3
|
||||
items:
|
||||
- const: core
|
||||
- const: extal
|
||||
- const: fsys
|
||||
- const: spba
|
||||
|
||||
dmas:
|
||||
minItems: 2
|
||||
maxItems: 2
|
||||
|
||||
dma-names:
|
||||
items:
|
||||
- const: rx
|
||||
- const: tx
|
||||
|
||||
fsl,fifo-depth:
|
||||
$ref: /schemas/types.yaml#/definitions/uint32
|
||||
default: 64
|
||||
description:
|
||||
The number of elements in the transmit and receive
|
||||
FIFOs. This number is the maximum allowed value for
|
||||
TFCR[TFWM] or RFCR[RFWM].
|
||||
|
||||
fsl,esai-synchronous:
|
||||
$ref: /schemas/types.yaml#/definitions/flag
|
||||
description:
|
||||
This is a boolean property. If present, indicating
|
||||
that ESAI would work in the synchronous mode, which
|
||||
means all the settings for Receiving would be
|
||||
duplicated from Transmission related registers.
|
||||
|
||||
big-endian:
|
||||
$ref: /schemas/types.yaml#/definitions/flag
|
||||
description:
|
||||
If this property is absent, the native endian mode
|
||||
will be in use as default, or the big endian mode
|
||||
will be in use for all the device registers.
|
||||
|
||||
required:
|
||||
- compatible
|
||||
- reg
|
||||
- interrupts
|
||||
- clocks
|
||||
- clock-names
|
||||
- dmas
|
||||
- dma-names
|
||||
|
||||
unevaluatedProperties: false
|
||||
|
||||
allOf:
|
||||
- $ref: dai-common.yaml#
|
||||
|
||||
examples:
|
||||
- |
|
||||
esai@2024000 {
|
||||
compatible = "fsl,imx35-esai";
|
||||
reg = <0x02024000 0x4000>;
|
||||
interrupts = <0 51 0x04>;
|
||||
clocks = <&clks 208>, <&clks 118>, <&clks 208>;
|
||||
clock-names = "core", "extal", "fsys";
|
||||
dmas = <&sdma 23 21 0>, <&sdma 24 21 0>;
|
||||
dma-names = "rx", "tx";
|
||||
fsl,fifo-depth = <128>;
|
||||
fsl,esai-synchronous;
|
||||
big-endian;
|
||||
};
|
||||
|
|
@ -74,6 +74,9 @@ properties:
|
|||
- const: asrck_f
|
||||
- const: spba
|
||||
|
||||
power-domains:
|
||||
maxItems: 1
|
||||
|
||||
fsl,asrc-rate:
|
||||
$ref: /schemas/types.yaml#/definitions/uint32
|
||||
description: The mutual sample rate used by DPCM Back Ends
|
||||
|
|
@ -131,6 +134,17 @@ allOf:
|
|||
properties:
|
||||
fsl,asrc-clk-map: false
|
||||
|
||||
- if:
|
||||
properties:
|
||||
compatible:
|
||||
contains:
|
||||
enum:
|
||||
- fsl,imx8qm-asrc
|
||||
- fsl,imx8qxp-asrc
|
||||
then:
|
||||
required:
|
||||
- power-domains
|
||||
|
||||
additionalProperties: false
|
||||
|
||||
examples:
|
||||
|
|
|
|||
|
|
@ -0,0 +1,66 @@
|
|||
# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
|
||||
%YAML 1.2
|
||||
---
|
||||
$id: http://devicetree.org/schemas/sound/fsl,imx-audio-spdif.yaml#
|
||||
$schema: http://devicetree.org/meta-schemas/core.yaml#
|
||||
|
||||
title: Freescale i.MX audio complex with S/PDIF transceiver
|
||||
|
||||
maintainers:
|
||||
- Shengjiu Wang <shengjiu.wang@nxp.com>
|
||||
|
||||
properties:
|
||||
compatible:
|
||||
oneOf:
|
||||
- items:
|
||||
- enum:
|
||||
- fsl,imx-sabreauto-spdif
|
||||
- fsl,imx6sx-sdb-spdif
|
||||
- const: fsl,imx-audio-spdif
|
||||
- enum:
|
||||
- fsl,imx-audio-spdif
|
||||
|
||||
model:
|
||||
$ref: /schemas/types.yaml#/definitions/string
|
||||
description: User specified audio sound card name
|
||||
|
||||
spdif-controller:
|
||||
$ref: /schemas/types.yaml#/definitions/phandle
|
||||
description: The phandle of the i.MX S/PDIF controller
|
||||
|
||||
spdif-out:
|
||||
type: boolean
|
||||
description:
|
||||
If present, the transmitting function of S/PDIF will be enabled,
|
||||
indicating there's a physical S/PDIF out connector or jack on the
|
||||
board or it's connecting to some other IP block, such as an HDMI
|
||||
encoder or display-controller.
|
||||
|
||||
spdif-in:
|
||||
type: boolean
|
||||
description:
|
||||
If present, the receiving function of S/PDIF will be enabled,
|
||||
indicating there is a physical S/PDIF in connector/jack on the board.
|
||||
|
||||
required:
|
||||
- compatible
|
||||
- model
|
||||
- spdif-controller
|
||||
|
||||
anyOf:
|
||||
- required:
|
||||
- spdif-in
|
||||
- required:
|
||||
- spdif-out
|
||||
|
||||
additionalProperties: false
|
||||
|
||||
examples:
|
||||
- |
|
||||
sound-spdif {
|
||||
compatible = "fsl,imx-audio-spdif";
|
||||
model = "imx-spdif";
|
||||
spdif-controller = <&spdif>;
|
||||
spdif-out;
|
||||
spdif-in;
|
||||
};
|
||||
|
|
@ -81,14 +81,12 @@ properties:
|
|||
|
||||
dmas:
|
||||
minItems: 1
|
||||
items:
|
||||
- description: DMA controller phandle and request line for RX
|
||||
- description: DMA controller phandle and request line for TX
|
||||
maxItems: 2
|
||||
|
||||
dma-names:
|
||||
minItems: 1
|
||||
items:
|
||||
- const: rx
|
||||
- enum: [ rx, tx ]
|
||||
- const: tx
|
||||
|
||||
interrupts:
|
||||
|
|
|
|||
|
|
@ -31,7 +31,10 @@ properties:
|
|||
maxItems: 1
|
||||
|
||||
interrupts:
|
||||
maxItems: 1
|
||||
minItems: 1
|
||||
items:
|
||||
- description: Combined or receive interrupt
|
||||
- description: Transmit interrupt
|
||||
|
||||
dmas:
|
||||
items:
|
||||
|
|
@ -86,6 +89,9 @@ properties:
|
|||
registers. Set this flag for HCDs with big endian descriptors and big
|
||||
endian registers.
|
||||
|
||||
power-domains:
|
||||
maxItems: 1
|
||||
|
||||
required:
|
||||
- compatible
|
||||
- reg
|
||||
|
|
@ -97,6 +103,33 @@ required:
|
|||
|
||||
additionalProperties: false
|
||||
|
||||
allOf:
|
||||
- if:
|
||||
properties:
|
||||
compatible:
|
||||
enum:
|
||||
- fsl,imx8qm-spdif
|
||||
- fsl,imx8qxp-spdif
|
||||
then:
|
||||
properties:
|
||||
interrupts:
|
||||
minItems: 2
|
||||
else:
|
||||
properties:
|
||||
interrupts:
|
||||
maxItems: 1
|
||||
|
||||
- if:
|
||||
properties:
|
||||
compatible:
|
||||
contains:
|
||||
enum:
|
||||
- fsl,imx8qm-spdif
|
||||
- fsl,imx8qxp-spdif
|
||||
then:
|
||||
required:
|
||||
- power-domains
|
||||
|
||||
examples:
|
||||
- |
|
||||
spdif@2004000 {
|
||||
|
|
|
|||
|
|
@ -1,87 +0,0 @@
|
|||
Freescale Synchronous Serial Interface
|
||||
|
||||
The SSI is a serial device that communicates with audio codecs. It can
|
||||
be programmed in AC97, I2S, left-justified, or right-justified modes.
|
||||
|
||||
Required properties:
|
||||
- compatible: Compatible list, should contain one of the following
|
||||
compatibles:
|
||||
fsl,mpc8610-ssi
|
||||
fsl,imx51-ssi
|
||||
fsl,imx35-ssi
|
||||
fsl,imx21-ssi
|
||||
- cell-index: The SSI, <0> = SSI1, <1> = SSI2, and so on.
|
||||
- reg: Offset and length of the register set for the device.
|
||||
- interrupts: <a b> where a is the interrupt number and b is a
|
||||
field that represents an encoding of the sense and
|
||||
level information for the interrupt. This should be
|
||||
encoded based on the information in section 2)
|
||||
depending on the type of interrupt controller you
|
||||
have.
|
||||
- fsl,fifo-depth: The number of elements in the transmit and receive FIFOs.
|
||||
This number is the maximum allowed value for SFCSR[TFWM0].
|
||||
- clocks: "ipg" - Required clock for the SSI unit
|
||||
"baud" - Required clock for SSI master mode. Otherwise this
|
||||
clock is not used
|
||||
|
||||
Required are also ac97 link bindings if ac97 is used. See
|
||||
Documentation/devicetree/bindings/sound/soc-ac97link.txt for the necessary
|
||||
bindings.
|
||||
|
||||
Optional properties:
|
||||
- codec-handle: Phandle to a 'codec' node that defines an audio
|
||||
codec connected to this SSI. This node is typically
|
||||
a child of an I2C or other control node.
|
||||
- fsl,fiq-stream-filter: Bool property. Disabled DMA and use FIQ instead to
|
||||
filter the codec stream. This is necessary for some boards
|
||||
where an incompatible codec is connected to this SSI, e.g.
|
||||
on pca100 and pcm043.
|
||||
- dmas: Generic dma devicetree binding as described in
|
||||
Documentation/devicetree/bindings/dma/dma.txt.
|
||||
- dma-names: Two dmas have to be defined, "tx" and "rx", if fsl,imx-fiq
|
||||
is not defined.
|
||||
- fsl,mode: The operating mode for the AC97 interface only.
|
||||
"ac97-slave" - AC97 mode, SSI is clock slave
|
||||
"ac97-master" - AC97 mode, SSI is clock master
|
||||
- fsl,ssi-asynchronous:
|
||||
If specified, the SSI is to be programmed in asynchronous
|
||||
mode. In this mode, pins SRCK, STCK, SRFS, and STFS must
|
||||
all be connected to valid signals. In synchronous mode,
|
||||
SRCK and SRFS are ignored. Asynchronous mode allows
|
||||
playback and capture to use different sample sizes and
|
||||
sample rates. Some drivers may require that SRCK and STCK
|
||||
be connected together, and SRFS and STFS be connected
|
||||
together. This would still allow different sample sizes,
|
||||
but not different sample rates.
|
||||
- fsl,playback-dma: Phandle to a node for the DMA channel to use for
|
||||
playback of audio. This is typically dictated by SOC
|
||||
design. See the notes below.
|
||||
Only used on Power Architecture.
|
||||
- fsl,capture-dma: Phandle to a node for the DMA channel to use for
|
||||
capture (recording) of audio. This is typically dictated
|
||||
by SOC design. See the notes below.
|
||||
Only used on Power Architecture.
|
||||
|
||||
Child 'codec' node required properties:
|
||||
- compatible: Compatible list, contains the name of the codec
|
||||
|
||||
Child 'codec' node optional properties:
|
||||
- clock-frequency: The frequency of the input clock, which typically comes
|
||||
from an on-board dedicated oscillator.
|
||||
|
||||
Notes on fsl,playback-dma and fsl,capture-dma:
|
||||
|
||||
On SOCs that have an SSI, specific DMA channels are hard-wired for playback
|
||||
and capture. On the MPC8610, for example, SSI1 must use DMA channel 0 for
|
||||
playback and DMA channel 1 for capture. SSI2 must use DMA channel 2 for
|
||||
playback and DMA channel 3 for capture. The developer can choose which
|
||||
DMA controller to use, but the channels themselves are hard-wired. The
|
||||
purpose of these two properties is to represent this hardware design.
|
||||
|
||||
The device tree nodes for the DMA channels that are referenced by
|
||||
"fsl,playback-dma" and "fsl,capture-dma" must be marked as compatible with
|
||||
"fsl,ssi-dma-channel". The SOC-specific compatible string (e.g.
|
||||
"fsl,mpc8610-dma-channel") can remain. If these nodes are left as
|
||||
"fsl,elo-dma-channel" or "fsl,eloplus-dma-channel", then the generic Elo DMA
|
||||
drivers (fsldma) will attempt to use them, and it will conflict with the
|
||||
sound drivers.
|
||||
|
|
@ -0,0 +1,194 @@
|
|||
# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
|
||||
%YAML 1.2
|
||||
---
|
||||
$id: http://devicetree.org/schemas/sound/fsl,ssi.yaml#
|
||||
$schema: http://devicetree.org/meta-schemas/core.yaml#
|
||||
|
||||
title: Freescale Synchronous Serial Interface
|
||||
|
||||
maintainers:
|
||||
- Shengjiu Wang <shengjiu.wang@nxp.com>
|
||||
|
||||
description:
|
||||
Notes on fsl,playback-dma and fsl,capture-dma
|
||||
On SOCs that have an SSI, specific DMA channels are hard-wired for playback
|
||||
and capture. On the MPC8610, for example, SSI1 must use DMA channel 0 for
|
||||
playback and DMA channel 1 for capture. SSI2 must use DMA channel 2 for
|
||||
playback and DMA channel 3 for capture. The developer can choose which
|
||||
DMA controller to use, but the channels themselves are hard-wired. The
|
||||
purpose of these two properties is to represent this hardware design.
|
||||
|
||||
The device tree nodes for the DMA channels that are referenced by
|
||||
"fsl,playback-dma" and "fsl,capture-dma" must be marked as compatible with
|
||||
"fsl,ssi-dma-channel". The SOC-specific compatible string (e.g.
|
||||
"fsl,mpc8610-dma-channel") can remain. If these nodes are left as
|
||||
"fsl,elo-dma-channel" or "fsl,eloplus-dma-channel", then the generic Elo DMA
|
||||
drivers (fsldma) will attempt to use them, and it will conflict with the
|
||||
sound drivers.
|
||||
|
||||
properties:
|
||||
compatible:
|
||||
oneOf:
|
||||
- items:
|
||||
- enum:
|
||||
- fsl,imx50-ssi
|
||||
- fsl,imx53-ssi
|
||||
- const: fsl,imx51-ssi
|
||||
- const: fsl,imx21-ssi
|
||||
- items:
|
||||
- enum:
|
||||
- fsl,imx25-ssi
|
||||
- fsl,imx27-ssi
|
||||
- fsl,imx35-ssi
|
||||
- fsl,imx51-ssi
|
||||
- const: fsl,imx21-ssi
|
||||
- items:
|
||||
- enum:
|
||||
- fsl,imx6q-ssi
|
||||
- fsl,imx6sl-ssi
|
||||
- fsl,imx6sx-ssi
|
||||
- const: fsl,imx51-ssi
|
||||
- items:
|
||||
- const: fsl,imx21-ssi
|
||||
- items:
|
||||
- const: fsl,mpc8610-ssi
|
||||
|
||||
reg:
|
||||
maxItems: 1
|
||||
|
||||
interrupts:
|
||||
maxItems: 1
|
||||
|
||||
clocks:
|
||||
items:
|
||||
- description: The ipg clock for register access
|
||||
- description: clock for SSI master mode
|
||||
minItems: 1
|
||||
|
||||
clock-names:
|
||||
items:
|
||||
- const: ipg
|
||||
- const: baud
|
||||
minItems: 1
|
||||
|
||||
dmas:
|
||||
oneOf:
|
||||
- items:
|
||||
- description: DMA controller phandle and request line for RX
|
||||
- description: DMA controller phandle and request line for TX
|
||||
- items:
|
||||
- description: DMA controller phandle and request line for RX0
|
||||
- description: DMA controller phandle and request line for TX0
|
||||
- description: DMA controller phandle and request line for RX1
|
||||
- description: DMA controller phandle and request line for TX1
|
||||
|
||||
dma-names:
|
||||
oneOf:
|
||||
- items:
|
||||
- const: rx
|
||||
- const: tx
|
||||
- items:
|
||||
- const: rx0
|
||||
- const: tx0
|
||||
- const: rx1
|
||||
- const: tx1
|
||||
|
||||
"#sound-dai-cells":
|
||||
const: 0
|
||||
description: optional, some dts node didn't add it.
|
||||
|
||||
cell-index:
|
||||
$ref: /schemas/types.yaml#/definitions/uint32
|
||||
enum: [0, 1, 2]
|
||||
description: The SSI index
|
||||
|
||||
ac97-gpios:
|
||||
$ref: /schemas/types.yaml#/definitions/phandle-array
|
||||
description: Please refer to soc-ac97link.txt
|
||||
|
||||
codec-handle:
|
||||
$ref: /schemas/types.yaml#/definitions/phandle
|
||||
description:
|
||||
Phandle to a 'codec' node that defines an audio
|
||||
codec connected to this SSI. This node is typically
|
||||
a child of an I2C or other control node.
|
||||
|
||||
fsl,fifo-depth:
|
||||
$ref: /schemas/types.yaml#/definitions/uint32
|
||||
description:
|
||||
The number of elements in the transmit and receive FIFOs.
|
||||
This number is the maximum allowed value for SFCSR[TFWM0].
|
||||
enum: [8, 15]
|
||||
|
||||
fsl,fiq-stream-filter:
|
||||
type: boolean
|
||||
description:
|
||||
Disabled DMA and use FIQ instead to filter the codec stream.
|
||||
This is necessary for some boards where an incompatible codec
|
||||
is connected to this SSI, e.g. on pca100 and pcm043.
|
||||
|
||||
fsl,mode:
|
||||
$ref: /schemas/types.yaml#/definitions/string
|
||||
enum: [ ac97-slave, ac97-master, i2s-slave, i2s-master,
|
||||
lj-slave, lj-master, rj-slave, rj-master ]
|
||||
description: |
|
||||
"ac97-slave" - AC97 mode, SSI is clock slave
|
||||
"ac97-master" - AC97 mode, SSI is clock master
|
||||
"i2s-slave" - I2S mode, SSI is clock slave
|
||||
"i2s-master" - I2S mode, SSI is clock master
|
||||
"lj-slave" - Left justified mode, SSI is clock slave
|
||||
"lj-master" - Left justified mode, SSI is clock master
|
||||
"rj-slave" - Right justified mode, SSI is clock slave
|
||||
"rj-master" - Right justified mode, SSI is clock master
|
||||
|
||||
fsl,ssi-asynchronous:
|
||||
type: boolean
|
||||
description: If specified, the SSI is to be programmed in asynchronous
|
||||
mode. In this mode, pins SRCK, STCK, SRFS, and STFS must
|
||||
all be connected to valid signals. In synchronous mode,
|
||||
SRCK and SRFS are ignored. Asynchronous mode allows
|
||||
playback and capture to use different sample sizes and
|
||||
sample rates. Some drivers may require that SRCK and STCK
|
||||
be connected together, and SRFS and STFS be connected
|
||||
together. This would still allow different sample sizes,
|
||||
but not different sample rates.
|
||||
|
||||
fsl,playback-dma:
|
||||
$ref: /schemas/types.yaml#/definitions/phandle
|
||||
description: Phandle to a node for the DMA channel to use for
|
||||
playback of audio. This is typically dictated by SOC
|
||||
design. Only used on Power Architecture.
|
||||
|
||||
fsl,capture-dma:
|
||||
$ref: /schemas/types.yaml#/definitions/phandle
|
||||
description: Phandle to a node for the DMA channel to use for
|
||||
capture (recording) of audio. This is typically dictated
|
||||
by SOC design. Only used on Power Architecture.
|
||||
|
||||
required:
|
||||
- compatible
|
||||
- reg
|
||||
- interrupts
|
||||
- fsl,fifo-depth
|
||||
|
||||
allOf:
|
||||
- $ref: dai-common.yaml#
|
||||
|
||||
unevaluatedProperties: false
|
||||
|
||||
examples:
|
||||
- |
|
||||
#include <dt-bindings/interrupt-controller/arm-gic.h>
|
||||
#include <dt-bindings/clock/imx6qdl-clock.h>
|
||||
ssi@2028000 {
|
||||
compatible = "fsl,imx6q-ssi", "fsl,imx51-ssi";
|
||||
reg = <0x02028000 0x4000>;
|
||||
interrupts = <GIC_SPI 46 IRQ_TYPE_LEVEL_HIGH>;
|
||||
clocks = <&clks IMX6QDL_CLK_SSI1_IPG>,
|
||||
<&clks IMX6QDL_CLK_SSI1>;
|
||||
clock-names = "ipg", "baud";
|
||||
dmas = <&sdma 37 1 0>, <&sdma 38 1 0>;
|
||||
dma-names = "rx", "tx";
|
||||
#sound-dai-cells = <0>;
|
||||
fsl,fifo-depth = <15>;
|
||||
};
|
||||
|
|
@ -1,117 +0,0 @@
|
|||
Freescale Generic ASoC Sound Card with ASRC support
|
||||
|
||||
The Freescale Generic ASoC Sound Card can be used, ideally, for all Freescale
|
||||
SoCs connecting with external CODECs.
|
||||
|
||||
The idea of this generic sound card is a bit like ASoC Simple Card. However,
|
||||
for Freescale SoCs (especially those released in recent years), most of them
|
||||
have ASRC (Documentation/devicetree/bindings/sound/fsl,asrc.txt) inside. And
|
||||
this is a specific feature that might be painstakingly controlled and merged
|
||||
into the Simple Card.
|
||||
|
||||
So having this generic sound card allows all Freescale SoC users to benefit
|
||||
from the simplification of a new card support and the capability of the wide
|
||||
sample rates support through ASRC.
|
||||
|
||||
Note: The card is initially designed for those sound cards who use AC'97, I2S
|
||||
and PCM DAI formats. However, it'll be also possible to support those non
|
||||
AC'97/I2S/PCM type sound cards, such as S/PDIF audio and HDMI audio, as
|
||||
long as the driver has been properly upgraded.
|
||||
|
||||
|
||||
The compatible list for this generic sound card currently:
|
||||
"fsl,imx-audio-ac97"
|
||||
|
||||
"fsl,imx-audio-cs42888"
|
||||
|
||||
"fsl,imx-audio-cs427x"
|
||||
(compatible with CS4271 and CS4272)
|
||||
|
||||
"fsl,imx-audio-wm8962"
|
||||
|
||||
"fsl,imx-audio-sgtl5000"
|
||||
(compatible with Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt)
|
||||
|
||||
"fsl,imx-audio-wm8960"
|
||||
|
||||
"fsl,imx-audio-mqs"
|
||||
|
||||
"fsl,imx-audio-wm8524"
|
||||
|
||||
"fsl,imx-audio-tlv320aic32x4"
|
||||
|
||||
"fsl,imx-audio-tlv320aic31xx"
|
||||
|
||||
"fsl,imx-audio-si476x"
|
||||
|
||||
"fsl,imx-audio-wm8958"
|
||||
|
||||
"fsl,imx-audio-nau8822"
|
||||
|
||||
Required properties:
|
||||
|
||||
- compatible : Contains one of entries in the compatible list.
|
||||
|
||||
- model : The user-visible name of this sound complex
|
||||
|
||||
- audio-cpu : The phandle of an CPU DAI controller
|
||||
|
||||
- audio-codec : The phandle of an audio codec
|
||||
|
||||
Optional properties:
|
||||
|
||||
- audio-asrc : The phandle of ASRC. It can be absent if there's no
|
||||
need to add ASRC support via DPCM.
|
||||
|
||||
- audio-routing : A list of the connections between audio components.
|
||||
Each entry is a pair of strings, the first being the
|
||||
connection's sink, the second being the connection's
|
||||
source. There're a few pre-designed board connectors:
|
||||
* Line Out Jack
|
||||
* Line In Jack
|
||||
* Headphone Jack
|
||||
* Mic Jack
|
||||
* Ext Spk
|
||||
* AMIC (stands for Analog Microphone Jack)
|
||||
* DMIC (stands for Digital Microphone Jack)
|
||||
|
||||
Note: The "Mic Jack" and "AMIC" are redundant while
|
||||
coexisting in order to support the old bindings
|
||||
of wm8962 and sgtl5000.
|
||||
|
||||
- hp-det-gpio : The GPIO that detect headphones are plugged in
|
||||
- mic-det-gpio : The GPIO that detect microphones are plugged in
|
||||
- bitclock-master : Indicates dai-link bit clock master; for details see simple-card.yaml.
|
||||
- frame-master : Indicates dai-link frame master; for details see simple-card.yaml.
|
||||
- dai-format : audio format, for details see simple-card.yaml.
|
||||
- frame-inversion : dai-link uses frame clock inversion, for details see simple-card.yaml.
|
||||
- bitclock-inversion : dai-link uses bit clock inversion, for details see simple-card.yaml.
|
||||
- mclk-id : main clock id, specific for each card configuration.
|
||||
|
||||
Optional unless SSI is selected as a CPU DAI:
|
||||
|
||||
- mux-int-port : The internal port of the i.MX audio muxer (AUDMUX)
|
||||
|
||||
- mux-ext-port : The external port of the i.MX audio muxer
|
||||
|
||||
Example:
|
||||
sound-cs42888 {
|
||||
compatible = "fsl,imx-audio-cs42888";
|
||||
model = "cs42888-audio";
|
||||
audio-cpu = <&esai>;
|
||||
audio-asrc = <&asrc>;
|
||||
audio-codec = <&cs42888>;
|
||||
audio-routing =
|
||||
"Line Out Jack", "AOUT1L",
|
||||
"Line Out Jack", "AOUT1R",
|
||||
"Line Out Jack", "AOUT2L",
|
||||
"Line Out Jack", "AOUT2R",
|
||||
"Line Out Jack", "AOUT3L",
|
||||
"Line Out Jack", "AOUT3R",
|
||||
"Line Out Jack", "AOUT4L",
|
||||
"Line Out Jack", "AOUT4R",
|
||||
"AIN1L", "Line In Jack",
|
||||
"AIN1R", "Line In Jack",
|
||||
"AIN2L", "Line In Jack",
|
||||
"AIN2R", "Line In Jack";
|
||||
};
|
||||
|
|
@ -0,0 +1,197 @@
|
|||
# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
|
||||
%YAML 1.2
|
||||
---
|
||||
$id: http://devicetree.org/schemas/sound/fsl-asoc-card.yaml#
|
||||
$schema: http://devicetree.org/meta-schemas/core.yaml#
|
||||
|
||||
title: Freescale Generic ASoC Sound Card with ASRC support
|
||||
|
||||
description:
|
||||
The Freescale Generic ASoC Sound Card can be used, ideally,
|
||||
for all Freescale SoCs connecting with external CODECs.
|
||||
|
||||
The idea of this generic sound card is a bit like ASoC Simple Card.
|
||||
However, for Freescale SoCs (especially those released in recent years),
|
||||
most of them have ASRC inside. And this is a specific feature that might
|
||||
be painstakingly controlled and merged into the Simple Card.
|
||||
|
||||
So having this generic sound card allows all Freescale SoC users to
|
||||
benefit from the simplification of a new card support and the capability
|
||||
of the wide sample rates support through ASRC.
|
||||
|
||||
Note, The card is initially designed for those sound cards who use AC'97, I2S
|
||||
and PCM DAI formats. However, it'll be also possible to support those non
|
||||
AC'97/I2S/PCM type sound cards, such as S/PDIF audio and HDMI audio, as
|
||||
long as the driver has been properly upgraded.
|
||||
|
||||
maintainers:
|
||||
- Shengjiu Wang <shengjiu.wang@nxp.com>
|
||||
|
||||
properties:
|
||||
compatible:
|
||||
oneOf:
|
||||
- items:
|
||||
- enum:
|
||||
- fsl,imx-sgtl5000
|
||||
- fsl,imx25-pdk-sgtl5000
|
||||
- fsl,imx53-cpuvo-sgtl5000
|
||||
- fsl,imx51-babbage-sgtl5000
|
||||
- fsl,imx53-m53evk-sgtl5000
|
||||
- fsl,imx53-qsb-sgtl5000
|
||||
- fsl,imx53-voipac-sgtl5000
|
||||
- fsl,imx6-armadeus-sgtl5000
|
||||
- fsl,imx6-rex-sgtl5000
|
||||
- fsl,imx6-sabreauto-cs42888
|
||||
- fsl,imx6-wandboard-sgtl5000
|
||||
- fsl,imx6dl-nit6xlite-sgtl5000
|
||||
- fsl,imx6q-ba16-sgtl5000
|
||||
- fsl,imx6q-nitrogen6_max-sgtl5000
|
||||
- fsl,imx6q-nitrogen6_som2-sgtl5000
|
||||
- fsl,imx6q-nitrogen6x-sgtl5000
|
||||
- fsl,imx6q-sabrelite-sgtl5000
|
||||
- fsl,imx6q-sabresd-wm8962
|
||||
- fsl,imx6q-udoo-ac97
|
||||
- fsl,imx6q-ventana-sgtl5000
|
||||
- fsl,imx6sl-evk-wm8962
|
||||
- fsl,imx6sx-sdb-mqs
|
||||
- fsl,imx6sx-sdb-wm8962
|
||||
- fsl,imx7d-evk-wm8960
|
||||
- karo,tx53-audio-sgtl5000
|
||||
- tq,imx53-mba53-sgtl5000
|
||||
- enum:
|
||||
- fsl,imx-audio-ac97
|
||||
- fsl,imx-audio-cs42888
|
||||
- fsl,imx-audio-mqs
|
||||
- fsl,imx-audio-sgtl5000
|
||||
- fsl,imx-audio-wm8960
|
||||
- fsl,imx-audio-wm8962
|
||||
- items:
|
||||
- enum:
|
||||
- fsl,imx-audio-ac97
|
||||
- fsl,imx-audio-cs42888
|
||||
- fsl,imx-audio-cs427x
|
||||
- fsl,imx-audio-mqs
|
||||
- fsl,imx-audio-nau8822
|
||||
- fsl,imx-audio-sgtl5000
|
||||
- fsl,imx-audio-si476x
|
||||
- fsl,imx-audio-tlv320aic31xx
|
||||
- fsl,imx-audio-tlv320aic32x4
|
||||
- fsl,imx-audio-wm8524
|
||||
- fsl,imx-audio-wm8904
|
||||
- fsl,imx-audio-wm8960
|
||||
- fsl,imx-audio-wm8962
|
||||
- fsl,imx-audio-wm8958
|
||||
|
||||
model:
|
||||
$ref: /schemas/types.yaml#/definitions/string
|
||||
description: The user-visible name of this sound complex
|
||||
|
||||
audio-asrc:
|
||||
$ref: /schemas/types.yaml#/definitions/phandle
|
||||
description:
|
||||
The phandle of ASRC. It can be absent if there's no
|
||||
need to add ASRC support via DPCM.
|
||||
|
||||
audio-codec:
|
||||
$ref: /schemas/types.yaml#/definitions/phandle
|
||||
description: The phandle of an audio codec
|
||||
|
||||
audio-cpu:
|
||||
$ref: /schemas/types.yaml#/definitions/phandle
|
||||
description: The phandle of an CPU DAI controller
|
||||
|
||||
audio-routing:
|
||||
$ref: /schemas/types.yaml#/definitions/non-unique-string-array
|
||||
description:
|
||||
A list of the connections between audio components. Each entry is a
|
||||
pair of strings, the first being the connection's sink, the second
|
||||
being the connection's source. There're a few pre-designed board
|
||||
connectors. "AMIC" stands for Analog Microphone Jack.
|
||||
"DMIC" stands for Digital Microphone Jack. The "Mic Jack" and "AMIC"
|
||||
are redundant while coexisting in order to support the old bindings
|
||||
of wm8962 and sgtl5000.
|
||||
|
||||
hp-det-gpio:
|
||||
deprecated: true
|
||||
maxItems: 1
|
||||
description: The GPIO that detect headphones are plugged in
|
||||
|
||||
hp-det-gpios:
|
||||
maxItems: 1
|
||||
description: The GPIO that detect headphones are plugged in
|
||||
|
||||
mic-det-gpio:
|
||||
deprecated: true
|
||||
maxItems: 1
|
||||
description: The GPIO that detect microphones are plugged in
|
||||
|
||||
mic-det-gpios:
|
||||
maxItems: 1
|
||||
description: The GPIO that detect microphones are plugged in
|
||||
|
||||
bitclock-master:
|
||||
$ref: simple-card.yaml#/definitions/bitclock-master
|
||||
description: Indicates dai-link bit clock master.
|
||||
|
||||
frame-master:
|
||||
$ref: simple-card.yaml#/definitions/frame-master
|
||||
description: Indicates dai-link frame master.
|
||||
|
||||
format:
|
||||
$ref: simple-card.yaml#/definitions/format
|
||||
description: audio format.
|
||||
|
||||
frame-inversion:
|
||||
$ref: simple-card.yaml#/definitions/frame-inversion
|
||||
description: dai-link uses frame clock inversion.
|
||||
|
||||
bitclock-inversion:
|
||||
$ref: simple-card.yaml#/definitions/bitclock-inversion
|
||||
description: dai-link uses bit clock inversion.
|
||||
|
||||
mclk-id:
|
||||
$ref: /schemas/types.yaml#/definitions/uint32
|
||||
description: main clock id, specific for each card configuration.
|
||||
|
||||
mux-int-port:
|
||||
$ref: /schemas/types.yaml#/definitions/uint32
|
||||
enum: [1, 2, 7]
|
||||
description: The internal port of the i.MX audio muxer (AUDMUX)
|
||||
|
||||
mux-ext-port:
|
||||
$ref: /schemas/types.yaml#/definitions/uint32
|
||||
enum: [3, 4, 5, 6]
|
||||
description: The external port of the i.MX audio muxer
|
||||
|
||||
ssi-controller:
|
||||
$ref: /schemas/types.yaml#/definitions/phandle
|
||||
description: The phandle of an CPU DAI controller
|
||||
|
||||
required:
|
||||
- compatible
|
||||
- model
|
||||
|
||||
unevaluatedProperties: false
|
||||
|
||||
examples:
|
||||
- |
|
||||
sound-cs42888 {
|
||||
compatible = "fsl,imx-audio-cs42888";
|
||||
model = "cs42888-audio";
|
||||
audio-cpu = <&esai>;
|
||||
audio-asrc = <&asrc>;
|
||||
audio-codec = <&cs42888>;
|
||||
audio-routing =
|
||||
"Line Out Jack", "AOUT1L",
|
||||
"Line Out Jack", "AOUT1R",
|
||||
"Line Out Jack", "AOUT2L",
|
||||
"Line Out Jack", "AOUT2R",
|
||||
"Line Out Jack", "AOUT3L",
|
||||
"Line Out Jack", "AOUT3R",
|
||||
"Line Out Jack", "AOUT4L",
|
||||
"Line Out Jack", "AOUT4R",
|
||||
"AIN1L", "Line In Jack",
|
||||
"AIN1R", "Line In Jack",
|
||||
"AIN2L", "Line In Jack",
|
||||
"AIN2R", "Line In Jack";
|
||||
};
|
||||
|
|
@ -1,36 +0,0 @@
|
|||
Freescale i.MX audio complex with S/PDIF transceiver
|
||||
|
||||
Required properties:
|
||||
|
||||
- compatible : "fsl,imx-audio-spdif"
|
||||
|
||||
- model : The user-visible name of this sound complex
|
||||
|
||||
- spdif-controller : The phandle of the i.MX S/PDIF controller
|
||||
|
||||
|
||||
Optional properties:
|
||||
|
||||
- spdif-out : This is a boolean property. If present, the
|
||||
transmitting function of S/PDIF will be enabled,
|
||||
indicating there's a physical S/PDIF out connector
|
||||
or jack on the board or it's connecting to some
|
||||
other IP block, such as an HDMI encoder or
|
||||
display-controller.
|
||||
|
||||
- spdif-in : This is a boolean property. If present, the receiving
|
||||
function of S/PDIF will be enabled, indicating there
|
||||
is a physical S/PDIF in connector/jack on the board.
|
||||
|
||||
* Note: At least one of these two properties should be set in the DT binding.
|
||||
|
||||
|
||||
Example:
|
||||
|
||||
sound-spdif {
|
||||
compatible = "fsl,imx-audio-spdif";
|
||||
model = "imx-spdif";
|
||||
spdif-controller = <&spdif>;
|
||||
spdif-out;
|
||||
spdif-in;
|
||||
};
|
||||
|
|
@ -0,0 +1,54 @@
|
|||
# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
|
||||
%YAML 1.2
|
||||
---
|
||||
$id: http://devicetree.org/schemas/sound/mediatek,mt2701-wm8960.yaml#
|
||||
$schema: http://devicetree.org/meta-schemas/core.yaml#
|
||||
|
||||
title: MediaTek MT2701 with WM8960 CODEC
|
||||
|
||||
maintainers:
|
||||
- Kartik Agarwala <agarwala.kartik@gmail.com>
|
||||
|
||||
properties:
|
||||
compatible:
|
||||
const: mediatek,mt2701-wm8960-machine
|
||||
|
||||
mediatek,platform:
|
||||
$ref: /schemas/types.yaml#/definitions/phandle
|
||||
description: The phandle of MT2701 ASoC platform.
|
||||
|
||||
audio-routing:
|
||||
$ref: /schemas/types.yaml#/definitions/non-unique-string-array
|
||||
description:
|
||||
A list of the connections between audio components. Each entry is a
|
||||
pair of strings, the first being the connection's sink, the second
|
||||
being the connection's source.
|
||||
|
||||
mediatek,audio-codec:
|
||||
$ref: /schemas/types.yaml#/definitions/phandle
|
||||
description: The phandle of the WM8960 audio codec.
|
||||
|
||||
unevaluatedProperties: false
|
||||
|
||||
required:
|
||||
- compatible
|
||||
- mediatek,platform
|
||||
- audio-routing
|
||||
- mediatek,audio-codec
|
||||
- pinctrl-names
|
||||
- pinctrl-0
|
||||
|
||||
examples:
|
||||
- |
|
||||
sound {
|
||||
compatible = "mediatek,mt2701-wm8960-machine";
|
||||
mediatek,platform = <&afe>;
|
||||
audio-routing =
|
||||
"Headphone", "HP_L",
|
||||
"Headphone", "HP_R",
|
||||
"LINPUT1", "AMIC",
|
||||
"RINPUT1", "AMIC";
|
||||
mediatek,audio-codec = <&wm8960>;
|
||||
pinctrl-names = "default";
|
||||
pinctrl-0 = <&aud_pins_default>;
|
||||
};
|
||||
|
|
@ -1,24 +0,0 @@
|
|||
MT2701 with WM8960 CODEC
|
||||
|
||||
Required properties:
|
||||
- compatible: "mediatek,mt2701-wm8960-machine"
|
||||
- mediatek,platform: the phandle of MT2701 ASoC platform
|
||||
- audio-routing: a list of the connections between audio
|
||||
- mediatek,audio-codec: the phandles of wm8960 codec
|
||||
- pinctrl-names: Should contain only one value - "default"
|
||||
- pinctrl-0: Should specify pin control groups used for this controller.
|
||||
|
||||
Example:
|
||||
|
||||
sound:sound {
|
||||
compatible = "mediatek,mt2701-wm8960-machine";
|
||||
mediatek,platform = <&afe>;
|
||||
audio-routing =
|
||||
"Headphone", "HP_L",
|
||||
"Headphone", "HP_R",
|
||||
"LINPUT1", "AMIC",
|
||||
"RINPUT1", "AMIC";
|
||||
mediatek,audio-codec = <&wm8960>;
|
||||
pinctrl-names = "default";
|
||||
pinctrl-0 = <&aud_pins_default>;
|
||||
};
|
||||
|
|
@ -12,17 +12,46 @@ maintainers:
|
|||
description:
|
||||
This binding describes the MT8186 sound card.
|
||||
|
||||
allOf:
|
||||
- $ref: sound-card-common.yaml#
|
||||
|
||||
properties:
|
||||
compatible:
|
||||
enum:
|
||||
- mediatek,mt8186-mt6366-da7219-max98357-sound
|
||||
|
||||
audio-routing:
|
||||
$ref: /schemas/types.yaml#/definitions/non-unique-string-array
|
||||
description:
|
||||
A list of the connections between audio components. Each entry is a
|
||||
pair of strings, the first being the connection's sink, the second
|
||||
being the connection's source.
|
||||
Valid names could be the input or output widgets of audio components,
|
||||
power supplies, MicBias of codec and the software switch.
|
||||
minItems: 2
|
||||
items:
|
||||
enum:
|
||||
# Sinks
|
||||
- HDMI1
|
||||
- Headphones
|
||||
- Line Out
|
||||
- MIC
|
||||
- Speakers
|
||||
|
||||
# Sources
|
||||
- Headset Mic
|
||||
- HPL
|
||||
- HPR
|
||||
- Speaker
|
||||
- TX
|
||||
|
||||
mediatek,platform:
|
||||
$ref: /schemas/types.yaml#/definitions/phandle
|
||||
description: The phandle of MT8186 ASoC platform.
|
||||
|
||||
headset-codec:
|
||||
type: object
|
||||
deprecated: true
|
||||
additionalProperties: false
|
||||
properties:
|
||||
sound-dai:
|
||||
|
|
@ -32,6 +61,7 @@ properties:
|
|||
|
||||
playback-codecs:
|
||||
type: object
|
||||
deprecated: true
|
||||
additionalProperties: false
|
||||
properties:
|
||||
sound-dai:
|
||||
|
|
@ -53,32 +83,115 @@ properties:
|
|||
A list of the desired dai-links in the sound card. Each entry is a
|
||||
name defined in the machine driver.
|
||||
|
||||
additionalProperties: false
|
||||
patternProperties:
|
||||
".*-dai-link$":
|
||||
type: object
|
||||
additionalProperties: false
|
||||
description:
|
||||
Container for dai-link level properties and CODEC sub-nodes.
|
||||
|
||||
properties:
|
||||
link-name:
|
||||
description: Indicates dai-link name and PCM stream name
|
||||
items:
|
||||
enum:
|
||||
- I2S0
|
||||
- I2S1
|
||||
- I2S2
|
||||
- I2S3
|
||||
|
||||
codec:
|
||||
description: Holds subnode which indicates codec dai.
|
||||
type: object
|
||||
additionalProperties: false
|
||||
properties:
|
||||
sound-dai:
|
||||
minItems: 1
|
||||
maxItems: 2
|
||||
required:
|
||||
- sound-dai
|
||||
|
||||
dai-format:
|
||||
description: audio format
|
||||
items:
|
||||
enum:
|
||||
- i2s
|
||||
- right_j
|
||||
- left_j
|
||||
- dsp_a
|
||||
- dsp_b
|
||||
|
||||
mediatek,clk-provider:
|
||||
$ref: /schemas/types.yaml#/definitions/string
|
||||
description: Indicates dai-link clock master.
|
||||
items:
|
||||
enum:
|
||||
- cpu
|
||||
- codec
|
||||
|
||||
required:
|
||||
- link-name
|
||||
|
||||
unevaluatedProperties: false
|
||||
|
||||
required:
|
||||
- compatible
|
||||
- mediatek,platform
|
||||
- headset-codec
|
||||
- playback-codecs
|
||||
|
||||
# Disallow legacy properties if xxx-dai-link nodes are specified
|
||||
if:
|
||||
not:
|
||||
patternProperties:
|
||||
".*-dai-link$": false
|
||||
then:
|
||||
properties:
|
||||
headset-codec: false
|
||||
speaker-codecs: false
|
||||
|
||||
examples:
|
||||
- |
|
||||
|
||||
sound: mt8186-sound {
|
||||
compatible = "mediatek,mt8186-mt6366-da7219-max98357-sound";
|
||||
mediatek,platform = <&afe>;
|
||||
model = "mt8186_da7219_m98357";
|
||||
pinctrl-names = "aud_clk_mosi_off",
|
||||
"aud_clk_mosi_on";
|
||||
pinctrl-0 = <&aud_clk_mosi_off>;
|
||||
pinctrl-1 = <&aud_clk_mosi_on>;
|
||||
mediatek,platform = <&afe>;
|
||||
|
||||
headset-codec {
|
||||
sound-dai = <&da7219>;
|
||||
audio-routing =
|
||||
"Headphones", "HPL",
|
||||
"Headphones", "HPR",
|
||||
"MIC", "Headset Mic",
|
||||
"Speakers", "Speaker",
|
||||
"HDMI1", "TX";
|
||||
|
||||
hs-playback-dai-link {
|
||||
link-name = "I2S0";
|
||||
dai-format = "i2s";
|
||||
mediatek,clk-provider = "cpu";
|
||||
codec {
|
||||
sound-dai = <&da7219>;
|
||||
};
|
||||
};
|
||||
|
||||
playback-codecs {
|
||||
sound-dai = <&anx_bridge_dp>,
|
||||
<&max98357a>;
|
||||
hs-capture-dai-link {
|
||||
link-name = "I2S1";
|
||||
dai-format = "i2s";
|
||||
mediatek,clk-provider = "cpu";
|
||||
codec {
|
||||
sound-dai = <&da7219>;
|
||||
};
|
||||
};
|
||||
|
||||
spk-dp-playback-dai-link {
|
||||
link-name = "I2S3";
|
||||
dai-format = "i2s";
|
||||
mediatek,clk-provider = "cpu";
|
||||
codec {
|
||||
sound-dai = <&anx_bridge_dp>, <&max98357a>;
|
||||
};
|
||||
};
|
||||
};
|
||||
|
||||
|
|
|
|||
|
|
@ -12,6 +12,9 @@ maintainers:
|
|||
description:
|
||||
This binding describes the MT8186 sound card.
|
||||
|
||||
allOf:
|
||||
- $ref: sound-card-common.yaml#
|
||||
|
||||
properties:
|
||||
compatible:
|
||||
enum:
|
||||
|
|
@ -19,6 +22,34 @@ properties:
|
|||
- mediatek,mt8186-mt6366-rt5682s-max98360-sound
|
||||
- mediatek,mt8186-mt6366-rt5650-sound
|
||||
|
||||
audio-routing:
|
||||
$ref: /schemas/types.yaml#/definitions/non-unique-string-array
|
||||
description:
|
||||
A list of the connections between audio components. Each entry is a
|
||||
pair of strings, the first being the connection's sink, the second
|
||||
being the connection's source.
|
||||
Valid names could be the input or output widgets of audio components,
|
||||
power supplies, MicBias of codec and the software switch.
|
||||
minItems: 2
|
||||
items:
|
||||
enum:
|
||||
# Sinks
|
||||
- HDMI1
|
||||
- Headphone
|
||||
- IN1P
|
||||
- IN1N
|
||||
- Line Out
|
||||
- Speakers
|
||||
|
||||
# Sources
|
||||
- Headset Mic
|
||||
- HPOL
|
||||
- HPOR
|
||||
- Speaker
|
||||
- SPOL
|
||||
- SPOR
|
||||
- TX
|
||||
|
||||
mediatek,platform:
|
||||
$ref: /schemas/types.yaml#/definitions/phandle
|
||||
description: The phandle of MT8186 ASoC platform.
|
||||
|
|
@ -32,6 +63,7 @@ properties:
|
|||
|
||||
headset-codec:
|
||||
type: object
|
||||
deprecated: true
|
||||
additionalProperties: false
|
||||
properties:
|
||||
sound-dai:
|
||||
|
|
@ -41,6 +73,7 @@ properties:
|
|||
|
||||
playback-codecs:
|
||||
type: object
|
||||
deprecated: true
|
||||
additionalProperties: false
|
||||
properties:
|
||||
sound-dai:
|
||||
|
|
@ -62,13 +95,56 @@ properties:
|
|||
A list of the desired dai-links in the sound card. Each entry is a
|
||||
name defined in the machine driver.
|
||||
|
||||
additionalProperties: false
|
||||
patternProperties:
|
||||
".*-dai-link$":
|
||||
type: object
|
||||
additionalProperties: false
|
||||
description:
|
||||
Container for dai-link level properties and CODEC sub-nodes.
|
||||
|
||||
properties:
|
||||
link-name:
|
||||
description: Indicates dai-link name and PCM stream name
|
||||
enum: [ I2S0, I2S1, I2S2, I2S3 ]
|
||||
|
||||
codec:
|
||||
description: Holds subnode which indicates codec dai.
|
||||
type: object
|
||||
additionalProperties: false
|
||||
properties:
|
||||
sound-dai:
|
||||
minItems: 1
|
||||
maxItems: 2
|
||||
required:
|
||||
- sound-dai
|
||||
|
||||
dai-format:
|
||||
description: audio format
|
||||
enum: [ i2s, right_j, left_j, dsp_a, dsp_b ]
|
||||
|
||||
mediatek,clk-provider:
|
||||
$ref: /schemas/types.yaml#/definitions/string
|
||||
description: Indicates dai-link clock master.
|
||||
enum: [ cpu, codec ]
|
||||
|
||||
required:
|
||||
- link-name
|
||||
|
||||
unevaluatedProperties: false
|
||||
|
||||
required:
|
||||
- compatible
|
||||
- mediatek,platform
|
||||
- headset-codec
|
||||
- playback-codecs
|
||||
|
||||
# Disallow legacy properties if xxx-dai-link nodes are specified
|
||||
if:
|
||||
not:
|
||||
patternProperties:
|
||||
".*-dai-link$": false
|
||||
then:
|
||||
properties:
|
||||
headset-codec: false
|
||||
speaker-codecs: false
|
||||
|
||||
examples:
|
||||
- |
|
||||
|
|
@ -76,23 +152,49 @@ examples:
|
|||
|
||||
sound: mt8186-sound {
|
||||
compatible = "mediatek,mt8186-mt6366-rt1019-rt5682s-sound";
|
||||
mediatek,platform = <&afe>;
|
||||
model = "mt8186_rt1019_rt5682s";
|
||||
pinctrl-names = "aud_clk_mosi_off",
|
||||
"aud_clk_mosi_on",
|
||||
"aud_gpio_dmic_sec";
|
||||
pinctrl-0 = <&aud_clk_mosi_off>;
|
||||
pinctrl-1 = <&aud_clk_mosi_on>;
|
||||
pinctrl-2 = <&aud_gpio_dmic_sec>;
|
||||
mediatek,platform = <&afe>;
|
||||
|
||||
dmic-gpios = <&pio 23 GPIO_ACTIVE_HIGH>;
|
||||
|
||||
headset-codec {
|
||||
sound-dai = <&rt5682s>;
|
||||
audio-routing =
|
||||
"Headphone", "HPOL",
|
||||
"Headphone", "HPOR",
|
||||
"IN1P", "Headset Mic",
|
||||
"Speakers", "Speaker",
|
||||
"HDMI1", "TX";
|
||||
|
||||
hs-playback-dai-link {
|
||||
link-name = "I2S0";
|
||||
dai-format = "i2s";
|
||||
mediatek,clk-provider = "cpu";
|
||||
codec {
|
||||
sound-dai = <&rt5682s 0>;
|
||||
};
|
||||
};
|
||||
|
||||
playback-codecs {
|
||||
sound-dai = <&it6505dptx>,
|
||||
<&rt1019p>;
|
||||
hs-capture-dai-link {
|
||||
link-name = "I2S1";
|
||||
dai-format = "i2s";
|
||||
mediatek,clk-provider = "cpu";
|
||||
codec {
|
||||
sound-dai = <&rt5682s 0>;
|
||||
};
|
||||
};
|
||||
|
||||
spk-hdmi-playback-dai-link {
|
||||
link-name = "I2S3";
|
||||
dai-format = "i2s";
|
||||
mediatek,clk-provider = "cpu";
|
||||
codec {
|
||||
sound-dai = <&it6505dptx>, <&rt1019p>;
|
||||
};
|
||||
};
|
||||
};
|
||||
|
||||
|
|
|
|||
|
|
@ -13,6 +13,9 @@ maintainers:
|
|||
description:
|
||||
This binding describes the MT8192 sound card.
|
||||
|
||||
allOf:
|
||||
- $ref: sound-card-common.yaml#
|
||||
|
||||
properties:
|
||||
compatible:
|
||||
enum:
|
||||
|
|
@ -20,6 +23,31 @@ properties:
|
|||
- mediatek,mt8192_mt6359_rt1015p_rt5682
|
||||
- mediatek,mt8192_mt6359_rt1015p_rt5682s
|
||||
|
||||
audio-routing:
|
||||
description:
|
||||
A list of the connections between audio components. Each entry is a
|
||||
pair of strings, the first being the connection's sink, the second
|
||||
being the connection's source.
|
||||
Valid names could be the input or output widgets of audio components,
|
||||
power supplies, MicBias of codec and the software switch.
|
||||
minItems: 2
|
||||
items:
|
||||
enum:
|
||||
# Sinks
|
||||
- Speakers
|
||||
- Headphone Jack
|
||||
- IN1P
|
||||
- Left Spk
|
||||
- Right Spk
|
||||
|
||||
# Sources
|
||||
- Headset Mic
|
||||
- HPOL
|
||||
- HPOR
|
||||
- Left SPO
|
||||
- Right SPO
|
||||
- Speaker
|
||||
|
||||
mediatek,platform:
|
||||
$ref: /schemas/types.yaml#/definitions/phandle
|
||||
description: The phandle of MT8192 ASoC platform.
|
||||
|
|
@ -27,10 +55,12 @@ properties:
|
|||
mediatek,hdmi-codec:
|
||||
$ref: /schemas/types.yaml#/definitions/phandle
|
||||
description: The phandle of HDMI codec.
|
||||
deprecated: true
|
||||
|
||||
headset-codec:
|
||||
type: object
|
||||
additionalProperties: false
|
||||
deprecated: true
|
||||
|
||||
properties:
|
||||
sound-dai:
|
||||
|
|
@ -41,6 +71,7 @@ properties:
|
|||
speaker-codecs:
|
||||
type: object
|
||||
additionalProperties: false
|
||||
deprecated: true
|
||||
|
||||
properties:
|
||||
sound-dai:
|
||||
|
|
@ -51,33 +82,121 @@ properties:
|
|||
required:
|
||||
- sound-dai
|
||||
|
||||
additionalProperties: false
|
||||
patternProperties:
|
||||
".*-dai-link$":
|
||||
type: object
|
||||
additionalProperties: false
|
||||
|
||||
description:
|
||||
Container for dai-link level properties and CODEC sub-nodes.
|
||||
|
||||
properties:
|
||||
link-name:
|
||||
description: Indicates dai-link name and PCM stream name
|
||||
enum:
|
||||
- I2S0
|
||||
- I2S1
|
||||
- I2S2
|
||||
- I2S3
|
||||
- I2S4
|
||||
- I2S5
|
||||
- I2S6
|
||||
- I2S7
|
||||
- I2S8
|
||||
- I2S9
|
||||
- TDM
|
||||
|
||||
codec:
|
||||
description: Holds subnode which indicates codec dai.
|
||||
type: object
|
||||
additionalProperties: false
|
||||
properties:
|
||||
sound-dai:
|
||||
minItems: 1
|
||||
maxItems: 2
|
||||
required:
|
||||
- sound-dai
|
||||
|
||||
dai-format:
|
||||
description: audio format
|
||||
enum: [ i2s, right_j, left_j, dsp_a, dsp_b ]
|
||||
|
||||
mediatek,clk-provider:
|
||||
$ref: /schemas/types.yaml#/definitions/string
|
||||
description: Indicates dai-link clock master.
|
||||
enum: [ cpu, codec ]
|
||||
|
||||
required:
|
||||
- link-name
|
||||
|
||||
unevaluatedProperties: false
|
||||
|
||||
required:
|
||||
- compatible
|
||||
- mediatek,platform
|
||||
- headset-codec
|
||||
- speaker-codecs
|
||||
|
||||
# Disallow legacy properties if xxx-dai-link nodes are specified
|
||||
if:
|
||||
not:
|
||||
patternProperties:
|
||||
".*-dai-link$": false
|
||||
then:
|
||||
properties:
|
||||
headset-codec: false
|
||||
speaker-codecs: false
|
||||
mediatek,hdmi-codec: false
|
||||
|
||||
examples:
|
||||
- |
|
||||
|
||||
sound: mt8192-sound {
|
||||
compatible = "mediatek,mt8192_mt6359_rt1015_rt5682";
|
||||
mediatek,platform = <&afe>;
|
||||
mediatek,hdmi-codec = <&anx_bridge_dp>;
|
||||
model = "mt8192_mt6359_rt1015_rt5682";
|
||||
pinctrl-names = "aud_clk_mosi_off",
|
||||
"aud_clk_mosi_on";
|
||||
pinctrl-0 = <&aud_clk_mosi_off>;
|
||||
pinctrl-1 = <&aud_clk_mosi_on>;
|
||||
mediatek,platform = <&afe>;
|
||||
|
||||
headset-codec {
|
||||
sound-dai = <&rt5682>;
|
||||
audio-routing =
|
||||
"Headphone Jack", "HPOL",
|
||||
"Headphone Jack", "HPOR",
|
||||
"IN1P", "Headset Mic",
|
||||
"Speakers", "Speaker";
|
||||
|
||||
spk-playback-dai-link {
|
||||
link-name = "I2S3";
|
||||
dai-format = "i2s";
|
||||
mediatek,clk-provider = "cpu";
|
||||
codec {
|
||||
sound-dai = <&rt1015p>;
|
||||
};
|
||||
};
|
||||
|
||||
speaker-codecs {
|
||||
sound-dai = <&rt1015_l>,
|
||||
<&rt1015_r>;
|
||||
hs-playback-dai-link {
|
||||
link-name = "I2S8";
|
||||
dai-format = "i2s";
|
||||
mediatek,clk-provider = "cpu";
|
||||
codec {
|
||||
sound-dai = <&rt5682 0>;
|
||||
};
|
||||
};
|
||||
|
||||
hs-capture-dai-link {
|
||||
link-name = "I2S9";
|
||||
dai-format = "i2s";
|
||||
mediatek,clk-provider = "cpu";
|
||||
codec {
|
||||
sound-dai = <&rt5682 0>;
|
||||
};
|
||||
};
|
||||
|
||||
displayport-dai-link {
|
||||
link-name = "TDM";
|
||||
dai-format = "dsp_a";
|
||||
codec {
|
||||
sound-dai = <&anx_bridge_dp>;
|
||||
};
|
||||
};
|
||||
};
|
||||
|
||||
|
|
|
|||
|
|
@ -12,6 +12,9 @@ maintainers:
|
|||
description:
|
||||
This binding describes the MT8195 sound card.
|
||||
|
||||
allOf:
|
||||
- $ref: sound-card-common.yaml#
|
||||
|
||||
properties:
|
||||
compatible:
|
||||
enum:
|
||||
|
|
@ -23,6 +26,33 @@ properties:
|
|||
$ref: /schemas/types.yaml#/definitions/string
|
||||
description: User specified audio sound card name
|
||||
|
||||
audio-routing:
|
||||
description:
|
||||
A list of the connections between audio components. Each entry is a
|
||||
pair of strings, the first being the connection's sink, the second
|
||||
being the connection's source.
|
||||
Valid names could be the input or output widgets of audio components,
|
||||
power supplies, MicBias of codec and the software switch.
|
||||
minItems: 2
|
||||
items:
|
||||
enum:
|
||||
# Sinks
|
||||
- Ext Spk
|
||||
- Headphone
|
||||
- IN1P
|
||||
- Left Spk
|
||||
- Right Spk
|
||||
|
||||
# Sources
|
||||
- Headset Mic
|
||||
- HPOL
|
||||
- HPOR
|
||||
- Left BE_OUT
|
||||
- Left SPO
|
||||
- Right BE_OUT
|
||||
- Right SPO
|
||||
- Speaker
|
||||
|
||||
mediatek,platform:
|
||||
$ref: /schemas/types.yaml#/definitions/phandle
|
||||
description: The phandle of MT8195 ASoC platform.
|
||||
|
|
@ -30,10 +60,12 @@ properties:
|
|||
mediatek,dptx-codec:
|
||||
$ref: /schemas/types.yaml#/definitions/phandle
|
||||
description: The phandle of MT8195 Display Port Tx codec node.
|
||||
deprecated: true
|
||||
|
||||
mediatek,hdmi-codec:
|
||||
$ref: /schemas/types.yaml#/definitions/phandle
|
||||
description: The phandle of MT8195 HDMI codec node.
|
||||
deprecated: true
|
||||
|
||||
mediatek,adsp:
|
||||
$ref: /schemas/types.yaml#/definitions/phandle
|
||||
|
|
@ -45,20 +77,122 @@ properties:
|
|||
A list of the desired dai-links in the sound card. Each entry is a
|
||||
name defined in the machine driver.
|
||||
|
||||
patternProperties:
|
||||
".*-dai-link$":
|
||||
type: object
|
||||
additionalProperties: false
|
||||
description:
|
||||
Container for dai-link level properties and CODEC sub-nodes.
|
||||
|
||||
properties:
|
||||
link-name:
|
||||
description: Indicates dai-link name and PCM stream name
|
||||
enum:
|
||||
- DPTX_BE
|
||||
- ETDM1_IN_BE
|
||||
- ETDM2_IN_BE
|
||||
- ETDM1_OUT_BE
|
||||
- ETDM2_OUT_BE
|
||||
- ETDM3_OUT_BE
|
||||
- PCM1_BE
|
||||
|
||||
codec:
|
||||
description: Holds subnode which indicates codec dai.
|
||||
type: object
|
||||
additionalProperties: false
|
||||
properties:
|
||||
sound-dai:
|
||||
minItems: 1
|
||||
maxItems: 2
|
||||
required:
|
||||
- sound-dai
|
||||
|
||||
dai-format:
|
||||
description: audio format
|
||||
enum: [ i2s, right_j, left_j, dsp_a, dsp_b ]
|
||||
|
||||
mediatek,clk-provider:
|
||||
$ref: /schemas/types.yaml#/definitions/string
|
||||
description: Indicates dai-link clock master.
|
||||
enum: [ cpu, codec ]
|
||||
|
||||
required:
|
||||
- link-name
|
||||
|
||||
additionalProperties: false
|
||||
|
||||
required:
|
||||
- compatible
|
||||
- mediatek,platform
|
||||
|
||||
# Disallow legacy properties if xxx-dai-link nodes are specified
|
||||
if:
|
||||
not:
|
||||
patternProperties:
|
||||
".*-dai-link$": false
|
||||
then:
|
||||
properties:
|
||||
mediatek,dptx-codec: false
|
||||
mediatek,hdmi-codec: false
|
||||
|
||||
examples:
|
||||
- |
|
||||
|
||||
sound: mt8195-sound {
|
||||
compatible = "mediatek,mt8195_mt6359_rt1019_rt5682";
|
||||
model = "mt8195_r1019_5682";
|
||||
mediatek,platform = <&afe>;
|
||||
pinctrl-names = "default";
|
||||
pinctrl-0 = <&aud_pins_default>;
|
||||
|
||||
audio-routing =
|
||||
"Headphone", "HPOL",
|
||||
"Headphone", "HPOR",
|
||||
"IN1P", "Headset Mic",
|
||||
"Ext Spk", "Speaker";
|
||||
|
||||
mm-dai-link {
|
||||
link-name = "ETDM1_IN_BE";
|
||||
mediatek,clk-provider = "cpu";
|
||||
};
|
||||
|
||||
hs-playback-dai-link {
|
||||
link-name = "ETDM1_OUT_BE";
|
||||
mediatek,clk-provider = "cpu";
|
||||
codec {
|
||||
sound-dai = <&headset_codec>;
|
||||
};
|
||||
};
|
||||
|
||||
hs-capture-dai-link {
|
||||
link-name = "ETDM2_IN_BE";
|
||||
mediatek,clk-provider = "cpu";
|
||||
codec {
|
||||
sound-dai = <&headset_codec>;
|
||||
};
|
||||
};
|
||||
|
||||
spk-playback-dai-link {
|
||||
link-name = "ETDM2_OUT_BE";
|
||||
mediatek,clk-provider = "cpu";
|
||||
codec {
|
||||
sound-dai = <&spk_amplifier>;
|
||||
};
|
||||
};
|
||||
|
||||
hdmi-dai-link {
|
||||
link-name = "ETDM3_OUT_BE";
|
||||
codec {
|
||||
sound-dai = <&hdmi_tx>;
|
||||
};
|
||||
};
|
||||
|
||||
displayport-dai-link {
|
||||
link-name = "DPTX_BE";
|
||||
codec {
|
||||
sound-dai = <&dp_tx>;
|
||||
};
|
||||
};
|
||||
};
|
||||
|
||||
...
|
||||
|
|
|
|||
|
|
@ -0,0 +1,80 @@
|
|||
# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
|
||||
%YAML 1.2
|
||||
---
|
||||
$id: http://devicetree.org/schemas/sound/nuvoton,nau8325.yaml#
|
||||
$schema: http://devicetree.org/meta-schemas/core.yaml#
|
||||
|
||||
title: NAU8325 audio Amplifier
|
||||
|
||||
maintainers:
|
||||
- Seven Lee <WTLI@nuvoton.com>
|
||||
|
||||
allOf:
|
||||
- $ref: dai-common.yaml#
|
||||
|
||||
properties:
|
||||
compatible:
|
||||
const: nuvoton,nau8325
|
||||
|
||||
reg:
|
||||
maxItems: 1
|
||||
|
||||
nuvoton,vref-impedance-ohms:
|
||||
description:
|
||||
The vref impedance to be used in ohms. Middle of voltage enables
|
||||
Tie-Off selection options. Due to the high impedance of the VREF
|
||||
pin, it is important to use a low-leakage capacitor.
|
||||
|
||||
enum: [0, 25000, 125000, 2500]
|
||||
|
||||
nuvoton,dac-vref-microvolt:
|
||||
description:
|
||||
The DAC vref to be used in voltage. DAC reference voltage setting. Can
|
||||
be used for minor tuning of the output level. Since the VDDA is range
|
||||
between 1.62 to 1.98 voltage, the typical value for design is 1.8V. After
|
||||
the minor tuning, the final microvolt are as the below.
|
||||
|
||||
enum: [1800000, 2700000, 2880000, 3060000]
|
||||
|
||||
nuvoton,alc-enable:
|
||||
description:
|
||||
Enable digital automatic level control (ALC) function.
|
||||
type: boolean
|
||||
|
||||
nuvoton,clock-detection-disable:
|
||||
description:
|
||||
When clock detection is enabled, it will detect whether MCLK
|
||||
and FS are within the range. MCLK range is from 2.048MHz to 24.576MHz.
|
||||
FS range is from 8kHz to 96kHz. And also needs to detect the ratio
|
||||
MCLK_SRC/LRCK of 256, 400 or 500, and needs to detect the BCLK
|
||||
to make sure data is present. There needs to be at least 8 BCLK
|
||||
cycles per Frame Sync.
|
||||
type: boolean
|
||||
|
||||
nuvoton,clock-det-data:
|
||||
description:
|
||||
Request clock detection to require 2048 non-zero samples before enabling
|
||||
the audio paths. If set then non-zero samples is required, otherwise it
|
||||
doesn't matter.
|
||||
type: boolean
|
||||
|
||||
required:
|
||||
- compatible
|
||||
- reg
|
||||
|
||||
unevaluatedProperties: false
|
||||
|
||||
examples:
|
||||
- |
|
||||
i2c {
|
||||
#address-cells = <1>;
|
||||
#size-cells = <0>;
|
||||
codec@21 {
|
||||
compatible = "nuvoton,nau8325";
|
||||
reg = <0x21>;
|
||||
nuvoton,vref-impedance-ohms = <125000>;
|
||||
nuvoton,dac-vref-microvolt = <2880000>;
|
||||
nuvoton,alc-enable;
|
||||
nuvoton,clock-det-data;
|
||||
};
|
||||
};
|
||||
|
|
@ -103,6 +103,12 @@ properties:
|
|||
just limited to the left adc for design demand.
|
||||
type: boolean
|
||||
|
||||
nuvoton,adc-delay-ms:
|
||||
description: Delay (in ms) to make input path stable and avoid pop noise.
|
||||
minimum: 125
|
||||
maximum: 500
|
||||
default: 125
|
||||
|
||||
'#sound-dai-cells':
|
||||
const: 0
|
||||
|
||||
|
|
@ -136,6 +142,7 @@ examples:
|
|||
nuvoton,jack-eject-debounce = <0>;
|
||||
nuvoton,dmic-clk-threshold = <3072000>;
|
||||
nuvoton,dmic-slew-rate = <0>;
|
||||
nuvoton,adc-delay-ms = <125>;
|
||||
#sound-dai-cells = <0>;
|
||||
};
|
||||
};
|
||||
|
|
|
|||
|
|
@ -1,36 +0,0 @@
|
|||
NVIDIA Tegra 20 AC97 controller
|
||||
|
||||
Required properties:
|
||||
- compatible : "nvidia,tegra20-ac97"
|
||||
- reg : Should contain AC97 controller registers location and length
|
||||
- interrupts : Should contain AC97 interrupt
|
||||
- resets : Must contain an entry for each entry in reset-names.
|
||||
See ../reset/reset.txt for details.
|
||||
- reset-names : Must include the following entries:
|
||||
- ac97
|
||||
- dmas : Must contain an entry for each entry in clock-names.
|
||||
See ../dma/dma.txt for details.
|
||||
- dma-names : Must include the following entries:
|
||||
- rx
|
||||
- tx
|
||||
- clocks : Must contain one entry, for the module clock.
|
||||
See ../clocks/clock-bindings.txt for details.
|
||||
- nvidia,codec-reset-gpio : The Tegra GPIO controller's phandle and the number
|
||||
of the GPIO used to reset the external AC97 codec
|
||||
- nvidia,codec-sync-gpio : The Tegra GPIO controller's phandle and the number
|
||||
of the GPIO corresponding with the AC97 DAP _FS line
|
||||
|
||||
Example:
|
||||
|
||||
ac97@70002000 {
|
||||
compatible = "nvidia,tegra20-ac97";
|
||||
reg = <0x70002000 0x200>;
|
||||
interrupts = <0 81 0x04>;
|
||||
nvidia,codec-reset-gpio = <&gpio 170 0>;
|
||||
nvidia,codec-sync-gpio = <&gpio 120 0>;
|
||||
clocks = <&tegra_car 3>;
|
||||
resets = <&tegra_car 3>;
|
||||
reset-names = "ac97";
|
||||
dmas = <&apbdma 12>, <&apbdma 12>;
|
||||
dma-names = "rx", "tx";
|
||||
};
|
||||
|
|
@ -0,0 +1,82 @@
|
|||
# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
|
||||
%YAML 1.2
|
||||
---
|
||||
$id: http://devicetree.org/schemas/sound/nvidia,tegra20-ac97.yaml#
|
||||
$schema: http://devicetree.org/meta-schemas/core.yaml#
|
||||
|
||||
title: NVIDIA Tegra20 AC97 controller
|
||||
|
||||
maintainers:
|
||||
- Thierry Reding <treding@nvidia.com>
|
||||
- Jon Hunter <jonathanh@nvidia.com>
|
||||
|
||||
properties:
|
||||
compatible:
|
||||
const: nvidia,tegra20-ac97
|
||||
|
||||
reg:
|
||||
maxItems: 1
|
||||
|
||||
resets:
|
||||
maxItems: 1
|
||||
|
||||
reset-names:
|
||||
const: ac97
|
||||
|
||||
interrupts:
|
||||
maxItems: 1
|
||||
|
||||
clocks:
|
||||
maxItems: 1
|
||||
|
||||
dmas:
|
||||
maxItems: 2
|
||||
|
||||
dma-names:
|
||||
items:
|
||||
- const: rx
|
||||
- const: tx
|
||||
|
||||
nvidia,codec-reset-gpios:
|
||||
description: Reset pin of external AC97 codec
|
||||
maxItems: 1
|
||||
|
||||
nvidia,codec-sync-gpios:
|
||||
description: AC97 DAP _FS line
|
||||
maxItems: 1
|
||||
|
||||
required:
|
||||
- compatible
|
||||
- reg
|
||||
- resets
|
||||
- reset-names
|
||||
- interrupts
|
||||
- clocks
|
||||
- dmas
|
||||
- dma-names
|
||||
- nvidia,codec-reset-gpios
|
||||
- nvidia,codec-sync-gpios
|
||||
|
||||
additionalProperties: false
|
||||
|
||||
examples:
|
||||
- |
|
||||
#include <dt-bindings/clock/tegra20-car.h>
|
||||
#include <dt-bindings/gpio/tegra-gpio.h>
|
||||
#include <dt-bindings/interrupt-controller/arm-gic.h>
|
||||
#include <dt-bindings/interrupt-controller/irq.h>
|
||||
#include <dt-bindings/gpio/gpio.h>
|
||||
|
||||
ac97@70002000 {
|
||||
compatible = "nvidia,tegra20-ac97";
|
||||
reg = <0x70002000 0x200>;
|
||||
resets = <&tegra_car 3>;
|
||||
reset-names = "ac97";
|
||||
interrupts = <GIC_SPI 81 IRQ_TYPE_LEVEL_HIGH>;
|
||||
clocks = <&tegra_car 3>;
|
||||
dmas = <&apbdma 12>, <&apbdma 12>;
|
||||
dma-names = "rx", "tx";
|
||||
nvidia,codec-reset-gpios = <&gpio TEGRA_GPIO(V, 2) GPIO_ACTIVE_HIGH>;
|
||||
nvidia,codec-sync-gpios = <&gpio TEGRA_GPIO(P, 0) GPIO_ACTIVE_HIGH>;
|
||||
};
|
||||
...
|
||||
|
|
@ -1,12 +0,0 @@
|
|||
NVIDIA Tegra 20 DAS (Digital Audio Switch) controller
|
||||
|
||||
Required properties:
|
||||
- compatible : "nvidia,tegra20-das"
|
||||
- reg : Should contain DAS registers location and length
|
||||
|
||||
Example:
|
||||
|
||||
das@70000c00 {
|
||||
compatible = "nvidia,tegra20-das";
|
||||
reg = <0x70000c00 0x80>;
|
||||
};
|
||||
|
|
@ -0,0 +1,36 @@
|
|||
# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
|
||||
%YAML 1.2
|
||||
---
|
||||
$id: http://devicetree.org/schemas/sound/nvidia,tegra20-das.yaml#
|
||||
$schema: http://devicetree.org/meta-schemas/core.yaml#
|
||||
|
||||
title: NVIDIA Tegra 20 DAS (Digital Audio Switch) controller
|
||||
|
||||
maintainers:
|
||||
- Thierry Reding <treding@nvidia.com>
|
||||
- Jon Hunter <jonathanh@nvidia.com>
|
||||
|
||||
properties:
|
||||
compatible:
|
||||
const: nvidia,tegra20-das
|
||||
|
||||
reg:
|
||||
maxItems: 1
|
||||
|
||||
required:
|
||||
- compatible
|
||||
- reg
|
||||
|
||||
additionalProperties: false
|
||||
|
||||
examples:
|
||||
- |
|
||||
bus {
|
||||
#address-cells = <1>;
|
||||
#size-cells = <1>;
|
||||
das@70000c00 {
|
||||
compatible = "nvidia,tegra20-das";
|
||||
reg = <0x70000c00 0x80>;
|
||||
};
|
||||
};
|
||||
...
|
||||
|
|
@ -1,27 +0,0 @@
|
|||
NVIDIA Tegra30 I2S controller
|
||||
|
||||
Required properties:
|
||||
- compatible : For Tegra30, must contain "nvidia,tegra30-i2s". For Tegra124,
|
||||
must contain "nvidia,tegra124-i2s". Otherwise, must contain
|
||||
"nvidia,<chip>-i2s" plus at least one of the above, where <chip> is
|
||||
tegra114 or tegra132.
|
||||
- reg : Should contain I2S registers location and length
|
||||
- clocks : Must contain one entry, for the module clock.
|
||||
See ../clocks/clock-bindings.txt for details.
|
||||
- resets : Must contain an entry for each entry in reset-names.
|
||||
See ../reset/reset.txt for details.
|
||||
- reset-names : Must include the following entries:
|
||||
- i2s
|
||||
- nvidia,ahub-cif-ids : The list of AHUB CIF IDs for this port, rx (playback)
|
||||
first, tx (capture) second. See nvidia,tegra30-ahub.txt for values.
|
||||
|
||||
Example:
|
||||
|
||||
i2s@70080300 {
|
||||
compatible = "nvidia,tegra30-i2s";
|
||||
reg = <0x70080300 0x100>;
|
||||
nvidia,ahub-cif-ids = <4 4>;
|
||||
clocks = <&tegra_car 11>;
|
||||
resets = <&tegra_car 11>;
|
||||
reset-names = "i2s";
|
||||
};
|
||||
|
|
@ -0,0 +1,67 @@
|
|||
# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
|
||||
%YAML 1.2
|
||||
---
|
||||
$id: http://devicetree.org/schemas/sound/nvidia,tegra30-i2s.yaml#
|
||||
$schema: http://devicetree.org/meta-schemas/core.yaml#
|
||||
|
||||
title: NVIDIA Tegra30 I2S controller
|
||||
|
||||
maintainers:
|
||||
- Thierry Reding <treding@nvidia.com>
|
||||
- Jon Hunter <jonathanh@nvidia.com>
|
||||
|
||||
properties:
|
||||
compatible:
|
||||
oneOf:
|
||||
- enum:
|
||||
- nvidia,tegra124-i2s
|
||||
- nvidia,tegra30-i2s
|
||||
- items:
|
||||
- const: nvidia,tegra114-i2s
|
||||
- const: nvidia,tegra30-i2s
|
||||
|
||||
reg:
|
||||
maxItems: 1
|
||||
|
||||
clocks:
|
||||
maxItems: 1
|
||||
|
||||
clock-names:
|
||||
const: i2s
|
||||
|
||||
resets:
|
||||
maxItems: 1
|
||||
|
||||
reset-names:
|
||||
const: i2s
|
||||
|
||||
nvidia,ahub-cif-ids:
|
||||
description: list of AHUB CIF IDs
|
||||
$ref: /schemas/types.yaml#/definitions/uint32-array
|
||||
items:
|
||||
- description: rx (playback)
|
||||
- description: tx (capture)
|
||||
|
||||
required:
|
||||
- compatible
|
||||
- reg
|
||||
- clocks
|
||||
- resets
|
||||
- reset-names
|
||||
- nvidia,ahub-cif-ids
|
||||
|
||||
additionalProperties: false
|
||||
|
||||
examples:
|
||||
- |
|
||||
#include <dt-bindings/clock/tegra30-car.h>
|
||||
|
||||
i2s@70080300 {
|
||||
compatible = "nvidia,tegra30-i2s";
|
||||
reg = <0x70080300 0x100>;
|
||||
nvidia,ahub-cif-ids = <4 4>;
|
||||
clocks = <&tegra_car TEGRA30_CLK_I2S0>;
|
||||
resets = <&tegra_car 30>;
|
||||
reset-names = "i2s";
|
||||
};
|
||||
...
|
||||
|
|
@ -29,6 +29,8 @@ properties:
|
|||
- enum:
|
||||
- qcom,apq8016-sbc-sndcard
|
||||
- qcom,msm8916-qdsp6-sndcard
|
||||
- qcom,qcm6490-idp-sndcard
|
||||
- qcom,qcs6490-rb3gen2-sndcard
|
||||
- qcom,qrb5165-rb5-sndcard
|
||||
- qcom,sc7180-qdsp6-sndcard
|
||||
- qcom,sc8280xp-sndcard
|
||||
|
|
|
|||
|
|
@ -48,13 +48,16 @@ properties:
|
|||
- const: renesas,rcar_sound-gen3
|
||||
# for Gen4 SoC
|
||||
- items:
|
||||
- const: renesas,rcar_sound-r8a779g0 # R-Car V4H
|
||||
- enum:
|
||||
- renesas,rcar_sound-r8a779g0 # R-Car V4H
|
||||
- renesas,rcar_sound-r8a779h0 # R-Car V4M
|
||||
- const: renesas,rcar_sound-gen4
|
||||
# for Generic
|
||||
- enum:
|
||||
- renesas,rcar_sound-gen1
|
||||
- renesas,rcar_sound-gen2
|
||||
- renesas,rcar_sound-gen3
|
||||
- renesas,rcar_sound-gen4
|
||||
|
||||
reg:
|
||||
minItems: 1
|
||||
|
|
|
|||
|
|
@ -0,0 +1,98 @@
|
|||
# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause
|
||||
%YAML 1.2
|
||||
---
|
||||
$id: http://devicetree.org/schemas/sound/rockchip,rk3308-codec.yaml#
|
||||
$schema: http://devicetree.org/meta-schemas/core.yaml#
|
||||
|
||||
title: Rockchip RK3308 Internal Codec
|
||||
|
||||
description: |
|
||||
This is the audio codec embedded in the Rockchip RK3308
|
||||
SoC. It has 8 24-bit ADCs and 2 24-bit DACs. The maximum supported
|
||||
sampling rate is 192 kHz.
|
||||
|
||||
It is connected internally to one out of a selection of the internal I2S
|
||||
controllers.
|
||||
|
||||
The RK3308 audio codec has 8 independent capture channels, but some
|
||||
features work on stereo pairs called groups:
|
||||
* grp 0 -- MIC1 / MIC2
|
||||
* grp 1 -- MIC3 / MIC4
|
||||
* grp 2 -- MIC5 / MIC6
|
||||
* grp 3 -- MIC7 / MIC8
|
||||
|
||||
maintainers:
|
||||
- Luca Ceresoli <luca.ceresoli@bootlin.com>
|
||||
|
||||
properties:
|
||||
compatible:
|
||||
const: rockchip,rk3308-codec
|
||||
|
||||
reg:
|
||||
maxItems: 1
|
||||
|
||||
rockchip,grf:
|
||||
$ref: /schemas/types.yaml#/definitions/phandle
|
||||
description:
|
||||
Phandle to the General Register Files (GRF)
|
||||
|
||||
clocks:
|
||||
items:
|
||||
- description: clock for TX
|
||||
- description: clock for RX
|
||||
- description: AHB clock driving the interface
|
||||
|
||||
clock-names:
|
||||
items:
|
||||
- const: mclk_tx
|
||||
- const: mclk_rx
|
||||
- const: hclk
|
||||
|
||||
resets:
|
||||
maxItems: 1
|
||||
|
||||
reset-names:
|
||||
items:
|
||||
- const: codec
|
||||
|
||||
"#sound-dai-cells":
|
||||
const: 0
|
||||
|
||||
rockchip,micbias-avdd-percent:
|
||||
description: |
|
||||
Voltage setting for the MICBIAS pins expressed as a percentage of
|
||||
AVDD.
|
||||
|
||||
E.g. if rockchip,micbias-avdd-percent = 85 and AVDD = 3v3, then the
|
||||
MIC BIAS voltage will be 3.3 V * 85% = 2.805 V.
|
||||
|
||||
enum: [ 50, 55, 60, 65, 70, 75, 80, 85 ]
|
||||
|
||||
required:
|
||||
- compatible
|
||||
- reg
|
||||
- rockchip,grf
|
||||
- clocks
|
||||
- resets
|
||||
- "#sound-dai-cells"
|
||||
|
||||
additionalProperties: false
|
||||
|
||||
examples:
|
||||
- |
|
||||
#include <dt-bindings/clock/rk3308-cru.h>
|
||||
|
||||
audio_codec: audio-codec@ff560000 {
|
||||
compatible = "rockchip,rk3308-codec";
|
||||
reg = <0xff560000 0x10000>;
|
||||
rockchip,grf = <&grf>;
|
||||
clock-names = "mclk_tx", "mclk_rx", "hclk";
|
||||
clocks = <&cru SCLK_I2S2_8CH_TX_OUT>,
|
||||
<&cru SCLK_I2S2_8CH_RX_OUT>,
|
||||
<&cru PCLK_ACODEC>;
|
||||
reset-names = "codec";
|
||||
resets = <&cru SRST_ACODEC_P>;
|
||||
#sound-dai-cells = <0>;
|
||||
};
|
||||
|
||||
...
|
||||
|
|
@ -1,15 +0,0 @@
|
|||
Texas Instruments PCM1681 8-channel PWM Processor
|
||||
|
||||
Required properties:
|
||||
|
||||
- compatible: Should contain "ti,pcm1681".
|
||||
- reg: The i2c address. Should contain <0x4c>.
|
||||
|
||||
Examples:
|
||||
|
||||
i2c_bus {
|
||||
pcm1681@4c {
|
||||
compatible = "ti,pcm1681";
|
||||
reg = <0x4c>;
|
||||
};
|
||||
};
|
||||
|
|
@ -0,0 +1,43 @@
|
|||
# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
|
||||
%YAML 1.2
|
||||
---
|
||||
$id: http://devicetree.org/schemas/sound/ti,pcm1681.yaml#
|
||||
$schema: http://devicetree.org/meta-schemas/core.yaml#
|
||||
|
||||
title: Texas Instruments PCM1681 8-channel PWM Processor
|
||||
|
||||
maintainers:
|
||||
- Shenghao Ding <shenghao-ding@ti.com>
|
||||
- Kevin Lu <kevin-lu@ti.com>
|
||||
- Baojun Xu <baojun.xu@ti.com>
|
||||
|
||||
allOf:
|
||||
- $ref: dai-common.yaml#
|
||||
|
||||
properties:
|
||||
compatible:
|
||||
const: ti,pcm1681
|
||||
|
||||
reg:
|
||||
maxItems: 1
|
||||
|
||||
"#sound-dai-cells":
|
||||
const: 0
|
||||
|
||||
required:
|
||||
- compatible
|
||||
- reg
|
||||
|
||||
unevaluatedProperties: false
|
||||
|
||||
examples:
|
||||
- |
|
||||
i2c {
|
||||
#address-cells = <1>;
|
||||
#size-cells = <0>;
|
||||
|
||||
pcm1681: audio-codec@4c {
|
||||
compatible = "ti,pcm1681";
|
||||
reg = <0x4c>;
|
||||
};
|
||||
};
|
||||
|
|
@ -0,0 +1,177 @@
|
|||
# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
|
||||
# Copyright (C) 2022 - 2024 Texas Instruments Incorporated
|
||||
%YAML 1.2
|
||||
---
|
||||
$id: http://devicetree.org/schemas/sound/ti,pcm6240.yaml#
|
||||
$schema: http://devicetree.org/meta-schemas/core.yaml#
|
||||
|
||||
title: Texas Instruments PCM6240 Family Audio ADC/DAC
|
||||
|
||||
maintainers:
|
||||
- Shenghao Ding <shenghao-ding@ti.com>
|
||||
|
||||
description: |
|
||||
The PCM6240 Family is a big family of Audio ADC/DAC for
|
||||
different Specifications, range from Personal Electric
|
||||
to Automotive Electric, even some professional fields.
|
||||
|
||||
Specifications about the audio chip can be found at:
|
||||
https://www.ti.com/lit/gpn/tlv320adc3120
|
||||
https://www.ti.com/lit/gpn/tlv320adc5120
|
||||
https://www.ti.com/lit/gpn/tlv320adc6120
|
||||
https://www.ti.com/lit/gpn/dix4192
|
||||
https://www.ti.com/lit/gpn/pcm1690
|
||||
https://www.ti.com/lit/gpn/pcm3120-q1
|
||||
https://www.ti.com/lit/gpn/pcm3140-q1
|
||||
https://www.ti.com/lit/gpn/pcm5120-q1
|
||||
https://www.ti.com/lit/gpn/pcm6120-q1
|
||||
https://www.ti.com/lit/gpn/pcm6260-q1
|
||||
https://www.ti.com/lit/gpn/pcm9211
|
||||
https://www.ti.com/lit/gpn/pcmd3140
|
||||
https://www.ti.com/lit/gpn/pcmd3180
|
||||
https://www.ti.com/lit/gpn/taa5212
|
||||
https://www.ti.com/lit/gpn/tad5212
|
||||
|
||||
properties:
|
||||
compatible:
|
||||
description: |
|
||||
ti,adc3120: Stereo-channel, 768-kHz, Burr-Brown™ audio analog-to-
|
||||
digital converter (ADC) with 106-dB SNR.
|
||||
|
||||
ti,adc5120: 2-Channel, 768-kHz, Burr-Brown™ Audio ADC with 120-dB SNR.
|
||||
|
||||
ti,adc6120: Stereo-channel, 768-kHz, Burr-Brown™ audio analog-to-
|
||||
digital converter (ADC) with 123-dB SNR.
|
||||
|
||||
ti,dix4192: 216-kHz digital audio converter with Quad-Channel In
|
||||
and One-Channel Out.
|
||||
|
||||
ti,pcm1690: Automotive Catalog 113dB SNR 8-Channel Audio DAC with
|
||||
Differential Outputs.
|
||||
|
||||
ti,pcm3120: Automotive, stereo, 106-dB SNR, 768-kHz, low-power
|
||||
software-controlled audio ADC.
|
||||
|
||||
ti,pcm3140: Automotive, Quad-Channel, 768-kHz, Burr-Brown™ Audio ADC
|
||||
with 106-dB SNR.
|
||||
|
||||
ti,pcm5120: Automotive, stereo, 120-dB SNR, 768-kHz, low-power
|
||||
software-controlled audio ADC.
|
||||
|
||||
ti,pcm5140: Automotive, Quad-Channel, 768-kHz, Burr-Brown™ Audio ADC
|
||||
with 120-dB SNR.
|
||||
|
||||
ti,pcm6120: Automotive, stereo, 123-dB SNR, 768-kHz, low-power
|
||||
software-controlled audio ADC.
|
||||
|
||||
ti,pcm6140: Automotive, Quad-Channel, 768-kHz, Burr-Brown™ Audio ADC
|
||||
with 123-dB SNR.
|
||||
|
||||
ti,pcm6240: Automotive 4-ch audio ADC with integrated programmable mic
|
||||
bias, boost and input diagnostics.
|
||||
|
||||
ti,pcm6260: Automotive 6-ch audio ADC with integrated programmable mic
|
||||
bias, boost and input diagnostics.
|
||||
|
||||
ti,pcm9211: 216-kHz digital audio converter With Stereo ADC and
|
||||
Routing.
|
||||
|
||||
ti,pcmd3140: Four-channel PDM-input to TDM or I2S output converter.
|
||||
|
||||
ti,pcmd3180: Eight-channel pulse-density-modulation input to TDM or
|
||||
I2S output converter.
|
||||
|
||||
ti,taa5212: Low-power high-performance stereo audio ADC with 118-dB
|
||||
dynamic range.
|
||||
|
||||
ti,tad5212: Low-power stereo audio DAC with 120-dB dynamic range.
|
||||
oneOf:
|
||||
- items:
|
||||
- enum:
|
||||
- ti,adc3120
|
||||
- ti,adc5120
|
||||
- ti,pcm3120
|
||||
- ti,pcm5120
|
||||
- ti,pcm6120
|
||||
- const: ti,adc6120
|
||||
- items:
|
||||
- enum:
|
||||
- ti,pcmd512x
|
||||
- ti,pcm9211
|
||||
- ti,taa5212
|
||||
- ti,tad5212
|
||||
- const: ti,adc6120
|
||||
- items:
|
||||
- enum:
|
||||
- ti,pcm3140
|
||||
- ti,pcm5140
|
||||
- ti,dix4192
|
||||
- ti,pcm6140
|
||||
- ti,pcm6260
|
||||
- const: ti,pcm6240
|
||||
- items:
|
||||
- enum:
|
||||
- ti,pcmd3140
|
||||
- ti,pcmd3180
|
||||
- ti,pcm1690
|
||||
- ti,taa5412
|
||||
- ti,tad5412
|
||||
- const: ti,pcm6240
|
||||
- enum:
|
||||
- ti,adc6120
|
||||
- ti,pcm6240
|
||||
|
||||
reg:
|
||||
description:
|
||||
I2C address, in multiple pcmdevices case, all the i2c address
|
||||
aggregate as one Audio Device to support multiple audio slots.
|
||||
minItems: 1
|
||||
maxItems: 4
|
||||
|
||||
reset-gpios:
|
||||
maxItems: 1
|
||||
|
||||
interrupts:
|
||||
maxItems: 1
|
||||
description:
|
||||
Invalid only for ti,pcm1690 because of no INT pin.
|
||||
|
||||
'#sound-dai-cells':
|
||||
const: 0
|
||||
|
||||
required:
|
||||
- compatible
|
||||
- reg
|
||||
|
||||
allOf:
|
||||
- $ref: dai-common.yaml#
|
||||
- if:
|
||||
properties:
|
||||
compatible:
|
||||
contains:
|
||||
enum:
|
||||
- ti,pcm1690
|
||||
then:
|
||||
properties:
|
||||
interrupts: false
|
||||
|
||||
additionalProperties: false
|
||||
|
||||
examples:
|
||||
- |
|
||||
#include <dt-bindings/gpio/gpio.h>
|
||||
i2c {
|
||||
/* example for two devices with interrupt support */
|
||||
#address-cells = <1>;
|
||||
#size-cells = <0>;
|
||||
pcm6240: audio-codec@48 {
|
||||
compatible = "ti,pcm6240";
|
||||
reg = <0x48>, /* primary-device */
|
||||
<0x4b>; /* secondary-device */
|
||||
#sound-dai-cells = <0>;
|
||||
reset-gpios = <&gpio1 10 GPIO_ACTIVE_HIGH>;
|
||||
interrupt-parent = <&gpio1>;
|
||||
interrupts = <15>;
|
||||
};
|
||||
};
|
||||
...
|
||||
|
|
@ -0,0 +1,41 @@
|
|||
# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause
|
||||
%YAML 1.2
|
||||
---
|
||||
$id: http://devicetree.org/schemas/sound/wlf,wm8776.yaml#
|
||||
$schema: http://devicetree.org/meta-schemas/core.yaml#
|
||||
|
||||
title: WM8776 audio CODEC
|
||||
|
||||
maintainers:
|
||||
- patches@opensource.cirrus.com
|
||||
|
||||
allOf:
|
||||
- $ref: dai-common.yaml#
|
||||
|
||||
properties:
|
||||
compatible:
|
||||
const: wlf,wm8776
|
||||
|
||||
reg:
|
||||
maxItems: 1
|
||||
|
||||
"#sound-dai-cells":
|
||||
const: 0
|
||||
|
||||
required:
|
||||
- compatible
|
||||
- reg
|
||||
|
||||
unevaluatedProperties: false
|
||||
|
||||
examples:
|
||||
- |
|
||||
i2c {
|
||||
#address-cells = <1>;
|
||||
#size-cells = <0>;
|
||||
|
||||
codec@1a {
|
||||
compatible = "wlf,wm8776";
|
||||
reg = <0x1a>;
|
||||
};
|
||||
};
|
||||
|
|
@ -1,15 +0,0 @@
|
|||
WM8974 audio CODEC
|
||||
|
||||
This device supports both I2C and SPI (configured with pin strapping
|
||||
on the board).
|
||||
|
||||
Required properties:
|
||||
- compatible: "wlf,wm8974"
|
||||
- reg: the I2C address or SPI chip select number of the device
|
||||
|
||||
Examples:
|
||||
|
||||
codec: wm8974@1a {
|
||||
compatible = "wlf,wm8974";
|
||||
reg = <0x1a>;
|
||||
};
|
||||
|
|
@ -0,0 +1,41 @@
|
|||
# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause
|
||||
%YAML 1.2
|
||||
---
|
||||
$id: http://devicetree.org/schemas/sound/wlf,wm8974.yaml#
|
||||
$schema: http://devicetree.org/meta-schemas/core.yaml#
|
||||
|
||||
title: WM8974 audio CODEC
|
||||
|
||||
maintainers:
|
||||
- patches@opensource.cirrus.com
|
||||
|
||||
allOf:
|
||||
- $ref: dai-common.yaml#
|
||||
|
||||
properties:
|
||||
compatible:
|
||||
const: wlf,wm8974
|
||||
|
||||
reg:
|
||||
maxItems: 1
|
||||
|
||||
"#sound-dai-cells":
|
||||
const: 0
|
||||
|
||||
required:
|
||||
- compatible
|
||||
- reg
|
||||
|
||||
unevaluatedProperties: false
|
||||
|
||||
examples:
|
||||
- |
|
||||
i2c {
|
||||
#address-cells = <1>;
|
||||
#size-cells = <0>;
|
||||
|
||||
codec@1a {
|
||||
compatible = "wlf,wm8974";
|
||||
reg = <0x1a>;
|
||||
};
|
||||
};
|
||||
|
|
@ -1,18 +0,0 @@
|
|||
WM8776 audio CODEC
|
||||
|
||||
This device supports both I2C and SPI (configured with pin strapping
|
||||
on the board).
|
||||
|
||||
Required properties:
|
||||
|
||||
- compatible : "wlf,wm8776"
|
||||
|
||||
- reg : the I2C address of the device for I2C, the chip select
|
||||
number for SPI.
|
||||
|
||||
Example:
|
||||
|
||||
wm8776: codec@1a {
|
||||
compatible = "wlf,wm8776";
|
||||
reg = <0x1a>;
|
||||
};
|
||||
|
|
@ -3976,7 +3976,7 @@ Driver with A Single Source File
|
|||
|
||||
Suppose you have a file xyz.c. Add the following two lines::
|
||||
|
||||
snd-xyz-objs := xyz.o
|
||||
snd-xyz-y := xyz.o
|
||||
obj-$(CONFIG_SND_XYZ) += snd-xyz.o
|
||||
|
||||
2. Create the Kconfig entry
|
||||
|
|
@ -4019,7 +4019,7 @@ located in the new subdirectory, sound/pci/xyz.
|
|||
|
||||
2. Under the directory ``sound/pci/xyz``, create a Makefile::
|
||||
|
||||
snd-xyz-objs := xyz.o abc.o def.o
|
||||
snd-xyz-y := xyz.o abc.o def.o
|
||||
obj-$(CONFIG_SND_XYZ) += snd-xyz.o
|
||||
|
||||
3. Create the Kconfig entry
|
||||
|
|
|
|||
|
|
@ -0,0 +1,375 @@
|
|||
<?xml version="1.0" encoding="UTF-8" standalone="no"?>
|
||||
<!DOCTYPE svg PUBLIC "-//W3C//DTD SVG 1.1//EN"
|
||||
"http://www.w3.org/Graphics/SVG/1.1/DTD/svg11.dtd">
|
||||
<!-- Generated by graphviz version 2.43.0 (0)
|
||||
-->
|
||||
<!-- Title: G Pages: 1 -->
|
||||
<svg width="900pt" height="630pt"
|
||||
viewBox="0.00 0.00 900.00 630.00" xmlns="http://www.w3.org/2000/svg" xmlns:xlink="http://www.w3.org/1999/xlink">
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<g id="graph0" class="graph" transform="scale(1 1) rotate(0) translate(4 626)">
|
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<title>G</title>
|
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<polygon fill="white" stroke="transparent" points="-4,4 -4,-626 896,-626 896,4 -4,4"/>
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<g id="clust1" class="cluster">
|
||||
<title>ROOT</title>
|
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<polygon fill="none" stroke="dodgerblue" points="8,-537 8,-614 102,-614 102,-537 8,-537"/>
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<text text-anchor="middle" x="55" y="-598.8" font-family="sans-serif" font-size="14.00">ROOT</text>
|
||||
</g>
|
||||
<g id="clust2" class="cluster">
|
||||
<title>4000b000.audio-controller</title>
|
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<polygon fill="none" stroke="dodgerblue" points="120,-378 120,-455 312,-455 312,-378 120,-378"/>
|
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<text text-anchor="middle" x="216" y="-439.8" font-family="sans-serif" font-size="14.00">4000b000.audio-controller</text>
|
||||
</g>
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<g id="clust5" class="cluster">
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<title>cs42l51.0-004a</title>
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<polygon fill="none" stroke="dodgerblue" points="330,-8 330,-614 884,-614 884,-8 330,-8"/>
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<text text-anchor="middle" x="607" y="-598.8" font-family="sans-serif" font-size="14.00">cs42l51.0-004a</text>
|
||||
</g>
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<g id="clust9" class="cluster">
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<title>hdmi-audio-codec.1.auto</title>
|
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<polygon fill="none" stroke="dodgerblue" points="110,-463 110,-614 314,-614 314,-463 110,-463"/>
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<text text-anchor="middle" x="212" y="-598.8" font-family="sans-serif" font-size="14.00">hdmi-audio-codec.1.auto</text>
|
||||
</g>
|
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<!-- ROOT_Amplifier -->
|
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<g id="node1" class="node">
|
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<title>ROOT_Amplifier</title>
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<polygon fill="#f2f2f2" stroke="#4d4d4d" points="93.5,-583 16.5,-583 16.5,-545 93.5,-545 93.5,-583"/>
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<text text-anchor="middle" x="55" y="-567.8" font-family="sans-serif" font-size="14.00">Amplifier</text>
|
||||
<text text-anchor="middle" x="55" y="-552.8" font-family="sans-serif" font-size="14.00">[out_drv]</text>
|
||||
</g>
|
||||
<!-- 4000b000.audio-controller_capture -->
|
||||
<g id="node2" class="node">
|
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<title>4000b000.audio-controller_capture</title>
|
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<polygon fill="#f2f2f2" stroke="#4d4d4d" points="202,-424 128,-424 128,-386 202,-386 202,-424"/>
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<text text-anchor="middle" x="165" y="-408.8" font-family="sans-serif" font-size="14.00">capture</text>
|
||||
<text text-anchor="middle" x="165" y="-393.8" font-family="sans-serif" font-size="14.00">[dai_out]</text>
|
||||
</g>
|
||||
<!-- 4000b000.audio-controller_playback -->
|
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<g id="node3" class="node">
|
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<title>4000b000.audio-controller_playback</title>
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<polygon fill="#f2f2f2" stroke="#4d4d4d" points="304,-424 230,-424 230,-386 304,-386 304,-424"/>
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<text text-anchor="middle" x="267" y="-408.8" font-family="sans-serif" font-size="14.00">playback</text>
|
||||
<text text-anchor="middle" x="267" y="-393.8" font-family="sans-serif" font-size="14.00">[dai_in]</text>
|
||||
</g>
|
||||
<!-- hdmi-audio-codec.1.auto_I2S Playback -->
|
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<g id="node28" class="node">
|
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<title>hdmi-audio-codec.1.auto_I2S Playback</title>
|
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<polygon fill="#f2f2f2" stroke="#4d4d4d" points="306,-583 208,-583 208,-545 306,-545 306,-583"/>
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<text text-anchor="middle" x="257" y="-567.8" font-family="sans-serif" font-size="14.00">I2S Playback</text>
|
||||
<text text-anchor="middle" x="257" y="-552.8" font-family="sans-serif" font-size="14.00">[dai_in]</text>
|
||||
</g>
|
||||
<!-- 4000b000.audio-controller_playback->hdmi-audio-codec.1.auto_I2S Playback -->
|
||||
<g id="edge21" class="edge">
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<title>4000b000.audio-controller_playback->hdmi-audio-codec.1.auto_I2S Playback</title>
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<path fill="none" stroke="black" d="M276.84,-424.14C282.19,-435.06 288.26,-449.42 291,-463 295.05,-483.04 296.67,-489.36 291,-509 288.25,-518.54 283.26,-528.01 277.93,-536.3"/>
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<polygon fill="black" stroke="black" points="274.89,-534.55 272.11,-544.78 280.66,-538.51 274.89,-534.55"/>
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</g>
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<!-- hdmi-audio-codec.1.auto_Capture -->
|
||||
<g id="node4" class="node">
|
||||
<title>hdmi-audio-codec.1.auto_Capture</title>
|
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<polygon fill="#f2f2f2" stroke="#4d4d4d" points="192,-509 118,-509 118,-471 192,-471 192,-509"/>
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<text text-anchor="middle" x="155" y="-493.8" font-family="sans-serif" font-size="14.00">Capture</text>
|
||||
<text text-anchor="middle" x="155" y="-478.8" font-family="sans-serif" font-size="14.00">[dai_out]</text>
|
||||
</g>
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<!-- hdmi-audio-codec.1.auto_Capture->4000b000.audio-controller_capture -->
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<g id="edge1" class="edge">
|
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<title>hdmi-audio-codec.1.auto_Capture->4000b000.audio-controller_capture</title>
|
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<path fill="none" stroke="black" d="M157.17,-470.99C158.46,-460.3 160.12,-446.5 161.58,-434.37"/>
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<polygon fill="black" stroke="black" points="165.08,-434.61 162.8,-424.26 158.13,-433.77 165.08,-434.61"/>
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</g>
|
||||
<!-- cs42l51.0-004a_AIN1L -->
|
||||
<g id="node5" class="node">
|
||||
<title>cs42l51.0-004a_AIN1L</title>
|
||||
<polygon fill="#f2f2f2" stroke="#4d4d4d" points="836.5,-583 775.5,-583 775.5,-545 836.5,-545 836.5,-583"/>
|
||||
<text text-anchor="middle" x="806" y="-567.8" font-family="sans-serif" font-size="14.00">AIN1L</text>
|
||||
<text text-anchor="middle" x="806" y="-552.8" font-family="sans-serif" font-size="14.00">[input]</text>
|
||||
</g>
|
||||
<!-- cs42l51.0-004a_PGA-ADC Mux Left -->
|
||||
<g id="node22" class="node">
|
||||
<title>cs42l51.0-004a_PGA-ADC Mux Left</title>
|
||||
<polygon fill="#f2f2f2" stroke="#4d4d4d" points="876,-509 736,-509 736,-471 876,-471 876,-509"/>
|
||||
<text text-anchor="middle" x="806" y="-493.8" font-family="sans-serif" font-size="14.00">PGA-ADC Mux Left</text>
|
||||
<text text-anchor="middle" x="806" y="-478.8" font-family="sans-serif" font-size="14.00">[mux]</text>
|
||||
</g>
|
||||
<!-- cs42l51.0-004a_AIN1L->cs42l51.0-004a_PGA-ADC Mux Left -->
|
||||
<g id="edge14" class="edge">
|
||||
<title>cs42l51.0-004a_AIN1L->cs42l51.0-004a_PGA-ADC Mux Left</title>
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<path fill="none" stroke="black" d="M806,-544.83C806,-537.13 806,-527.97 806,-519.42"/>
|
||||
<polygon fill="black" stroke="black" points="809.5,-519.41 806,-509.41 802.5,-519.41 809.5,-519.41"/>
|
||||
</g>
|
||||
<!-- cs42l51.0-004a_AIN1R -->
|
||||
<g id="node6" class="node">
|
||||
<title>cs42l51.0-004a_AIN1R</title>
|
||||
<polygon fill="#f2f2f2" stroke="#4d4d4d" points="738.5,-583 677.5,-583 677.5,-545 738.5,-545 738.5,-583"/>
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||||
<text text-anchor="middle" x="708" y="-567.8" font-family="sans-serif" font-size="14.00">AIN1R</text>
|
||||
<text text-anchor="middle" x="708" y="-552.8" font-family="sans-serif" font-size="14.00">[input]</text>
|
||||
</g>
|
||||
<!-- cs42l51.0-004a_PGA-ADC Mux Right -->
|
||||
<g id="node23" class="node">
|
||||
<title>cs42l51.0-004a_PGA-ADC Mux Right</title>
|
||||
<polygon fill="#f2f2f2" stroke="#4d4d4d" points="717.5,-509 568.5,-509 568.5,-471 717.5,-471 717.5,-509"/>
|
||||
<text text-anchor="middle" x="643" y="-493.8" font-family="sans-serif" font-size="14.00">PGA-ADC Mux Right</text>
|
||||
<text text-anchor="middle" x="643" y="-478.8" font-family="sans-serif" font-size="14.00">[mux]</text>
|
||||
</g>
|
||||
<!-- cs42l51.0-004a_AIN1R->cs42l51.0-004a_PGA-ADC Mux Right -->
|
||||
<g id="edge15" class="edge">
|
||||
<title>cs42l51.0-004a_AIN1R->cs42l51.0-004a_PGA-ADC Mux Right</title>
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||||
<path fill="none" stroke="black" d="M691.6,-544.83C683.96,-536.37 674.73,-526.15 666.38,-516.9"/>
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||||
<polygon fill="black" stroke="black" points="668.92,-514.49 659.62,-509.41 663.73,-519.18 668.92,-514.49"/>
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||||
</g>
|
||||
<!-- cs42l51.0-004a_AIN2L -->
|
||||
<g id="node7" class="node">
|
||||
<title>cs42l51.0-004a_AIN2L</title>
|
||||
<polygon fill="#f2f2f2" stroke="#4d4d4d" points="659.5,-583 598.5,-583 598.5,-545 659.5,-545 659.5,-583"/>
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||||
<text text-anchor="middle" x="629" y="-567.8" font-family="sans-serif" font-size="14.00">AIN2L</text>
|
||||
<text text-anchor="middle" x="629" y="-552.8" font-family="sans-serif" font-size="14.00">[input]</text>
|
||||
</g>
|
||||
<!-- cs42l51.0-004a_AIN2R -->
|
||||
<g id="node8" class="node">
|
||||
<title>cs42l51.0-004a_AIN2R</title>
|
||||
<polygon fill="#f2f2f2" stroke="#4d4d4d" points="580.5,-583 519.5,-583 519.5,-545 580.5,-545 580.5,-583"/>
|
||||
<text text-anchor="middle" x="550" y="-567.8" font-family="sans-serif" font-size="14.00">AIN2R</text>
|
||||
<text text-anchor="middle" x="550" y="-552.8" font-family="sans-serif" font-size="14.00">[input]</text>
|
||||
</g>
|
||||
<!-- cs42l51.0-004a_Capture -->
|
||||
<g id="node9" class="node">
|
||||
<title>cs42l51.0-004a_Capture</title>
|
||||
<polygon fill="#f2f2f2" stroke="#4d4d4d" points="692,-276 618,-276 618,-238 692,-238 692,-276"/>
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||||
<text text-anchor="middle" x="655" y="-260.8" font-family="sans-serif" font-size="14.00">Capture</text>
|
||||
<text text-anchor="middle" x="655" y="-245.8" font-family="sans-serif" font-size="14.00">[dai_out]</text>
|
||||
</g>
|
||||
<!-- cs42l51.0-004a_DAC Mux -->
|
||||
<g id="node10" class="node">
|
||||
<title>cs42l51.0-004a_DAC Mux</title>
|
||||
<polygon fill="none" stroke="#008b00" stroke-width="2" points="598.5,-202 521.5,-202 521.5,-164 598.5,-164 598.5,-202"/>
|
||||
<text text-anchor="middle" x="560" y="-186.8" font-family="sans-serif" font-size="14.00">DAC Mux</text>
|
||||
<text text-anchor="middle" x="560" y="-171.8" font-family="sans-serif" font-size="14.00">[mux]</text>
|
||||
</g>
|
||||
<!-- cs42l51.0-004a_Left DAC -->
|
||||
<g id="node14" class="node">
|
||||
<title>cs42l51.0-004a_Left DAC</title>
|
||||
<polygon fill="none" stroke="#008b00" stroke-width="2" points="548,-128 474,-128 474,-90 548,-90 548,-128"/>
|
||||
<text text-anchor="middle" x="511" y="-112.8" font-family="sans-serif" font-size="14.00">Left DAC</text>
|
||||
<text text-anchor="middle" x="511" y="-97.8" font-family="sans-serif" font-size="14.00">[dac]</text>
|
||||
</g>
|
||||
<!-- cs42l51.0-004a_DAC Mux->cs42l51.0-004a_Left DAC -->
|
||||
<g id="edge9" class="edge">
|
||||
<title>cs42l51.0-004a_DAC Mux->cs42l51.0-004a_Left DAC</title>
|
||||
<path fill="none" stroke="black" d="M547.64,-163.83C542.05,-155.62 535.34,-145.76 529.19,-136.73"/>
|
||||
<polygon fill="black" stroke="black" points="532.05,-134.71 523.53,-128.41 526.26,-138.65 532.05,-134.71"/>
|
||||
</g>
|
||||
<!-- cs42l51.0-004a_Right DAC -->
|
||||
<g id="node26" class="node">
|
||||
<title>cs42l51.0-004a_Right DAC</title>
|
||||
<polygon fill="none" stroke="#008b00" stroke-width="2" points="649.5,-128 566.5,-128 566.5,-90 649.5,-90 649.5,-128"/>
|
||||
<text text-anchor="middle" x="608" y="-112.8" font-family="sans-serif" font-size="14.00">Right DAC</text>
|
||||
<text text-anchor="middle" x="608" y="-97.8" font-family="sans-serif" font-size="14.00">[dac]</text>
|
||||
</g>
|
||||
<!-- cs42l51.0-004a_DAC Mux->cs42l51.0-004a_Right DAC -->
|
||||
<g id="edge18" class="edge">
|
||||
<title>cs42l51.0-004a_DAC Mux->cs42l51.0-004a_Right DAC</title>
|
||||
<path fill="none" stroke="black" d="M572.11,-163.83C577.53,-155.71 584.02,-145.96 589.99,-137.01"/>
|
||||
<polygon fill="black" stroke="black" points="593.09,-138.68 595.72,-128.41 587.27,-134.79 593.09,-138.68"/>
|
||||
</g>
|
||||
<!-- cs42l51.0-004a_HPL -->
|
||||
<g id="node11" class="node">
|
||||
<title>cs42l51.0-004a_HPL</title>
|
||||
<polygon fill="none" stroke="#008b00" stroke-width="2" points="546.5,-54 475.5,-54 475.5,-16 546.5,-16 546.5,-54"/>
|
||||
<text text-anchor="middle" x="511" y="-38.8" font-family="sans-serif" font-size="14.00">HPL</text>
|
||||
<text text-anchor="middle" x="511" y="-23.8" font-family="sans-serif" font-size="14.00">[output]</text>
|
||||
</g>
|
||||
<!-- cs42l51.0-004a_HPR -->
|
||||
<g id="node12" class="node">
|
||||
<title>cs42l51.0-004a_HPR</title>
|
||||
<polygon fill="none" stroke="#008b00" stroke-width="2" points="643.5,-54 572.5,-54 572.5,-16 643.5,-16 643.5,-54"/>
|
||||
<text text-anchor="middle" x="608" y="-38.8" font-family="sans-serif" font-size="14.00">HPR</text>
|
||||
<text text-anchor="middle" x="608" y="-23.8" font-family="sans-serif" font-size="14.00">[output]</text>
|
||||
</g>
|
||||
<!-- cs42l51.0-004a_Left ADC -->
|
||||
<g id="node13" class="node">
|
||||
<title>cs42l51.0-004a_Left ADC</title>
|
||||
<polygon fill="#f2f2f2" stroke="#4d4d4d" points="822,-350 748,-350 748,-312 822,-312 822,-350"/>
|
||||
<text text-anchor="middle" x="785" y="-334.8" font-family="sans-serif" font-size="14.00">Left ADC</text>
|
||||
<text text-anchor="middle" x="785" y="-319.8" font-family="sans-serif" font-size="14.00">[adc]</text>
|
||||
</g>
|
||||
<!-- cs42l51.0-004a_Left ADC->cs42l51.0-004a_Capture -->
|
||||
<g id="edge4" class="edge">
|
||||
<title>cs42l51.0-004a_Left ADC->cs42l51.0-004a_Capture</title>
|
||||
<path fill="none" stroke="black" d="M752.2,-311.83C735.41,-302.54 714.8,-291.12 696.88,-281.2"/>
|
||||
<polygon fill="black" stroke="black" points="698.24,-277.95 687.79,-276.16 694.85,-284.07 698.24,-277.95"/>
|
||||
</g>
|
||||
<!-- cs42l51.0-004a_Left DAC->cs42l51.0-004a_HPL -->
|
||||
<g id="edge6" class="edge">
|
||||
<title>cs42l51.0-004a_Left DAC->cs42l51.0-004a_HPL</title>
|
||||
<path fill="none" stroke="black" d="M511,-89.83C511,-82.13 511,-72.97 511,-64.42"/>
|
||||
<polygon fill="black" stroke="black" points="514.5,-64.41 511,-54.41 507.5,-64.41 514.5,-64.41"/>
|
||||
</g>
|
||||
<!-- cs42l51.0-004a_Left PGA -->
|
||||
<g id="node15" class="node">
|
||||
<title>cs42l51.0-004a_Left PGA</title>
|
||||
<polygon fill="#f2f2f2" stroke="#4d4d4d" points="838,-424 764,-424 764,-386 838,-386 838,-424"/>
|
||||
<text text-anchor="middle" x="801" y="-408.8" font-family="sans-serif" font-size="14.00">Left PGA</text>
|
||||
<text text-anchor="middle" x="801" y="-393.8" font-family="sans-serif" font-size="14.00">[pga]</text>
|
||||
</g>
|
||||
<!-- cs42l51.0-004a_Left PGA->cs42l51.0-004a_Left ADC -->
|
||||
<g id="edge8" class="edge">
|
||||
<title>cs42l51.0-004a_Left PGA->cs42l51.0-004a_Left ADC</title>
|
||||
<path fill="none" stroke="black" d="M796.96,-385.83C795.25,-378.13 793.22,-368.97 791.31,-360.42"/>
|
||||
<polygon fill="black" stroke="black" points="794.68,-359.42 789.09,-350.41 787.84,-360.93 794.68,-359.42"/>
|
||||
</g>
|
||||
<!-- cs42l51.0-004a_MCLK -->
|
||||
<g id="node16" class="node">
|
||||
<title>cs42l51.0-004a_MCLK</title>
|
||||
<polygon fill="none" stroke="#008b00" stroke-width="2" points="594.5,-350 525.5,-350 525.5,-312 594.5,-312 594.5,-350"/>
|
||||
<text text-anchor="middle" x="560" y="-334.8" font-family="sans-serif" font-size="14.00">MCLK</text>
|
||||
<text text-anchor="middle" x="560" y="-319.8" font-family="sans-serif" font-size="14.00">[supply]</text>
|
||||
</g>
|
||||
<!-- cs42l51.0-004a_MCLK->cs42l51.0-004a_Capture -->
|
||||
<g id="edge2" class="edge">
|
||||
<title>cs42l51.0-004a_MCLK->cs42l51.0-004a_Capture</title>
|
||||
<path fill="none" stroke="black" d="M583.97,-311.83C595.79,-302.88 610.2,-291.96 622.94,-282.3"/>
|
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<polygon fill="black" stroke="black" points="625.18,-284.99 631.04,-276.16 620.95,-279.41 625.18,-284.99"/>
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</g>
|
||||
<!-- cs42l51.0-004a_Playback -->
|
||||
<g id="node24" class="node">
|
||||
<title>cs42l51.0-004a_Playback</title>
|
||||
<polygon fill="none" stroke="#008b00" stroke-width="2" points="597,-276 523,-276 523,-238 597,-238 597,-276"/>
|
||||
<text text-anchor="middle" x="560" y="-260.8" font-family="sans-serif" font-size="14.00">Playback</text>
|
||||
<text text-anchor="middle" x="560" y="-245.8" font-family="sans-serif" font-size="14.00">[dai_in]</text>
|
||||
</g>
|
||||
<!-- cs42l51.0-004a_MCLK->cs42l51.0-004a_Playback -->
|
||||
<g id="edge16" class="edge">
|
||||
<title>cs42l51.0-004a_MCLK->cs42l51.0-004a_Playback</title>
|
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<path fill="none" stroke="black" d="M560,-311.83C560,-304.13 560,-294.97 560,-286.42"/>
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<polygon fill="black" stroke="black" points="563.5,-286.41 560,-276.41 556.5,-286.41 563.5,-286.41"/>
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</g>
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<!-- cs42l51.0-004a_MICL -->
|
||||
<g id="node17" class="node">
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||||
<title>cs42l51.0-004a_MICL</title>
|
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<polygon fill="#f2f2f2" stroke="#4d4d4d" points="399.5,-509 338.5,-509 338.5,-471 399.5,-471 399.5,-509"/>
|
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<text text-anchor="middle" x="369" y="-493.8" font-family="sans-serif" font-size="14.00">MICL</text>
|
||||
<text text-anchor="middle" x="369" y="-478.8" font-family="sans-serif" font-size="14.00">[input]</text>
|
||||
</g>
|
||||
<!-- cs42l51.0-004a_Mic Preamp Left -->
|
||||
<g id="node20" class="node">
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||||
<title>cs42l51.0-004a_Mic Preamp Left</title>
|
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<polygon fill="#f2f2f2" stroke="#4d4d4d" points="461.5,-424 338.5,-424 338.5,-386 461.5,-386 461.5,-424"/>
|
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<text text-anchor="middle" x="400" y="-408.8" font-family="sans-serif" font-size="14.00">Mic Preamp Left</text>
|
||||
<text text-anchor="middle" x="400" y="-393.8" font-family="sans-serif" font-size="14.00">[mixer]</text>
|
||||
</g>
|
||||
<!-- cs42l51.0-004a_MICL->cs42l51.0-004a_Mic Preamp Left -->
|
||||
<g id="edge12" class="edge">
|
||||
<title>cs42l51.0-004a_MICL->cs42l51.0-004a_Mic Preamp Left</title>
|
||||
<path fill="none" stroke="black" d="M375.73,-470.99C379.8,-460.08 385.08,-445.94 389.68,-433.64"/>
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<polygon fill="black" stroke="black" points="392.96,-434.85 393.18,-424.26 386.4,-432.4 392.96,-434.85"/>
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</g>
|
||||
<!-- cs42l51.0-004a_MICR -->
|
||||
<g id="node18" class="node">
|
||||
<title>cs42l51.0-004a_MICR</title>
|
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<polygon fill="#f2f2f2" stroke="#4d4d4d" points="501.5,-583 440.5,-583 440.5,-545 501.5,-545 501.5,-583"/>
|
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<text text-anchor="middle" x="471" y="-567.8" font-family="sans-serif" font-size="14.00">MICR</text>
|
||||
<text text-anchor="middle" x="471" y="-552.8" font-family="sans-serif" font-size="14.00">[input]</text>
|
||||
</g>
|
||||
<!-- cs42l51.0-004a_Mic Preamp Right -->
|
||||
<g id="node21" class="node">
|
||||
<title>cs42l51.0-004a_Mic Preamp Right</title>
|
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<polygon fill="#f2f2f2" stroke="#4d4d4d" points="550.5,-509 417.5,-509 417.5,-471 550.5,-471 550.5,-509"/>
|
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<text text-anchor="middle" x="484" y="-493.8" font-family="sans-serif" font-size="14.00">Mic Preamp Right</text>
|
||||
<text text-anchor="middle" x="484" y="-478.8" font-family="sans-serif" font-size="14.00">[mixer]</text>
|
||||
</g>
|
||||
<!-- cs42l51.0-004a_MICR->cs42l51.0-004a_Mic Preamp Right -->
|
||||
<g id="edge13" class="edge">
|
||||
<title>cs42l51.0-004a_MICR->cs42l51.0-004a_Mic Preamp Right</title>
|
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<path fill="none" stroke="black" d="M474.28,-544.83C475.67,-537.13 477.32,-527.97 478.87,-519.42"/>
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<polygon fill="black" stroke="black" points="482.34,-519.88 480.68,-509.41 475.45,-518.63 482.34,-519.88"/>
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</g>
|
||||
<!-- cs42l51.0-004a_Mic Bias -->
|
||||
<g id="node19" class="node">
|
||||
<title>cs42l51.0-004a_Mic Bias</title>
|
||||
<polygon fill="#f2f2f2" stroke="#4d4d4d" points="409.5,-583 338.5,-583 338.5,-545 409.5,-545 409.5,-583"/>
|
||||
<text text-anchor="middle" x="374" y="-567.8" font-family="sans-serif" font-size="14.00">Mic Bias</text>
|
||||
<text text-anchor="middle" x="374" y="-552.8" font-family="sans-serif" font-size="14.00">[supply]</text>
|
||||
</g>
|
||||
<!-- cs42l51.0-004a_Mic Bias->cs42l51.0-004a_MICL -->
|
||||
<g id="edge11" class="edge">
|
||||
<title>cs42l51.0-004a_Mic Bias->cs42l51.0-004a_MICL</title>
|
||||
<path fill="none" stroke="black" d="M372.74,-544.83C372.2,-537.13 371.57,-527.97 370.97,-519.42"/>
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<polygon fill="black" stroke="black" points="374.46,-519.15 370.28,-509.41 367.48,-519.63 374.46,-519.15"/>
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</g>
|
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<!-- cs42l51.0-004a_PGA-ADC Mux Left->cs42l51.0-004a_Left PGA -->
|
||||
<g id="edge10" class="edge">
|
||||
<title>cs42l51.0-004a_PGA-ADC Mux Left->cs42l51.0-004a_Left PGA</title>
|
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<path fill="none" stroke="black" d="M804.92,-470.99C804.27,-460.3 803.44,-446.5 802.71,-434.37"/>
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<polygon fill="black" stroke="black" points="806.2,-434.03 802.1,-424.26 799.21,-434.45 806.2,-434.03"/>
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</g>
|
||||
<!-- cs42l51.0-004a_Right PGA -->
|
||||
<g id="node27" class="node">
|
||||
<title>cs42l51.0-004a_Right PGA</title>
|
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<polygon fill="#f2f2f2" stroke="#4d4d4d" points="688.5,-424 605.5,-424 605.5,-386 688.5,-386 688.5,-424"/>
|
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<text text-anchor="middle" x="647" y="-408.8" font-family="sans-serif" font-size="14.00">Right PGA</text>
|
||||
<text text-anchor="middle" x="647" y="-393.8" font-family="sans-serif" font-size="14.00">[pga]</text>
|
||||
</g>
|
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<!-- cs42l51.0-004a_PGA-ADC Mux Right->cs42l51.0-004a_Right PGA -->
|
||||
<g id="edge19" class="edge">
|
||||
<title>cs42l51.0-004a_PGA-ADC Mux Right->cs42l51.0-004a_Right PGA</title>
|
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<path fill="none" stroke="black" d="M643.87,-470.99C644.38,-460.3 645.05,-446.5 645.63,-434.37"/>
|
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<polygon fill="black" stroke="black" points="649.13,-434.42 646.12,-424.26 642.14,-434.08 649.13,-434.42"/>
|
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</g>
|
||||
<!-- cs42l51.0-004a_Playback->cs42l51.0-004a_DAC Mux -->
|
||||
<g id="edge5" class="edge">
|
||||
<title>cs42l51.0-004a_Playback->cs42l51.0-004a_DAC Mux</title>
|
||||
<path fill="none" stroke="black" d="M560,-237.83C560,-230.13 560,-220.97 560,-212.42"/>
|
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<polygon fill="black" stroke="black" points="563.5,-212.41 560,-202.41 556.5,-212.41 563.5,-212.41"/>
|
||||
</g>
|
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<!-- cs42l51.0-004a_Right ADC -->
|
||||
<g id="node25" class="node">
|
||||
<title>cs42l51.0-004a_Right ADC</title>
|
||||
<polygon fill="#f2f2f2" stroke="#4d4d4d" points="697,-350 613,-350 613,-312 697,-312 697,-350"/>
|
||||
<text text-anchor="middle" x="655" y="-334.8" font-family="sans-serif" font-size="14.00">Right ADC</text>
|
||||
<text text-anchor="middle" x="655" y="-319.8" font-family="sans-serif" font-size="14.00">[adc]</text>
|
||||
</g>
|
||||
<!-- cs42l51.0-004a_Right ADC->cs42l51.0-004a_Capture -->
|
||||
<g id="edge3" class="edge">
|
||||
<title>cs42l51.0-004a_Right ADC->cs42l51.0-004a_Capture</title>
|
||||
<path fill="none" stroke="black" d="M655,-311.83C655,-304.13 655,-294.97 655,-286.42"/>
|
||||
<polygon fill="black" stroke="black" points="658.5,-286.41 655,-276.41 651.5,-286.41 658.5,-286.41"/>
|
||||
</g>
|
||||
<!-- cs42l51.0-004a_Right DAC->cs42l51.0-004a_HPR -->
|
||||
<g id="edge7" class="edge">
|
||||
<title>cs42l51.0-004a_Right DAC->cs42l51.0-004a_HPR</title>
|
||||
<path fill="none" stroke="black" d="M608,-89.83C608,-82.13 608,-72.97 608,-64.42"/>
|
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<polygon fill="black" stroke="black" points="611.5,-64.41 608,-54.41 604.5,-64.41 611.5,-64.41"/>
|
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</g>
|
||||
<!-- cs42l51.0-004a_Right PGA->cs42l51.0-004a_Right ADC -->
|
||||
<g id="edge17" class="edge">
|
||||
<title>cs42l51.0-004a_Right PGA->cs42l51.0-004a_Right ADC</title>
|
||||
<path fill="none" stroke="black" d="M649.02,-385.83C649.87,-378.13 650.89,-368.97 651.84,-360.42"/>
|
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<polygon fill="black" stroke="black" points="655.33,-360.74 652.95,-350.41 648.37,-359.97 655.33,-360.74"/>
|
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</g>
|
||||
<!-- hdmi-audio-codec.1.auto_TX -->
|
||||
<g id="node30" class="node">
|
||||
<title>hdmi-audio-codec.1.auto_TX</title>
|
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<polygon fill="#f2f2f2" stroke="#4d4d4d" points="281.5,-509 210.5,-509 210.5,-471 281.5,-471 281.5,-509"/>
|
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<text text-anchor="middle" x="246" y="-493.8" font-family="sans-serif" font-size="14.00">TX</text>
|
||||
<text text-anchor="middle" x="246" y="-478.8" font-family="sans-serif" font-size="14.00">[output]</text>
|
||||
</g>
|
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<!-- hdmi-audio-codec.1.auto_I2S Playback->hdmi-audio-codec.1.auto_TX -->
|
||||
<g id="edge22" class="edge">
|
||||
<title>hdmi-audio-codec.1.auto_I2S Playback->hdmi-audio-codec.1.auto_TX</title>
|
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<path fill="none" stroke="black" d="M254.22,-544.83C253.05,-537.13 251.65,-527.97 250.34,-519.42"/>
|
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<polygon fill="black" stroke="black" points="253.78,-518.77 248.81,-509.41 246.86,-519.83 253.78,-518.77"/>
|
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</g>
|
||||
<!-- hdmi-audio-codec.1.auto_RX -->
|
||||
<g id="node29" class="node">
|
||||
<title>hdmi-audio-codec.1.auto_RX</title>
|
||||
<polygon fill="#f2f2f2" stroke="#4d4d4d" points="189.5,-583 118.5,-583 118.5,-545 189.5,-545 189.5,-583"/>
|
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<text text-anchor="middle" x="154" y="-567.8" font-family="sans-serif" font-size="14.00">RX</text>
|
||||
<text text-anchor="middle" x="154" y="-552.8" font-family="sans-serif" font-size="14.00">[output]</text>
|
||||
</g>
|
||||
<!-- hdmi-audio-codec.1.auto_RX->hdmi-audio-codec.1.auto_Capture -->
|
||||
<g id="edge20" class="edge">
|
||||
<title>hdmi-audio-codec.1.auto_RX->hdmi-audio-codec.1.auto_Capture</title>
|
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<path fill="none" stroke="black" d="M154.25,-544.83C154.36,-537.13 154.49,-527.97 154.61,-519.42"/>
|
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<polygon fill="black" stroke="black" points="158.1,-519.46 154.74,-509.41 151.11,-519.36 158.1,-519.46"/>
|
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</g>
|
||||
</g>
|
||||
</svg>
|
||||
|
After Width: | Height: | Size: 23 KiB |
|
|
@ -7,8 +7,8 @@ Description
|
|||
|
||||
Dynamic Audio Power Management (DAPM) is designed to allow portable
|
||||
Linux devices to use the minimum amount of power within the audio
|
||||
subsystem at all times. It is independent of other kernel PM and as
|
||||
such, can easily co-exist with the other PM systems.
|
||||
subsystem at all times. It is independent of other kernel power
|
||||
management frameworks and, as such, can easily co-exist with them.
|
||||
|
||||
DAPM is also completely transparent to all user space applications as
|
||||
all power switching is done within the ASoC core. No code changes or
|
||||
|
|
@ -16,11 +16,29 @@ recompiling are required for user space applications. DAPM makes power
|
|||
switching decisions based upon any audio stream (capture/playback)
|
||||
activity and audio mixer settings within the device.
|
||||
|
||||
DAPM spans the whole machine. It covers power control within the entire
|
||||
audio subsystem, this includes internal codec power blocks and machine
|
||||
level power systems.
|
||||
DAPM is based on two basic elements, called widgets and routes:
|
||||
|
||||
There are 4 power domains within DAPM
|
||||
* a **widget** is every part of the audio hardware that can be enabled by
|
||||
software when in use and disabled to save power when not in use
|
||||
* a **route** is an interconnection between widgets that exists when sound
|
||||
can flow from one widget to the other
|
||||
|
||||
All DAPM power switching decisions are made automatically by consulting an
|
||||
audio routing graph. This graph is specific to each sound card and spans
|
||||
the whole sound card, so some DAPM routes connect two widgets belonging to
|
||||
different components (e.g. the LINE OUT pin of a CODEC and the input pin of
|
||||
an amplifier).
|
||||
|
||||
The graph for the STM32MP1-DK1 sound card is shown in picture:
|
||||
|
||||
.. kernel-figure:: dapm-graph.svg
|
||||
:alt: Example DAPM graph
|
||||
:align: center
|
||||
|
||||
DAPM power domains
|
||||
==================
|
||||
|
||||
There are 4 power domains within DAPM:
|
||||
|
||||
Codec bias domain
|
||||
VREF, VMID (core codec and audio power)
|
||||
|
|
@ -47,17 +65,11 @@ Stream domain
|
|||
Enabled and disabled when stream playback/capture is started and
|
||||
stopped respectively. e.g. aplay, arecord.
|
||||
|
||||
All DAPM power switching decisions are made automatically by consulting an audio
|
||||
routing map of the whole machine. This map is specific to each machine and
|
||||
consists of the interconnections between every audio component (including
|
||||
internal codec components). All audio components that effect power are called
|
||||
widgets hereafter.
|
||||
|
||||
|
||||
DAPM Widgets
|
||||
============
|
||||
|
||||
Audio DAPM widgets fall into a number of types:-
|
||||
Audio DAPM widgets fall into a number of types:
|
||||
|
||||
Mixer
|
||||
Mixes several analog signals into a single analog signal.
|
||||
|
|
@ -141,14 +153,14 @@ Stream Widgets relate to the stream power domain and only consist of ADCs
|
|||
(analog to digital converters), DACs (digital to analog converters),
|
||||
AIF IN and AIF OUT.
|
||||
|
||||
Stream widgets have the following format:-
|
||||
Stream widgets have the following format:
|
||||
::
|
||||
|
||||
SND_SOC_DAPM_DAC(name, stream name, reg, shift, invert),
|
||||
SND_SOC_DAPM_AIF_IN(name, stream, slot, reg, shift, invert)
|
||||
|
||||
NOTE: the stream name must match the corresponding stream name in your codec
|
||||
snd_soc_codec_dai.
|
||||
snd_soc_dai_driver.
|
||||
|
||||
e.g. stream widgets for HiFi playback and capture
|
||||
::
|
||||
|
|
@ -167,7 +179,7 @@ Path Domain Widgets
|
|||
-------------------
|
||||
|
||||
Path domain widgets have a ability to control or affect the audio signal or
|
||||
audio paths within the audio subsystem. They have the following form:-
|
||||
audio paths within the audio subsystem. They have the following form:
|
||||
::
|
||||
|
||||
SND_SOC_DAPM_PGA(name, reg, shift, invert, controls, num_controls)
|
||||
|
|
@ -207,7 +219,7 @@ powered. e.g.
|
|||
A machine widget can have an optional call back.
|
||||
|
||||
e.g. Jack connector widget for an external Mic that enables Mic Bias
|
||||
when the Mic is inserted:-::
|
||||
when the Mic is inserted::
|
||||
|
||||
static int spitz_mic_bias(struct snd_soc_dapm_widget* w, int event)
|
||||
{
|
||||
|
|
@ -221,7 +233,7 @@ when the Mic is inserted:-::
|
|||
Codec (BIAS) Domain
|
||||
-------------------
|
||||
|
||||
The codec bias power domain has no widgets and is handled by the codecs DAPM
|
||||
The codec bias power domain has no widgets and is handled by the codec DAPM
|
||||
event handler. This handler is called when the codec powerstate is changed wrt
|
||||
to any stream event or by kernel PM events.
|
||||
|
||||
|
|
@ -229,17 +241,58 @@ to any stream event or by kernel PM events.
|
|||
Virtual Widgets
|
||||
---------------
|
||||
|
||||
Sometimes widgets exist in the codec or machine audio map that don't have any
|
||||
Sometimes widgets exist in the codec or machine audio graph that don't have any
|
||||
corresponding soft power control. In this case it is necessary to create
|
||||
a virtual widget - a widget with no control bits e.g.
|
||||
::
|
||||
|
||||
SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
|
||||
|
||||
This can be used to merge to signal paths together in software.
|
||||
This can be used to merge two signal paths together in software.
|
||||
|
||||
After all the widgets have been defined, they can then be added to the DAPM
|
||||
subsystem individually with a call to snd_soc_dapm_new_control().
|
||||
Registering DAPM controls
|
||||
=========================
|
||||
|
||||
In many cases the DAPM widgets are implemented statically in a ``static
|
||||
const struct snd_soc_dapm_widget`` array in a codec driver, and simply
|
||||
declared via the ``dapm_widgets`` and ``num_dapm_widgets`` fields of the
|
||||
``struct snd_soc_component_driver``.
|
||||
|
||||
Similarly, routes connecting them are implemented statically in a ``static
|
||||
const struct snd_soc_dapm_route`` array and declared via the
|
||||
``dapm_routes`` and ``num_dapm_routes`` fields of the same struct.
|
||||
|
||||
With the above declared, the driver registration will take care of
|
||||
populating them::
|
||||
|
||||
static const struct snd_soc_dapm_widget wm2000_dapm_widgets[] = {
|
||||
SND_SOC_DAPM_OUTPUT("SPKN"),
|
||||
SND_SOC_DAPM_OUTPUT("SPKP"),
|
||||
...
|
||||
};
|
||||
|
||||
/* Target, Path, Source */
|
||||
static const struct snd_soc_dapm_route wm2000_audio_map[] = {
|
||||
{ "SPKN", NULL, "ANC Engine" },
|
||||
{ "SPKP", NULL, "ANC Engine" },
|
||||
...
|
||||
};
|
||||
|
||||
static const struct snd_soc_component_driver soc_component_dev_wm2000 = {
|
||||
...
|
||||
.dapm_widgets = wm2000_dapm_widgets,
|
||||
.num_dapm_widgets = ARRAY_SIZE(wm2000_dapm_widgets),
|
||||
.dapm_routes = wm2000_audio_map,
|
||||
.num_dapm_routes = ARRAY_SIZE(wm2000_audio_map),
|
||||
...
|
||||
};
|
||||
|
||||
In more complex cases the list of DAPM widgets and/or routes can be only
|
||||
known at probe time. This happens for example when a driver supports
|
||||
different models having a different set of features. In those cases
|
||||
separate widgets and routes arrays implementing the case-specific features
|
||||
can be registered programmatically by calling snd_soc_dapm_new_controls()
|
||||
and snd_soc_dapm_add_routes().
|
||||
|
||||
|
||||
Codec/DSP Widget Interconnections
|
||||
|
|
@ -247,31 +300,29 @@ Codec/DSP Widget Interconnections
|
|||
|
||||
Widgets are connected to each other within the codec, platform and machine by
|
||||
audio paths (called interconnections). Each interconnection must be defined in
|
||||
order to create a map of all audio paths between widgets.
|
||||
order to create a graph of all audio paths between widgets.
|
||||
|
||||
This is easiest with a diagram of the codec or DSP (and schematic of the machine
|
||||
audio system), as it requires joining widgets together via their audio signal
|
||||
paths.
|
||||
|
||||
e.g., from the WM8731 output mixer (wm8731.c)
|
||||
|
||||
The WM8731 output mixer has 3 inputs (sources)
|
||||
For example the WM8731 output mixer (wm8731.c) has 3 inputs (sources):
|
||||
|
||||
1. Line Bypass Input
|
||||
2. DAC (HiFi playback)
|
||||
3. Mic Sidetone Input
|
||||
|
||||
Each input in this example has a kcontrol associated with it (defined in example
|
||||
above) and is connected to the output mixer via its kcontrol name. We can now
|
||||
connect the destination widget (wrt audio signal) with its source widgets.
|
||||
::
|
||||
Each input in this example has a kcontrol associated with it (defined in
|
||||
the example above) and is connected to the output mixer via its kcontrol
|
||||
name. We can now connect the destination widget (wrt audio signal) with its
|
||||
source widgets. ::
|
||||
|
||||
/* output mixer */
|
||||
{"Output Mixer", "Line Bypass Switch", "Line Input"},
|
||||
{"Output Mixer", "HiFi Playback Switch", "DAC"},
|
||||
{"Output Mixer", "Mic Sidetone Switch", "Mic Bias"},
|
||||
|
||||
So we have :-
|
||||
So we have:
|
||||
|
||||
* Destination Widget <=== Path Name <=== Source Widget, or
|
||||
* Sink, Path, Source, or
|
||||
|
|
@ -280,12 +331,11 @@ So we have :-
|
|||
When there is no path name connecting widgets (e.g. a direct connection) we
|
||||
pass NULL for the path name.
|
||||
|
||||
Interconnections are created with a call to:-
|
||||
::
|
||||
Interconnections are created with a call to::
|
||||
|
||||
snd_soc_dapm_connect_input(codec, sink, path, source);
|
||||
|
||||
Finally, snd_soc_dapm_new_widgets(codec) must be called after all widgets and
|
||||
Finally, snd_soc_dapm_new_widgets() must be called after all widgets and
|
||||
interconnections have been registered with the core. This causes the core to
|
||||
scan the codec and machine so that the internal DAPM state matches the
|
||||
physical state of the machine.
|
||||
|
|
@ -326,35 +376,44 @@ jacks can also be switched OFF.
|
|||
DAPM Widget Events
|
||||
==================
|
||||
|
||||
Some widgets can register their interest with the DAPM core in PM events.
|
||||
e.g. A Speaker with an amplifier registers a widget so the amplifier can be
|
||||
powered only when the spk is in use.
|
||||
::
|
||||
Widgets needing to implement a more complex behaviour than what DAPM can do
|
||||
can set a custom "event handler" by setting a function pointer. An example
|
||||
is a power supply needing to enable a GPIO::
|
||||
|
||||
/* turn speaker amplifier on/off depending on use */
|
||||
static int corgi_amp_event(struct snd_soc_dapm_widget *w, int event)
|
||||
static int sof_es8316_speaker_power_event(struct snd_soc_dapm_widget *w,
|
||||
struct snd_kcontrol *kcontrol, int event)
|
||||
{
|
||||
gpio_set_value(CORGI_GPIO_APM_ON, SND_SOC_DAPM_EVENT_ON(event));
|
||||
return 0;
|
||||
if (SND_SOC_DAPM_EVENT_ON(event))
|
||||
gpiod_set_value_cansleep(gpio_pa, true);
|
||||
else
|
||||
gpiod_set_value_cansleep(gpio_pa, false);
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
/* corgi machine dapm widgets */
|
||||
static const struct snd_soc_dapm_widget wm8731_dapm_widgets =
|
||||
SND_SOC_DAPM_SPK("Ext Spk", corgi_amp_event);
|
||||
static const struct snd_soc_dapm_widget st_widgets[] = {
|
||||
...
|
||||
SND_SOC_DAPM_SUPPLY("Speaker Power", SND_SOC_NOPM, 0, 0,
|
||||
sof_es8316_speaker_power_event,
|
||||
SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
|
||||
};
|
||||
|
||||
Please see soc-dapm.h for all other widgets that support events.
|
||||
See soc-dapm.h for all other widgets that support events.
|
||||
|
||||
|
||||
Event types
|
||||
-----------
|
||||
|
||||
The following event types are supported by event widgets.
|
||||
::
|
||||
The following event types are supported by event widgets::
|
||||
|
||||
/* dapm event types */
|
||||
#define SND_SOC_DAPM_PRE_PMU 0x1 /* before widget power up */
|
||||
#define SND_SOC_DAPM_POST_PMU 0x2 /* after widget power up */
|
||||
#define SND_SOC_DAPM_PRE_PMD 0x4 /* before widget power down */
|
||||
#define SND_SOC_DAPM_POST_PMD 0x8 /* after widget power down */
|
||||
#define SND_SOC_DAPM_PRE_REG 0x10 /* before audio path setup */
|
||||
#define SND_SOC_DAPM_POST_REG 0x20 /* after audio path setup */
|
||||
#define SND_SOC_DAPM_PRE_PMU 0x1 /* before widget power up */
|
||||
#define SND_SOC_DAPM_POST_PMU 0x2 /* after widget power up */
|
||||
#define SND_SOC_DAPM_PRE_PMD 0x4 /* before widget power down */
|
||||
#define SND_SOC_DAPM_POST_PMD 0x8 /* after widget power down */
|
||||
#define SND_SOC_DAPM_PRE_REG 0x10 /* before audio path setup */
|
||||
#define SND_SOC_DAPM_POST_REG 0x20 /* after audio path setup */
|
||||
#define SND_SOC_DAPM_WILL_PMU 0x40 /* called at start of sequence */
|
||||
#define SND_SOC_DAPM_WILL_PMD 0x80 /* called at start of sequence */
|
||||
#define SND_SOC_DAPM_PRE_POST_PMD (SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD)
|
||||
#define SND_SOC_DAPM_PRE_POST_PMU (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU)
|
||||
|
|
|
|||
13
MAINTAINERS
13
MAINTAINERS
|
|
@ -19240,6 +19240,13 @@ S: Maintained
|
|||
F: Documentation/devicetree/bindings/media/rockchip-rga.yaml
|
||||
F: drivers/media/platform/rockchip/rga/
|
||||
|
||||
ROCKCHIP RK3308 INTERNAL AUDIO CODEC
|
||||
M: Luca Ceresoli <luca.ceresoli@bootlin.com>
|
||||
S: Maintained
|
||||
F: Documentation/devicetree/bindings/sound/rockchip,rk3308-codec.yaml
|
||||
F: sound/soc/codecs/rk3308_codec.c
|
||||
F: sound/soc/codecs/rk3308_codec.h
|
||||
|
||||
ROCKCHIP VIDEO DECODER DRIVER
|
||||
M: Ezequiel Garcia <ezequiel@vanguardiasur.com.ar>
|
||||
L: linux-media@vger.kernel.org
|
||||
|
|
@ -20824,6 +20831,12 @@ F: include/trace/events/sof*.h
|
|||
F: include/uapi/sound/asoc.h
|
||||
F: sound/soc/
|
||||
|
||||
SOUND - SOC LAYER / dapm-graph
|
||||
M: Luca Ceresoli <luca.ceresoli@bootlin.com>
|
||||
L: linux-sound@vger.kernel.org
|
||||
S: Maintained
|
||||
F: tools/sound/dapm-graph
|
||||
|
||||
SOUND - SOUND OPEN FIRMWARE (SOF) DRIVERS
|
||||
M: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
|
||||
M: Liam Girdwood <lgirdwood@gmail.com>
|
||||
|
|
|
|||
|
|
@ -819,6 +819,33 @@ int cs_dsp_coeff_write_ctrl(struct cs_dsp_coeff_ctl *ctl,
|
|||
}
|
||||
EXPORT_SYMBOL_NS_GPL(cs_dsp_coeff_write_ctrl, FW_CS_DSP);
|
||||
|
||||
/**
|
||||
* cs_dsp_coeff_lock_and_write_ctrl() - Writes the given buffer to the given coefficient control
|
||||
* @ctl: pointer to coefficient control
|
||||
* @off: word offset at which data should be written
|
||||
* @buf: the buffer to write to the given control
|
||||
* @len: the length of the buffer in bytes
|
||||
*
|
||||
* Same as cs_dsp_coeff_write_ctrl() but takes pwr_lock.
|
||||
*
|
||||
* Return: A negative number on error, 1 when the control value changed and 0 when it has not.
|
||||
*/
|
||||
int cs_dsp_coeff_lock_and_write_ctrl(struct cs_dsp_coeff_ctl *ctl,
|
||||
unsigned int off, const void *buf, size_t len)
|
||||
{
|
||||
struct cs_dsp *dsp = ctl->dsp;
|
||||
int ret;
|
||||
|
||||
lockdep_assert_not_held(&dsp->pwr_lock);
|
||||
|
||||
mutex_lock(&dsp->pwr_lock);
|
||||
ret = cs_dsp_coeff_write_ctrl(ctl, off, buf, len);
|
||||
mutex_unlock(&dsp->pwr_lock);
|
||||
|
||||
return ret;
|
||||
}
|
||||
EXPORT_SYMBOL_GPL(cs_dsp_coeff_lock_and_write_ctrl);
|
||||
|
||||
static int cs_dsp_coeff_read_ctrl_raw(struct cs_dsp_coeff_ctl *ctl,
|
||||
unsigned int off, void *buf, size_t len)
|
||||
{
|
||||
|
|
@ -891,6 +918,33 @@ int cs_dsp_coeff_read_ctrl(struct cs_dsp_coeff_ctl *ctl,
|
|||
}
|
||||
EXPORT_SYMBOL_NS_GPL(cs_dsp_coeff_read_ctrl, FW_CS_DSP);
|
||||
|
||||
/**
|
||||
* cs_dsp_coeff_lock_and_read_ctrl() - Reads the given coefficient control into the given buffer
|
||||
* @ctl: pointer to coefficient control
|
||||
* @off: word offset at which data should be read
|
||||
* @buf: the buffer to store to the given control
|
||||
* @len: the length of the buffer in bytes
|
||||
*
|
||||
* Same as cs_dsp_coeff_read_ctrl() but takes pwr_lock.
|
||||
*
|
||||
* Return: Zero for success, a negative number on error.
|
||||
*/
|
||||
int cs_dsp_coeff_lock_and_read_ctrl(struct cs_dsp_coeff_ctl *ctl,
|
||||
unsigned int off, void *buf, size_t len)
|
||||
{
|
||||
struct cs_dsp *dsp = ctl->dsp;
|
||||
int ret;
|
||||
|
||||
lockdep_assert_not_held(&dsp->pwr_lock);
|
||||
|
||||
mutex_lock(&dsp->pwr_lock);
|
||||
ret = cs_dsp_coeff_read_ctrl(ctl, off, buf, len);
|
||||
mutex_unlock(&dsp->pwr_lock);
|
||||
|
||||
return ret;
|
||||
}
|
||||
EXPORT_SYMBOL_GPL(cs_dsp_coeff_lock_and_read_ctrl);
|
||||
|
||||
static int cs_dsp_coeff_init_control_caches(struct cs_dsp *dsp)
|
||||
{
|
||||
struct cs_dsp_coeff_ctl *ctl;
|
||||
|
|
|
|||
|
|
@ -238,8 +238,12 @@ void cs_dsp_cleanup_debugfs(struct cs_dsp *dsp);
|
|||
int cs_dsp_coeff_write_acked_control(struct cs_dsp_coeff_ctl *ctl, unsigned int event_id);
|
||||
int cs_dsp_coeff_write_ctrl(struct cs_dsp_coeff_ctl *ctl, unsigned int off,
|
||||
const void *buf, size_t len);
|
||||
int cs_dsp_coeff_lock_and_write_ctrl(struct cs_dsp_coeff_ctl *ctl, unsigned int off,
|
||||
const void *buf, size_t len);
|
||||
int cs_dsp_coeff_read_ctrl(struct cs_dsp_coeff_ctl *ctl, unsigned int off,
|
||||
void *buf, size_t len);
|
||||
int cs_dsp_coeff_lock_and_read_ctrl(struct cs_dsp_coeff_ctl *ctl, unsigned int off,
|
||||
void *buf, size_t len);
|
||||
struct cs_dsp_coeff_ctl *cs_dsp_get_ctl(struct cs_dsp *dsp, const char *name, int type,
|
||||
unsigned int alg);
|
||||
|
||||
|
|
|
|||
|
|
@ -3109,6 +3109,7 @@
|
|||
#define PCI_DEVICE_ID_INTEL_HDA_CML_S 0xa3f0
|
||||
#define PCI_DEVICE_ID_INTEL_HDA_LNL_P 0xa828
|
||||
#define PCI_DEVICE_ID_INTEL_S21152BB 0xb152
|
||||
#define PCI_DEVICE_ID_INTEL_HDA_BMG 0xe2f7
|
||||
#define PCI_DEVICE_ID_INTEL_HDA_CML_R 0xf0c8
|
||||
#define PCI_DEVICE_ID_INTEL_HDA_RKL_S 0xf1c8
|
||||
|
||||
|
|
|
|||
|
|
@ -25,16 +25,6 @@ struct davinci_mcasp_pdata {
|
|||
unsigned sram_size_capture;
|
||||
struct gen_pool *sram_pool;
|
||||
|
||||
/*
|
||||
* If McBSP peripheral gets the clock from an external pin,
|
||||
* there are three chooses, that are MCBSP_CLKX, MCBSP_CLKR
|
||||
* and MCBSP_CLKS.
|
||||
* Depending on different hardware connections it is possible
|
||||
* to use this setting to change the behaviour of McBSP
|
||||
* driver.
|
||||
*/
|
||||
int clk_input_pin;
|
||||
|
||||
/*
|
||||
* This flag works when both clock and FS are outputs for the cpu
|
||||
* and makes clock more accurate (FS is not symmetrical and the
|
||||
|
|
@ -91,11 +81,6 @@ enum {
|
|||
MCASP_VERSION_OMAP, /* OMAP4/5 */
|
||||
};
|
||||
|
||||
enum mcbsp_clk_input_pin {
|
||||
MCBSP_CLKR = 0, /* as in DM365 */
|
||||
MCBSP_CLKS,
|
||||
};
|
||||
|
||||
#define INACTIVE_MODE 0
|
||||
#define TX_MODE 1
|
||||
#define RX_MODE 2
|
||||
|
|
|
|||
|
|
@ -167,6 +167,29 @@ snd_ctl_find_id_mixer(struct snd_card *card, const char *name)
|
|||
return snd_ctl_find_id(card, &id);
|
||||
}
|
||||
|
||||
/**
|
||||
* snd_ctl_find_id_mixer_locked - find the control instance with the given name string
|
||||
* @card: the card instance
|
||||
* @name: the name string
|
||||
*
|
||||
* Finds the control instance with the given name and
|
||||
* @SNDRV_CTL_ELEM_IFACE_MIXER. Other fields are set to zero.
|
||||
*
|
||||
* This is merely a wrapper to snd_ctl_find_id_locked().
|
||||
* The caller must down card->controls_rwsem before calling this function.
|
||||
*
|
||||
* Return: The pointer of the instance if found, or %NULL if not.
|
||||
*/
|
||||
static inline struct snd_kcontrol *
|
||||
snd_ctl_find_id_mixer_locked(struct snd_card *card, const char *name)
|
||||
{
|
||||
struct snd_ctl_elem_id id = {};
|
||||
|
||||
id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
|
||||
strscpy(id.name, name, sizeof(id.name));
|
||||
return snd_ctl_find_id_locked(card, &id);
|
||||
}
|
||||
|
||||
int snd_ctl_create(struct snd_card *card);
|
||||
|
||||
int snd_ctl_register_ioctl(snd_kctl_ioctl_func_t fcn);
|
||||
|
|
|
|||
|
|
@ -554,6 +554,11 @@
|
|||
#define CS35L41_LRCLK_FRC_SHIFT 1
|
||||
|
||||
#define CS35L41_AMP_GAIN_PCM_MASK 0x3E0
|
||||
#define CS35L41_AMP_GAIN_PCM_SHIFT 5
|
||||
#define CS35L41_AMP_GAIN_PDM_MASK 0x1F
|
||||
#define CS35L41_AMP_GAIN_PDM_SHIFT 0
|
||||
#define CS35L41_AMP_GAIN_PCM_MAX 20
|
||||
#define CS35L41_AMP_GAIN_PDM_MAX 20
|
||||
#define CS35L41_AMP_GAIN_ZC_MASK 0x0400
|
||||
#define CS35L41_AMP_GAIN_ZC_SHIFT 10
|
||||
|
||||
|
|
|
|||
|
|
@ -118,6 +118,7 @@ int snd_dmaengine_pcm_refine_runtime_hwparams(
|
|||
* which do not use devicetree.
|
||||
* @process: Callback used to apply processing on samples transferred from/to
|
||||
* user space.
|
||||
* @name: Component name. If null, dev_name will be used.
|
||||
* @compat_filter_fn: Will be used as the filter function when requesting a
|
||||
* channel for platforms which do not use devicetree. The filter parameter
|
||||
* will be the DAI's DMA data.
|
||||
|
|
@ -143,6 +144,7 @@ struct snd_dmaengine_pcm_config {
|
|||
int (*process)(struct snd_pcm_substream *substream,
|
||||
int channel, unsigned long hwoff,
|
||||
unsigned long bytes);
|
||||
const char *name;
|
||||
dma_filter_fn compat_filter_fn;
|
||||
struct device *dma_dev;
|
||||
const char *chan_names[SNDRV_PCM_STREAM_LAST + 1];
|
||||
|
|
|
|||
|
|
@ -598,17 +598,25 @@ SUB_REG(PEFE, FILTERAMOUNT, 0x000000ff) /* Filter envlope amount */
|
|||
// In stereo mode, the two channels' caches are concatenated into one,
|
||||
// and hold the interleaved frames.
|
||||
// The cache holds 64 frames, so the upper half is not used in 8-bit mode.
|
||||
// All registers mentioned below count in frames.
|
||||
// The cache is a ring buffer; CCR_READADDRESS operates modulo 64.
|
||||
// The cache is filled from (CCCA_CURRADDR - CCR_CACHEINVALIDSIZE)
|
||||
// into (CCR_READADDRESS - CCR_CACHEINVALIDSIZE).
|
||||
// All registers mentioned below count in frames. Shortcuts:
|
||||
// CA = CCCA_CURRADDR, CRA = CCR_READADDRESS,
|
||||
// CLA = CCR_CACHELOOPADDRHI:CLP_CACHELOOPADDR,
|
||||
// CIS = CCR_CACHEINVALIDSIZE, LIS = CCR_LOOPINVALSIZE,
|
||||
// CLF = CCR_CACHELOOPFLAG, LF = CCR_LOOPFLAG
|
||||
// The cache is a ring buffer; CRA operates modulo 64.
|
||||
// The cache is filled from (CA - CIS) into (CRA - CIS).
|
||||
// The engine has a fetch threshold of 32 bytes, so it tries to keep
|
||||
// CCR_CACHEINVALIDSIZE below 8 (16-bit stereo), 16 (16-bit mono,
|
||||
// 8-bit stereo), or 32 (8-bit mono). The actual transfers are pretty
|
||||
// unpredictable, especially if several voices are running.
|
||||
// Frames are consumed at CCR_READADDRESS, which is incremented afterwards,
|
||||
// along with CCCA_CURRADDR and CCR_CACHEINVALIDSIZE. This implies that the
|
||||
// actual playback position always lags CCCA_CURRADDR by exactly 64 frames.
|
||||
// CIS below 8 (16-bit stereo), 16 (16-bit mono, 8-bit stereo), or
|
||||
// 32 (8-bit mono). The actual transfers are pretty unpredictable,
|
||||
// especially if several voices are running.
|
||||
// Frames are consumed at CRA, which is incremented afterwards,
|
||||
// along with CA and CIS. This implies that the actual playback
|
||||
// position always lags CA by exactly 64 frames.
|
||||
// When CA reaches DSL_LOOPENDADDR, LF is set for one frame's time.
|
||||
// LF's rising edge causes the current values of CA and CIS to be
|
||||
// copied into CLA and LIS, resp., and CLF to be set.
|
||||
// If CLF is set, the first LIS of the CIS frames are instead
|
||||
// filled from (CLA - LIS), and CLF is subsequently reset.
|
||||
#define CD0 0x20 /* Cache data registers 0 .. 0x1f */
|
||||
|
||||
#define PTB 0x40 /* Page table base register */
|
||||
|
|
@ -1843,6 +1851,7 @@ void snd_emu1010_fpga_link_dst_src_write(struct snd_emu10k1 *emu, u32 dst, u32 s
|
|||
u32 snd_emu1010_fpga_link_dst_src_read(struct snd_emu10k1 *emu, u32 dst);
|
||||
int snd_emu1010_get_raw_rate(struct snd_emu10k1 *emu, u8 src);
|
||||
void snd_emu1010_update_clock(struct snd_emu10k1 *emu);
|
||||
void snd_emu1010_load_firmware_entry(struct snd_emu10k1 *emu, int dock, const struct firmware *fw_entry);
|
||||
unsigned int snd_emu10k1_efx_read(struct snd_emu10k1 *emu, unsigned int pc);
|
||||
void snd_emu10k1_intr_enable(struct snd_emu10k1 *emu, unsigned int intrenb);
|
||||
void snd_emu10k1_intr_disable(struct snd_emu10k1 *emu, unsigned int intrenb);
|
||||
|
|
@ -1885,8 +1894,8 @@ int snd_emu10k1_alloc_pages_maybe_wider(struct snd_emu10k1 *emu, size_t size,
|
|||
struct snd_dma_buffer *dmab);
|
||||
struct snd_util_memblk *snd_emu10k1_synth_alloc(struct snd_emu10k1 *emu, unsigned int size);
|
||||
int snd_emu10k1_synth_free(struct snd_emu10k1 *emu, struct snd_util_memblk *blk);
|
||||
int snd_emu10k1_synth_bzero(struct snd_emu10k1 *emu, struct snd_util_memblk *blk, int offset, int size);
|
||||
int snd_emu10k1_synth_copy_from_user(struct snd_emu10k1 *emu, struct snd_util_memblk *blk, int offset, const char __user *data, int size);
|
||||
int snd_emu10k1_synth_memset(struct snd_emu10k1 *emu, struct snd_util_memblk *blk, int offset, int size, u8 value);
|
||||
int snd_emu10k1_synth_copy_from_user(struct snd_emu10k1 *emu, struct snd_util_memblk *blk, int offset, const char __user *data, int size, u32 xor);
|
||||
int snd_emu10k1_memblk_map(struct snd_emu10k1 *emu, struct snd_emu10k1_memblk *blk);
|
||||
|
||||
/* voice allocation */
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
* This file is provided under a dual BSD/GPLv2 license. When using or
|
||||
* redistributing this file, you may do so under either license.
|
||||
*
|
||||
* Copyright(c) 2022-2023 Intel Corporation. All rights reserved.
|
||||
* Copyright(c) 2022-2023 Intel Corporation
|
||||
*/
|
||||
|
||||
struct hdac_bus;
|
||||
|
|
|
|||
|
|
@ -109,11 +109,9 @@ struct hda_codec_ops {
|
|||
void (*unsol_event)(struct hda_codec *codec, unsigned int res);
|
||||
void (*set_power_state)(struct hda_codec *codec, hda_nid_t fg,
|
||||
unsigned int power_state);
|
||||
#ifdef CONFIG_PM
|
||||
int (*suspend)(struct hda_codec *codec);
|
||||
int (*resume)(struct hda_codec *codec);
|
||||
int (*check_power_status)(struct hda_codec *codec, hda_nid_t nid);
|
||||
#endif
|
||||
void (*stream_pm)(struct hda_codec *codec, hda_nid_t nid, bool on);
|
||||
};
|
||||
|
||||
|
|
@ -259,11 +257,9 @@ struct hda_codec {
|
|||
unsigned int no_stream_clean_at_suspend:1; /* do not clean streams at suspend */
|
||||
unsigned int ctl_dev_id:1; /* old control element id build behaviour */
|
||||
|
||||
#ifdef CONFIG_PM
|
||||
unsigned long power_on_acct;
|
||||
unsigned long power_off_acct;
|
||||
unsigned long power_jiffies;
|
||||
#endif
|
||||
|
||||
/* filter the requested power state per nid */
|
||||
unsigned int (*power_filter)(struct hda_codec *codec, hda_nid_t nid,
|
||||
|
|
@ -481,10 +477,8 @@ extern const struct dev_pm_ops hda_codec_driver_pm;
|
|||
static inline
|
||||
int hda_call_check_power_status(struct hda_codec *codec, hda_nid_t nid)
|
||||
{
|
||||
#ifdef CONFIG_PM
|
||||
if (codec->patch_ops.check_power_status)
|
||||
return codec->patch_ops.check_power_status(codec, nid);
|
||||
#endif
|
||||
return 0;
|
||||
}
|
||||
|
||||
|
|
@ -495,14 +489,9 @@ int hda_call_check_power_status(struct hda_codec *codec, hda_nid_t nid)
|
|||
#define snd_hda_power_up_pm(codec) snd_hdac_power_up_pm(&(codec)->core)
|
||||
#define snd_hda_power_down(codec) snd_hdac_power_down(&(codec)->core)
|
||||
#define snd_hda_power_down_pm(codec) snd_hdac_power_down_pm(&(codec)->core)
|
||||
#ifdef CONFIG_PM
|
||||
void snd_hda_codec_set_power_save(struct hda_codec *codec, int delay);
|
||||
void snd_hda_set_power_save(struct hda_bus *bus, int delay);
|
||||
void snd_hda_update_power_acct(struct hda_codec *codec);
|
||||
#else
|
||||
static inline void snd_hda_codec_set_power_save(struct hda_codec *codec, int delay) {}
|
||||
static inline void snd_hda_set_power_save(struct hda_bus *bus, int delay) {}
|
||||
#endif
|
||||
|
||||
static inline bool hda_codec_need_resume(struct hda_codec *codec)
|
||||
{
|
||||
|
|
|
|||
|
|
@ -351,6 +351,7 @@ struct hdac_bus {
|
|||
bool needs_damn_long_delay:1;
|
||||
bool not_use_interrupts:1; /* prohibiting the RIRB IRQ */
|
||||
bool access_sdnctl_in_dword:1; /* accessing the sdnctl register by dword */
|
||||
bool use_pio_for_commands:1; /* Use PIO instead of CORB for commands */
|
||||
|
||||
int poll_count;
|
||||
|
||||
|
|
@ -731,6 +732,7 @@ static inline unsigned int snd_array_index(struct snd_array *array, void *ptr)
|
|||
{ PCI_DEVICE(PCI_VENDOR_ID_INTEL, PCI_DEVICE_ID_INTEL_HDA_DG2_0) }, \
|
||||
{ PCI_DEVICE(PCI_VENDOR_ID_INTEL, PCI_DEVICE_ID_INTEL_HDA_DG2_1) }, \
|
||||
{ PCI_DEVICE(PCI_VENDOR_ID_INTEL, PCI_DEVICE_ID_INTEL_HDA_DG2_2) }, \
|
||||
{ PCI_DEVICE(PCI_VENDOR_ID_INTEL, PCI_DEVICE_ID_INTEL_HDA_BMG) }, \
|
||||
{ } \
|
||||
}, pci) || HDA_CONTROLLER_IS_HSW(pci))
|
||||
|
||||
|
|
|
|||
|
|
@ -120,6 +120,8 @@ struct snd_pcm_ops {
|
|||
#define SNDRV_PCM_RATE_192000 (1U<<12) /* 192000Hz */
|
||||
#define SNDRV_PCM_RATE_352800 (1U<<13) /* 352800Hz */
|
||||
#define SNDRV_PCM_RATE_384000 (1U<<14) /* 384000Hz */
|
||||
#define SNDRV_PCM_RATE_705600 (1U<<15) /* 705600Hz */
|
||||
#define SNDRV_PCM_RATE_768000 (1U<<16) /* 768000Hz */
|
||||
|
||||
#define SNDRV_PCM_RATE_CONTINUOUS (1U<<30) /* continuous range */
|
||||
#define SNDRV_PCM_RATE_KNOT (1U<<31) /* supports more non-continuos rates */
|
||||
|
|
@ -135,6 +137,9 @@ struct snd_pcm_ops {
|
|||
#define SNDRV_PCM_RATE_8000_384000 (SNDRV_PCM_RATE_8000_192000|\
|
||||
SNDRV_PCM_RATE_352800|\
|
||||
SNDRV_PCM_RATE_384000)
|
||||
#define SNDRV_PCM_RATE_8000_768000 (SNDRV_PCM_RATE_8000_384000|\
|
||||
SNDRV_PCM_RATE_705600|\
|
||||
SNDRV_PCM_RATE_768000)
|
||||
#define _SNDRV_PCM_FMTBIT(fmt) (1ULL << (__force int)SNDRV_PCM_FORMAT_##fmt)
|
||||
#define SNDRV_PCM_FMTBIT_S8 _SNDRV_PCM_FMTBIT(S8)
|
||||
#define SNDRV_PCM_FMTBIT_U8 _SNDRV_PCM_FMTBIT(U8)
|
||||
|
|
|
|||
|
|
@ -1,6 +1,6 @@
|
|||
/* SPDX-License-Identifier: GPL-2.0
|
||||
*
|
||||
* Copyright (C) 2017, Intel Corporation. All rights reserved.
|
||||
* Copyright (C) 2017, Intel Corporation
|
||||
*/
|
||||
|
||||
#ifndef __LINUX_SND_SOC_ACPI_INTEL_MATCH_H
|
||||
|
|
|
|||
|
|
@ -3,8 +3,8 @@
|
|||
* Copyright(c) 2023 Intel Corporation.
|
||||
*/
|
||||
|
||||
#ifndef __SOF_SSP_COMMON_H
|
||||
#define __SOF_SSP_COMMON_H
|
||||
#ifndef __LINUX_SND_SOC_ACPI_INTEL_SSP_COMMON_H
|
||||
#define __LINUX_SND_SOC_ACPI_INTEL_SSP_COMMON_H
|
||||
|
||||
/* Cirrus Logic */
|
||||
#define CS35L41_ACPI_HID "CSC3541"
|
||||
|
|
@ -37,7 +37,7 @@
|
|||
#define RT5682_ACPI_HID "10EC5682"
|
||||
#define RT5682S_ACPI_HID "RTL5682"
|
||||
|
||||
enum sof_ssp_codec {
|
||||
enum snd_soc_acpi_intel_codec {
|
||||
CODEC_NONE,
|
||||
|
||||
/* headphone codec */
|
||||
|
|
@ -65,16 +65,17 @@ enum sof_ssp_codec {
|
|||
CODEC_RT1308,
|
||||
};
|
||||
|
||||
enum sof_ssp_codec sof_ssp_detect_codec_type(struct device *dev);
|
||||
enum sof_ssp_codec sof_ssp_detect_amp_type(struct device *dev);
|
||||
enum snd_soc_acpi_intel_codec
|
||||
snd_soc_acpi_intel_detect_codec_type(struct device *dev);
|
||||
enum snd_soc_acpi_intel_codec
|
||||
snd_soc_acpi_intel_detect_amp_type(struct device *dev);
|
||||
|
||||
#if IS_ENABLED(CONFIG_SND_SOC_INTEL_SOF_SSP_COMMON)
|
||||
const char *sof_ssp_get_codec_name(enum sof_ssp_codec codec_type);
|
||||
#else
|
||||
static inline const char *sof_ssp_get_codec_name(enum sof_ssp_codec codec_type)
|
||||
{
|
||||
return NULL;
|
||||
}
|
||||
#endif
|
||||
const char *
|
||||
snd_soc_acpi_intel_get_codec_name(enum snd_soc_acpi_intel_codec codec_type);
|
||||
|
||||
#endif /* __SOF_SSP_COMMON_H */
|
||||
const char *
|
||||
snd_soc_acpi_intel_get_codec_tplg_suffix(enum snd_soc_acpi_intel_codec codec_type);
|
||||
const char *
|
||||
snd_soc_acpi_intel_get_amp_tplg_suffix(enum snd_soc_acpi_intel_codec codec_type);
|
||||
|
||||
#endif /* __LINUX_SND_SOC_ACPI_INTEL_SSP_COMMON_H */
|
||||
|
|
@ -1,6 +1,6 @@
|
|||
/* SPDX-License-Identifier: GPL-2.0-only
|
||||
*
|
||||
* Copyright (C) 2013-15, Intel Corporation. All rights reserved.
|
||||
* Copyright (C) 2013-15, Intel Corporation
|
||||
*/
|
||||
|
||||
#ifndef __LINUX_SND_SOC_ACPI_H
|
||||
|
|
@ -151,6 +151,18 @@ struct snd_soc_acpi_link_adr {
|
|||
*/
|
||||
#define SND_SOC_ACPI_TPLG_INTEL_DMIC_NUMBER BIT(2)
|
||||
|
||||
/*
|
||||
* when set the speaker amplifier name suffix (i.e. "-max98360a") will be
|
||||
* appended to topology file name
|
||||
*/
|
||||
#define SND_SOC_ACPI_TPLG_INTEL_AMP_NAME BIT(3)
|
||||
|
||||
/*
|
||||
* when set the headphone codec name suffix (i.e. "-rt5682") will be appended to
|
||||
* topology file name
|
||||
*/
|
||||
#define SND_SOC_ACPI_TPLG_INTEL_CODEC_NAME BIT(4)
|
||||
|
||||
/**
|
||||
* snd_soc_acpi_mach: ACPI-based machine descriptor. Most of the fields are
|
||||
* related to the hardware, except for the firmware and topology file names.
|
||||
|
|
|
|||
|
|
@ -44,7 +44,6 @@ struct snd_soc_jack_zone {
|
|||
/**
|
||||
* struct snd_soc_jack_gpio - Describes a gpio pin for jack detection
|
||||
*
|
||||
* @gpio: legacy gpio number
|
||||
* @idx: gpio descriptor index within the function of the GPIO
|
||||
* consumer device
|
||||
* @gpiod_dev: GPIO consumer device
|
||||
|
|
@ -59,7 +58,6 @@ struct snd_soc_jack_zone {
|
|||
* ADC).
|
||||
*/
|
||||
struct snd_soc_jack_gpio {
|
||||
unsigned int gpio;
|
||||
unsigned int idx;
|
||||
struct device *gpiod_dev;
|
||||
const char *name;
|
||||
|
|
|
|||
|
|
@ -11,20 +11,30 @@
|
|||
#define __LINUX_SND_SOC_H
|
||||
|
||||
#include <linux/args.h>
|
||||
#include <linux/of.h>
|
||||
#include <linux/platform_device.h>
|
||||
#include <linux/types.h>
|
||||
#include <linux/notifier.h>
|
||||
#include <linux/workqueue.h>
|
||||
#include <linux/array_size.h>
|
||||
#include <linux/device.h>
|
||||
#include <linux/errno.h>
|
||||
#include <linux/interrupt.h>
|
||||
#include <linux/kernel.h>
|
||||
#include <linux/regmap.h>
|
||||
#include <linux/lockdep.h>
|
||||
#include <linux/log2.h>
|
||||
#include <sound/core.h>
|
||||
#include <sound/pcm.h>
|
||||
#include <linux/mutex.h>
|
||||
#include <linux/notifier.h>
|
||||
#include <linux/of.h>
|
||||
#include <linux/types.h>
|
||||
#include <linux/workqueue.h>
|
||||
|
||||
#include <sound/ac97_codec.h>
|
||||
#include <sound/compress_driver.h>
|
||||
#include <sound/control.h>
|
||||
#include <sound/ac97_codec.h>
|
||||
#include <sound/core.h>
|
||||
#include <sound/pcm.h>
|
||||
|
||||
struct module;
|
||||
struct platform_device;
|
||||
|
||||
/* For the current users of sound/soc.h to avoid build issues */
|
||||
#include <linux/platform_device.h>
|
||||
#include <linux/regmap.h>
|
||||
|
||||
/*
|
||||
* Convenience kcontrol builders
|
||||
|
|
@ -149,6 +159,18 @@
|
|||
{.reg = xreg, .rreg = xreg, \
|
||||
.shift = shift_left, .rshift = shift_right, \
|
||||
.max = xmax, .min = xmin} }
|
||||
#define SOC_DOUBLE_RANGE_TLV(xname, xreg, xshift_left, xshift_right, xmin, xmax, \
|
||||
xinvert, tlv_array) \
|
||||
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
|
||||
.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
|
||||
SNDRV_CTL_ELEM_ACCESS_READWRITE,\
|
||||
.tlv.p = (tlv_array), \
|
||||
.info = snd_soc_info_volsw, \
|
||||
.get = snd_soc_get_volsw, .put = snd_soc_put_volsw, \
|
||||
.private_value = (unsigned long)&(struct soc_mixer_control) \
|
||||
{.reg = xreg, .rreg = xreg, \
|
||||
.shift = xshift_left, .rshift = xshift_right, \
|
||||
.min = xmin, .max = xmax, .invert = xinvert} }
|
||||
#define SOC_DOUBLE_R_TLV(xname, reg_left, reg_right, xshift, xmax, xinvert, tlv_array) \
|
||||
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
|
||||
.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
|
||||
|
|
@ -400,7 +422,6 @@
|
|||
#define SOC_ENUM_SINGLE_VIRT_DECL(name, xtexts) \
|
||||
const struct soc_enum name = SOC_ENUM_SINGLE_VIRT(ARRAY_SIZE(xtexts), xtexts)
|
||||
|
||||
struct device_node;
|
||||
struct snd_jack;
|
||||
struct snd_soc_card;
|
||||
struct snd_soc_pcm_stream;
|
||||
|
|
@ -415,6 +436,7 @@ struct soc_enum;
|
|||
struct snd_soc_jack;
|
||||
struct snd_soc_jack_zone;
|
||||
struct snd_soc_jack_pin;
|
||||
|
||||
#include <sound/soc-dapm.h>
|
||||
#include <sound/soc-dpcm.h>
|
||||
#include <sound/soc-topology.h>
|
||||
|
|
@ -1193,8 +1215,12 @@ struct snd_soc_pcm_runtime {
|
|||
/* see soc_new_pcm_runtime() */
|
||||
#define snd_soc_rtd_to_cpu(rtd, n) (rtd)->dais[n]
|
||||
#define snd_soc_rtd_to_codec(rtd, n) (rtd)->dais[n + (rtd)->dai_link->num_cpus]
|
||||
#define snd_soc_substream_to_rtd(substream) \
|
||||
(struct snd_soc_pcm_runtime *)snd_pcm_substream_chip(substream)
|
||||
|
||||
static inline struct snd_soc_pcm_runtime *
|
||||
snd_soc_substream_to_rtd(const struct snd_pcm_substream *substream)
|
||||
{
|
||||
return snd_pcm_substream_chip(substream);
|
||||
}
|
||||
|
||||
#define for_each_rtd_components(rtd, i, component) \
|
||||
for ((i) = 0, component = NULL; \
|
||||
|
|
@ -1213,6 +1239,10 @@ struct snd_soc_pcm_runtime {
|
|||
((i) < (rtd)->dai_link->num_cpus + (rtd)->dai_link->num_codecs) && \
|
||||
((dai) = (rtd)->dais[i]); \
|
||||
(i)++)
|
||||
#define for_each_rtd_dais_reverse(rtd, i, dai) \
|
||||
for ((i) = (rtd)->dai_link->num_cpus + (rtd)->dai_link->num_codecs - 1; \
|
||||
(i) >= 0 && ((dai) = (rtd)->dais[i]); \
|
||||
(i)--)
|
||||
#define for_each_rtd_ch_maps(rtd, i, ch_maps) for_each_link_ch_maps(rtd->dai_link, i, ch_maps)
|
||||
|
||||
void snd_soc_close_delayed_work(struct snd_soc_pcm_runtime *rtd);
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
* This file is provided under a dual BSD/GPLv2 license. When using or
|
||||
* redistributing this file, you may do so under either license.
|
||||
*
|
||||
* Copyright(c) 2018 Intel Corporation. All rights reserved.
|
||||
* Copyright(c) 2018 Intel Corporation
|
||||
*
|
||||
* Author: Liam Girdwood <liam.r.girdwood@linux.intel.com>
|
||||
*/
|
||||
|
|
@ -166,7 +166,7 @@ struct sof_dev_desc {
|
|||
/* default firmware name */
|
||||
const char *default_fw_filename[SOF_IPC_TYPE_COUNT];
|
||||
|
||||
struct snd_sof_dsp_ops *ops;
|
||||
const struct snd_sof_dsp_ops *ops;
|
||||
int (*ops_init)(struct snd_sof_dev *sdev);
|
||||
void (*ops_free)(struct snd_sof_dev *sdev);
|
||||
};
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
* This file is provided under a dual BSD/GPLv2 license. When using or
|
||||
* redistributing this file, you may do so under either license.
|
||||
*
|
||||
* Copyright(c) 2019 Intel Corporation. All rights reserved.
|
||||
* Copyright(c) 2019 Intel Corporation
|
||||
*/
|
||||
|
||||
#ifndef __IPC_CHANNEL_MAP_H__
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
* This file is provided under a dual BSD/GPLv2 license. When using or
|
||||
* redistributing this file, you may do so under either license.
|
||||
*
|
||||
* Copyright(c) 2018 Intel Corporation. All rights reserved.
|
||||
* Copyright(c) 2018 Intel Corporation
|
||||
*/
|
||||
|
||||
#ifndef __INCLUDE_SOUND_SOF_CONTROL_H__
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
* This file is provided under a dual BSD/GPLv2 license. When using or
|
||||
* redistributing this file, you may do so under either license.
|
||||
*
|
||||
* Copyright(c) 2018 Intel Corporation. All rights reserved.
|
||||
* Copyright(c) 2018 Intel Corporation
|
||||
*/
|
||||
|
||||
#ifndef __INCLUDE_SOUND_SOF_DAI_INTEL_H__
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
* This file is provided under a dual BSD/GPLv2 license. When using or
|
||||
* redistributing this file, you may do so under either license.
|
||||
*
|
||||
* Copyright(c) 2018 Intel Corporation. All rights reserved.
|
||||
* Copyright(c) 2018 Intel Corporation
|
||||
*/
|
||||
|
||||
#ifndef __INCLUDE_SOUND_SOF_DAI_H__
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
* This file is provided under a dual BSD/GPLv2 license. When using or
|
||||
* redistributing this file, you may do so under either license.
|
||||
*
|
||||
* Copyright(c) 2020 Intel Corporation. All rights reserved.
|
||||
* Copyright(c) 2020 Intel Corporation
|
||||
*
|
||||
* Author: Karol Trzcinski <karolx.trzcinski@linux.intel.com>
|
||||
*/
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
* This file is provided under a dual BSD/GPLv2 license. When using or
|
||||
* redistributing this file, you may do so under either license.
|
||||
*
|
||||
* Copyright(c) 2020 Intel Corporation. All rights reserved.
|
||||
* Copyright(c) 2020 Intel Corporation
|
||||
*/
|
||||
|
||||
/*
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
* This file is provided under a dual BSD/GPLv2 license. When using or
|
||||
* redistributing this file, you may do so under either license.
|
||||
*
|
||||
* Copyright(c) 2022 Intel Corporation. All rights reserved.
|
||||
* Copyright(c) 2022 Intel Corporation
|
||||
*/
|
||||
|
||||
/*
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
* This file is provided under a dual BSD/GPLv2 license. When using or
|
||||
* redistributing this file, you may do so under either license.
|
||||
*
|
||||
* Copyright(c) 2018 Intel Corporation. All rights reserved.
|
||||
* Copyright(c) 2018 Intel Corporation
|
||||
*/
|
||||
|
||||
#ifndef __INCLUDE_SOUND_SOF_HEADER_H__
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
* This file is provided under a dual BSD/GPLv2 license. When using or
|
||||
* redistributing this file, you may do so under either license.
|
||||
*
|
||||
* Copyright(c) 2018 Intel Corporation. All rights reserved.
|
||||
* Copyright(c) 2018 Intel Corporation
|
||||
*/
|
||||
|
||||
#ifndef __INCLUDE_SOUND_SOF_INFO_H__
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
* This file is provided under a dual BSD/GPLv2 license. When using or
|
||||
* redistributing this file, you may do so under either license.
|
||||
*
|
||||
* Copyright(c) 2022 Intel Corporation. All rights reserved.
|
||||
* Copyright(c) 2022 Intel Corporation
|
||||
*/
|
||||
|
||||
#ifndef __INCLUDE_SOUND_SOF_IPC4_HEADER_H__
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
* This file is provided under a dual BSD/GPLv2 license. When using or
|
||||
* redistributing this file, you may do so under either license.
|
||||
*
|
||||
* Copyright(c) 2018 Intel Corporation. All rights reserved.
|
||||
* Copyright(c) 2018 Intel Corporation
|
||||
*/
|
||||
|
||||
#ifndef __INCLUDE_SOUND_SOF_PM_H__
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
* This file is provided under a dual BSD/GPLv2 license. When using or
|
||||
* redistributing this file, you may do so under either license.
|
||||
*
|
||||
* Copyright(c) 2018 Intel Corporation. All rights reserved.
|
||||
* Copyright(c) 2018 Intel Corporation
|
||||
*/
|
||||
|
||||
#ifndef __INCLUDE_SOUND_SOF_STREAM_H__
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
* This file is provided under a dual BSD/GPLv2 license. When using or
|
||||
* redistributing this file, you may do so under either license.
|
||||
*
|
||||
* Copyright(c) 2018 Intel Corporation. All rights reserved.
|
||||
* Copyright(c) 2018 Intel Corporation
|
||||
*/
|
||||
|
||||
#ifndef __INCLUDE_SOUND_SOF_TOPOLOGY_H__
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
* This file is provided under a dual BSD/GPLv2 license. When using or
|
||||
* redistributing this file, you may do so under either license.
|
||||
*
|
||||
* Copyright(c) 2018 Intel Corporation. All rights reserved.
|
||||
* Copyright(c) 2018 Intel Corporation
|
||||
*/
|
||||
|
||||
#ifndef __INCLUDE_SOUND_SOF_TRACE_H__
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
* This file is provided under a dual BSD/GPLv2 license. When using or
|
||||
* redistributing this file, you may do so under either license.
|
||||
*
|
||||
* Copyright(c) 2018 Intel Corporation. All rights reserved.
|
||||
* Copyright(c) 2018 Intel Corporation
|
||||
*/
|
||||
|
||||
#ifndef __INCLUDE_SOUND_SOF_XTENSA_H__
|
||||
|
|
|
|||
|
|
@ -89,7 +89,7 @@ struct snd_sf_list {
|
|||
int snd_soundfont_load(struct snd_sf_list *sflist, const void __user *data,
|
||||
long count, int client);
|
||||
int snd_soundfont_load_guspatch(struct snd_sf_list *sflist, const char __user *data,
|
||||
long count, int client);
|
||||
long count);
|
||||
int snd_soundfont_close_check(struct snd_sf_list *sflist, int client);
|
||||
|
||||
struct snd_sf_list *snd_sf_new(struct snd_sf_callback *callback,
|
||||
|
|
|
|||
|
|
@ -12,6 +12,8 @@
|
|||
#define DAPM_DIRECT "(direct)"
|
||||
#define DAPM_ARROW(dir) (((dir) == SND_SOC_DAPM_DIR_OUT) ? "->" : "<-")
|
||||
|
||||
TRACE_DEFINE_ENUM(SND_SOC_DAPM_DIR_OUT);
|
||||
|
||||
struct snd_soc_jack;
|
||||
struct snd_soc_card;
|
||||
struct snd_soc_dapm_widget;
|
||||
|
|
|
|||
|
|
@ -1,6 +1,6 @@
|
|||
/* SPDX-License-Identifier: GPL-2.0-only */
|
||||
/*
|
||||
* Copyright(c) 2022 Intel Corporation. All rights reserved.
|
||||
* Copyright(c) 2022 Intel Corporation
|
||||
*
|
||||
* Author: Noah Klayman <noah.klayman@intel.com>
|
||||
*/
|
||||
|
|
|
|||
|
|
@ -1,6 +1,6 @@
|
|||
/* SPDX-License-Identifier: GPL-2.0-only */
|
||||
/*
|
||||
* Copyright(c) 2022 Intel Corporation. All rights reserved.
|
||||
* Copyright(c) 2022 Intel Corporation
|
||||
*
|
||||
* Author: Noah Klayman <noah.klayman@intel.com>
|
||||
*/
|
||||
|
|
|
|||
|
|
@ -576,60 +576,4 @@ struct snd_soc_tplg_dai {
|
|||
struct snd_soc_tplg_private priv;
|
||||
} __attribute__((packed));
|
||||
|
||||
/*
|
||||
* Old version of ABI structs, supported for backward compatibility.
|
||||
*/
|
||||
|
||||
/* Manifest v4 */
|
||||
struct snd_soc_tplg_manifest_v4 {
|
||||
__le32 size; /* in bytes of this structure */
|
||||
__le32 control_elems; /* number of control elements */
|
||||
__le32 widget_elems; /* number of widget elements */
|
||||
__le32 graph_elems; /* number of graph elements */
|
||||
__le32 pcm_elems; /* number of PCM elements */
|
||||
__le32 dai_link_elems; /* number of DAI link elements */
|
||||
struct snd_soc_tplg_private priv;
|
||||
} __packed;
|
||||
|
||||
/* Stream Capabilities v4 */
|
||||
struct snd_soc_tplg_stream_caps_v4 {
|
||||
__le32 size; /* in bytes of this structure */
|
||||
char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
|
||||
__le64 formats; /* supported formats SNDRV_PCM_FMTBIT_* */
|
||||
__le32 rates; /* supported rates SNDRV_PCM_RATE_* */
|
||||
__le32 rate_min; /* min rate */
|
||||
__le32 rate_max; /* max rate */
|
||||
__le32 channels_min; /* min channels */
|
||||
__le32 channels_max; /* max channels */
|
||||
__le32 periods_min; /* min number of periods */
|
||||
__le32 periods_max; /* max number of periods */
|
||||
__le32 period_size_min; /* min period size bytes */
|
||||
__le32 period_size_max; /* max period size bytes */
|
||||
__le32 buffer_size_min; /* min buffer size bytes */
|
||||
__le32 buffer_size_max; /* max buffer size bytes */
|
||||
} __packed;
|
||||
|
||||
/* PCM v4 */
|
||||
struct snd_soc_tplg_pcm_v4 {
|
||||
__le32 size; /* in bytes of this structure */
|
||||
char pcm_name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
|
||||
char dai_name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
|
||||
__le32 pcm_id; /* unique ID - used to match with DAI link */
|
||||
__le32 dai_id; /* unique ID - used to match */
|
||||
__le32 playback; /* supports playback mode */
|
||||
__le32 capture; /* supports capture mode */
|
||||
__le32 compress; /* 1 = compressed; 0 = PCM */
|
||||
struct snd_soc_tplg_stream stream[SND_SOC_TPLG_STREAM_CONFIG_MAX]; /* for DAI link */
|
||||
__le32 num_streams; /* number of streams */
|
||||
struct snd_soc_tplg_stream_caps_v4 caps[2]; /* playback and capture for DAI */
|
||||
} __packed;
|
||||
|
||||
/* Physical link config v4 */
|
||||
struct snd_soc_tplg_link_config_v4 {
|
||||
__le32 size; /* in bytes of this structure */
|
||||
__le32 id; /* unique ID - used to match */
|
||||
struct snd_soc_tplg_stream stream[SND_SOC_TPLG_STREAM_CONFIG_MAX]; /* supported configs playback and captrure */
|
||||
__le32 num_streams; /* number of streams */
|
||||
} __packed;
|
||||
|
||||
#endif
|
||||
|
|
|
|||
|
|
@ -1,6 +1,6 @@
|
|||
/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */
|
||||
/*
|
||||
* Copyright(c) 2021 Intel Corporation. All rights reserved.
|
||||
* Copyright(c) 2021 Intel Corporation
|
||||
*
|
||||
* Authors: Cezary Rojewski <cezary.rojewski@intel.com>
|
||||
* Amadeusz Slawinski <amadeuszx.slawinski@linux.intel.com>
|
||||
|
|
|
|||
|
|
@ -165,78 +165,4 @@ enum skl_tuple_type {
|
|||
SKL_TYPE_DATA
|
||||
};
|
||||
|
||||
/* v4 configuration data */
|
||||
|
||||
struct skl_dfw_v4_module_pin {
|
||||
__u16 module_id;
|
||||
__u16 instance_id;
|
||||
} __packed;
|
||||
|
||||
struct skl_dfw_v4_module_fmt {
|
||||
__u32 channels;
|
||||
__u32 freq;
|
||||
__u32 bit_depth;
|
||||
__u32 valid_bit_depth;
|
||||
__u32 ch_cfg;
|
||||
__u32 interleaving_style;
|
||||
__u32 sample_type;
|
||||
__u32 ch_map;
|
||||
} __packed;
|
||||
|
||||
struct skl_dfw_v4_module_caps {
|
||||
__u32 set_params:2;
|
||||
__u32 rsvd:30;
|
||||
__u32 param_id;
|
||||
__u32 caps_size;
|
||||
__u32 caps[HDA_SST_CFG_MAX];
|
||||
} __packed;
|
||||
|
||||
struct skl_dfw_v4_pipe {
|
||||
__u8 pipe_id;
|
||||
__u8 pipe_priority;
|
||||
__u16 conn_type:4;
|
||||
__u16 rsvd:4;
|
||||
__u16 memory_pages:8;
|
||||
} __packed;
|
||||
|
||||
struct skl_dfw_v4_module {
|
||||
char uuid[SKL_UUID_STR_SZ];
|
||||
|
||||
__u16 module_id;
|
||||
__u16 instance_id;
|
||||
__u32 max_mcps;
|
||||
__u32 mem_pages;
|
||||
__u32 obs;
|
||||
__u32 ibs;
|
||||
__u32 vbus_id;
|
||||
|
||||
__u32 max_in_queue:8;
|
||||
__u32 max_out_queue:8;
|
||||
__u32 time_slot:8;
|
||||
__u32 core_id:4;
|
||||
__u32 rsvd1:4;
|
||||
|
||||
__u32 module_type:8;
|
||||
__u32 conn_type:4;
|
||||
__u32 dev_type:4;
|
||||
__u32 hw_conn_type:4;
|
||||
__u32 rsvd2:12;
|
||||
|
||||
__u32 params_fixup:8;
|
||||
__u32 converter:8;
|
||||
__u32 input_pin_type:1;
|
||||
__u32 output_pin_type:1;
|
||||
__u32 is_dynamic_in_pin:1;
|
||||
__u32 is_dynamic_out_pin:1;
|
||||
__u32 is_loadable:1;
|
||||
__u32 rsvd3:11;
|
||||
|
||||
struct skl_dfw_v4_pipe pipe;
|
||||
struct skl_dfw_v4_module_fmt in_fmt[MAX_IN_QUEUE];
|
||||
struct skl_dfw_v4_module_fmt out_fmt[MAX_OUT_QUEUE];
|
||||
struct skl_dfw_v4_module_pin in_pin[MAX_IN_QUEUE];
|
||||
struct skl_dfw_v4_module_pin out_pin[MAX_OUT_QUEUE];
|
||||
struct skl_dfw_v4_module_caps caps;
|
||||
} __packed;
|
||||
|
||||
#endif
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
* This file is provided under a dual BSD/GPLv2 license. When using or
|
||||
* redistributing this file, you may do so under either license.
|
||||
*
|
||||
* Copyright(c) 2018 Intel Corporation. All rights reserved.
|
||||
* Copyright(c) 2018 Intel Corporation
|
||||
*/
|
||||
|
||||
/**
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
* This file is provided under a dual BSD/GPLv2 license. When using or
|
||||
* redistributing this file, you may do so under either license.
|
||||
*
|
||||
* Copyright(c) 2018 Intel Corporation. All rights reserved.
|
||||
* Copyright(c) 2018 Intel Corporation
|
||||
*/
|
||||
|
||||
/*
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
* This file is provided under a dual BSD/GPLv2 license. When using or
|
||||
* redistributing this file, you may do so under either license.
|
||||
*
|
||||
* Copyright(c) 2018 Intel Corporation. All rights reserved.
|
||||
* Copyright(c) 2018 Intel Corporation
|
||||
*/
|
||||
|
||||
#ifndef __INCLUDE_UAPI_SOUND_SOF_USER_HEADER_H__
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
* This file is provided under a dual BSD/GPLv2 license. When using or
|
||||
* redistributing this file, you may do so under either license.
|
||||
*
|
||||
* Copyright(c) 2018 Intel Corporation. All rights reserved.
|
||||
* Copyright(c) 2018 Intel Corporation
|
||||
* Author: Liam Girdwood <liam.r.girdwood@linux.intel.com>
|
||||
* Keyon Jie <yang.jie@linux.intel.com>
|
||||
*/
|
||||
|
|
|
|||
|
|
@ -17,4 +17,4 @@ ifeq ($(CONFIG_SND),y)
|
|||
obj-y += last.o
|
||||
endif
|
||||
|
||||
soundcore-objs := sound_core.o
|
||||
soundcore-y := sound_core.o
|
||||
|
|
|
|||
|
|
@ -551,5 +551,6 @@ static void __exit ac97_bus_exit(void)
|
|||
}
|
||||
module_exit(ac97_bus_exit);
|
||||
|
||||
MODULE_DESCRIPTION("AC97 bus interface");
|
||||
MODULE_LICENSE("GPL");
|
||||
MODULE_AUTHOR("Robert Jarzmik <robert.jarzmik@free.fr>");
|
||||
|
|
|
|||
|
|
@ -95,4 +95,5 @@ module_exit(ac97_bus_exit);
|
|||
|
||||
EXPORT_SYMBOL(ac97_bus_type);
|
||||
|
||||
MODULE_DESCRIPTION("Legacy AC97 bus interface");
|
||||
MODULE_LICENSE("GPL");
|
||||
|
|
|
|||
|
|
@ -1,7 +1,7 @@
|
|||
# SPDX-License-Identifier: GPL-2.0
|
||||
snd-aoa-codec-onyx-objs := onyx.o
|
||||
snd-aoa-codec-tas-objs := tas.o
|
||||
snd-aoa-codec-toonie-objs := toonie.o
|
||||
snd-aoa-codec-onyx-y := onyx.o
|
||||
snd-aoa-codec-tas-y := tas.o
|
||||
snd-aoa-codec-toonie-y := toonie.o
|
||||
|
||||
obj-$(CONFIG_SND_AOA_ONYX) += snd-aoa-codec-onyx.o
|
||||
obj-$(CONFIG_SND_AOA_TAS) += snd-aoa-codec-tas.o
|
||||
|
|
|
|||
|
|
@ -1,6 +1,6 @@
|
|||
# SPDX-License-Identifier: GPL-2.0
|
||||
obj-$(CONFIG_SND_AOA) += snd-aoa.o
|
||||
snd-aoa-objs := core.o \
|
||||
snd-aoa-y := core.o \
|
||||
alsa.o \
|
||||
gpio-pmf.o \
|
||||
gpio-feature.o
|
||||
|
|
|
|||
|
|
@ -1,4 +1,4 @@
|
|||
# SPDX-License-Identifier: GPL-2.0-only
|
||||
snd-aoa-fabric-layout-objs += layout.o
|
||||
snd-aoa-fabric-layout-y += layout.o
|
||||
|
||||
obj-$(CONFIG_SND_AOA_FABRIC_LAYOUT) += snd-aoa-fabric-layout.o
|
||||
|
|
|
|||
|
|
@ -1,4 +1,4 @@
|
|||
# SPDX-License-Identifier: GPL-2.0-only
|
||||
obj-$(CONFIG_SND_AOA_SOUNDBUS) += snd-aoa-soundbus.o
|
||||
snd-aoa-soundbus-objs := core.o sysfs.o
|
||||
snd-aoa-soundbus-y := core.o sysfs.o
|
||||
obj-$(CONFIG_SND_AOA_SOUNDBUS_I2S) += i2sbus/
|
||||
|
|
|
|||
|
|
@ -1,3 +1,3 @@
|
|||
# SPDX-License-Identifier: GPL-2.0-only
|
||||
obj-$(CONFIG_SND_AOA_SOUNDBUS_I2S) += snd-aoa-i2sbus.o
|
||||
snd-aoa-i2sbus-objs := core.o pcm.o control.o
|
||||
snd-aoa-i2sbus-y := core.o pcm.o control.o
|
||||
|
|
|
|||
|
|
@ -255,24 +255,24 @@ static void i2sbus_wait_for_stop(struct i2sbus_dev *i2sdev,
|
|||
{
|
||||
unsigned long flags;
|
||||
DECLARE_COMPLETION_ONSTACK(done);
|
||||
long timeout;
|
||||
unsigned long time_left;
|
||||
|
||||
spin_lock_irqsave(&i2sdev->low_lock, flags);
|
||||
if (pi->dbdma_ring.stopping) {
|
||||
pi->stop_completion = &done;
|
||||
spin_unlock_irqrestore(&i2sdev->low_lock, flags);
|
||||
timeout = wait_for_completion_timeout(&done, HZ);
|
||||
time_left = wait_for_completion_timeout(&done, HZ);
|
||||
spin_lock_irqsave(&i2sdev->low_lock, flags);
|
||||
pi->stop_completion = NULL;
|
||||
if (timeout == 0) {
|
||||
if (time_left == 0) {
|
||||
/* timeout expired, stop dbdma forcefully */
|
||||
printk(KERN_ERR "i2sbus_wait_for_stop: timed out\n");
|
||||
/* make sure RUN, PAUSE and S0 bits are cleared */
|
||||
out_le32(&pi->dbdma->control, (RUN | PAUSE | 1) << 16);
|
||||
pi->dbdma_ring.stopping = 0;
|
||||
timeout = 10;
|
||||
time_left = 10;
|
||||
while (in_le32(&pi->dbdma->status) & ACTIVE) {
|
||||
if (--timeout <= 0)
|
||||
if (--time_left <= 0)
|
||||
break;
|
||||
udelay(1);
|
||||
}
|
||||
|
|
|
|||
|
|
@ -4,11 +4,11 @@
|
|||
#
|
||||
|
||||
obj-$(CONFIG_SND_ARMAACI) += snd-aaci.o
|
||||
snd-aaci-objs := aaci.o
|
||||
snd-aaci-y := aaci.o
|
||||
|
||||
obj-$(CONFIG_SND_PXA2XX_LIB) += snd-pxa2xx-lib.o
|
||||
snd-pxa2xx-lib-y := pxa2xx-pcm-lib.o
|
||||
snd-pxa2xx-lib-$(CONFIG_SND_PXA2XX_LIB_AC97) += pxa2xx-ac97-lib.o
|
||||
|
||||
obj-$(CONFIG_SND_PXA2XX_AC97) += snd-pxa2xx-ac97.o
|
||||
snd-pxa2xx-ac97-objs := pxa2xx-ac97.o
|
||||
snd-pxa2xx-ac97-y := pxa2xx-ac97.o
|
||||
|
|
|
|||
|
|
@ -33,7 +33,7 @@ int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream,
|
|||
struct snd_pcm_hw_params *params)
|
||||
{
|
||||
struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream);
|
||||
struct snd_soc_pcm_runtime *rtd = substream->private_data;
|
||||
struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
|
||||
struct snd_dmaengine_dai_dma_data *dma_params;
|
||||
struct dma_slave_config config;
|
||||
int ret;
|
||||
|
|
@ -79,7 +79,7 @@ EXPORT_SYMBOL(pxa2xx_pcm_prepare);
|
|||
|
||||
int pxa2xx_pcm_open(struct snd_pcm_substream *substream)
|
||||
{
|
||||
struct snd_soc_pcm_runtime *rtd = substream->private_data;
|
||||
struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
|
||||
struct snd_pcm_runtime *runtime = substream->runtime;
|
||||
struct snd_dmaengine_dai_dma_data *dma_params;
|
||||
int ret;
|
||||
|
|
|
|||
|
|
@ -1,4 +1,4 @@
|
|||
# SPDX-License-Identifier: GPL-2.0-only
|
||||
snd-atmel-ac97c-objs := ac97c.o
|
||||
snd-atmel-ac97c-y := ac97c.o
|
||||
|
||||
obj-$(CONFIG_SND_ATMEL_AC97C) += snd-atmel-ac97c.o
|
||||
|
|
|
|||
|
|
@ -24,18 +24,18 @@ snd-pcm-$(CONFIG_SND_PCM_IEC958) += pcm_iec958.o
|
|||
CFLAGS_pcm_lib.o := -I$(src)
|
||||
CFLAGS_pcm_native.o := -I$(src)
|
||||
|
||||
snd-pcm-dmaengine-objs := pcm_dmaengine.o
|
||||
snd-pcm-dmaengine-y := pcm_dmaengine.o
|
||||
|
||||
snd-ctl-led-objs := control_led.o
|
||||
snd-rawmidi-objs := rawmidi.o
|
||||
snd-ump-objs := ump.o
|
||||
snd-ctl-led-y := control_led.o
|
||||
snd-rawmidi-y := rawmidi.o
|
||||
snd-ump-y := ump.o
|
||||
snd-ump-$(CONFIG_SND_UMP_LEGACY_RAWMIDI) += ump_convert.o
|
||||
snd-timer-objs := timer.o
|
||||
snd-hrtimer-objs := hrtimer.o
|
||||
snd-hwdep-objs := hwdep.o
|
||||
snd-seq-device-objs := seq_device.o
|
||||
snd-timer-y := timer.o
|
||||
snd-hrtimer-y := hrtimer.o
|
||||
snd-hwdep-y := hwdep.o
|
||||
snd-seq-device-y := seq_device.o
|
||||
|
||||
snd-compress-objs := compress_offload.o
|
||||
snd-compress-y := compress_offload.o
|
||||
|
||||
obj-$(CONFIG_SND) += snd.o
|
||||
obj-$(CONFIG_SND_CTL_LED) += snd-ctl-led.o
|
||||
|
|
|
|||
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Reference in New Issue