How to model 3xADAU1761 codec in TDM8 with 2xSAI in ALSA ?
Hello,
I am working on bringup of a custom board that has three ADAU1761 codecs, and a NXP LS1028A SoC. The codecs are supposed to be configured in TDM8 mode and share a common playback I2S bus connected to SAI4, and a common capture I2S bus connected to SAI3. What is the recommended way of modelling this setup in the Linux device tree ?
So far I have tried the following:
&i2c0 { status = "okay";
i2c_codec1: adau1761@38 { compatible = "adi,adau1761"; #sound-dai-cells=<0>; reg = <0x38>; tdm-offset = <0>; clocks = <&adau1761_mclk>; clock-names = "mclk"; };
i2c_codec2: adau1761@39 { compatible = "adi,adau1761"; #sound-dai-cells=<0>; reg = <0x39>; tdm-offset = <16>; clocks = <&adau1761_mclk>; clock-names = "mclk"; };
i2c_codec3: adau1761@3a { compatible = "adi,adau1761"; #sound-dai-cells=<0>; reg = <0x3a>; tdm-offset = <32>; clocks = <&adau1761_mclk>; clock-names = "mclk"; }; };
At first I attempted creating three simple-audio-card's. That attempted failed since sharing the same cpu (SAI3 and SAI4) failed with no ALSA device as a result.
Then I attempted creating a single simple-audio-card with two dai-links (one for playback and one for capture), each containing three codec sub-nodes. That doesn't really work either. Can someone please point me in the right direction? Many of the parameters that I have put in the dts have no effect and are assumed not supported.
Regards Rolf
sound { status = "okay"; model = "Super Sound System"; compatible = "simple-audio-card"; simple-audio-card,name = "sound"; simple-audio-card,format = "i2s"; simple-audio-card,bitclock-master = <&p_snd_codec_o1>; simple-audio-card,frame-master = <&p_snd_codec_o1>; simple-audio-card,widgets = "Microphone", "Mic In", "Headphone", "Headphone Out", "Line", "Line In", "Line", "Line Out";
simple-audio-card,routing = "Line Out", "LOUT", "Line Out", "ROUT", "Headphone Out", "LHP", "Headphone Out", "RHP", "Mic In", "MICBIAS", "LINN", "Mic In", "RINN", "Mic In", "LINP", "Mic In", "RINP", "Mic In", "LAUX", "Line In", "RAUX", "Line In";
playback_link: simple-audio-card,dai-link@0 { format = "i2s"; convert-channels = <8>; bitclock-master = <&p_snd_codec_o1>; frame-master = <&p_snd_codec_o1>; name-prefix = "p";
snd_dai_playback: cpu@0 {
cpu-dai-name = "sai4_0"; sound-dai-name = "sai4_0";
sound-dai = <&sai4 0>; dai-tdm-slot-num = <8>; dai-tdm-slot-width = <16>; dai-tdm-slot-tx-mask = <1 0 1 0 1 0 0 0>; dai-tdm-slot-rx-mask = <0 0 0 0 0 0 0 0>; };
p_snd_codec_o1: codec@0 { sound-dai = <&i2c_codec1>; frame-master; bitclock-master; clocks=<&adau1761_mclk>; clock-frequency = <12288000>; clock-names = "mclk"; name-prefix = "c1"; dai-tdm-slot-tx-mask = <1 0 0 0 0 0 0 0>; dai-tdm-slot-rx-mask = <0 0 0 0 0 0 0 0>; };
p_snd_codec_o2: codec@1 { sound-dai = <&i2c_codec2>; clocks=<&adau1761_mclk>; clock-frequency = <12288000>; clock-names = "mclk"; name-prefix = "c2"; dai-tdm-slot-tx-mask = <0 0 1 0 0 0 0 0>; dai-tdm-slot-rx-mask = <0 0 0 0 0 0 0 0>; }; p_snd_codec_o3: codec@2 { sound-dai = <&i2c_codec3>; clocks=<&adau1761_mclk>; clock-frequency = <12288000>; clock-names = "mclk"; name-prefix = "c3"; dai-tdm-slot-tx-mask = <0 0 0 0 1 0 0 0>; dai-tdm-slot-rx-mask = <0 0 0 0 0 0 0 0>; };
};
capture_link: simple-audio-card,dai-link@1 { format = "i2s"; convert-channels = <8>; bitclock-master = <&c_snd_codec_o1>; frame-master = <&c_snd_codec_o1>; name-prefix = "c";
snd_dai_capture: cpu@0 { sound-dai = <&sai3>; dai-tdm-slot-num = <8>; dai-tdm-slot-width = <16>; dai-tdm-slot-tx-mask = <0 0 0 0 0 0 0 0>; dai-tdm-slot-rx-mask = <0 1 0 1 0 1 0 0>; };
c_snd_codec_o1: codec@0 { sound-dai = <&i2c_codec1>; frame-master; bitclock-master; clocks=<&adau1761_mclk>; clock-names = "mclk"; clock-frequency = <12288000>; name-prefix = "c1"; dai-tdm-slot-tx-mask = <0 0 0 0 0 0 0 0>; dai-tdm-slot-rx-mask = <0 1 0 0 0 0 0 0>; };
c_snd_codec_o2: codec@1 { sound-dai = <&i2c_codec2>; clocks=<&adau1761_mclk>; clock-names = "mclk"; clock-frequency = <12288000>; name-prefix = "c2"; dai-tdm-slot-tx-mask = <0 0 0 0 0 0 0 0>; dai-tdm-slot-rx-mask = <0 0 0 1 0 0 0 0>; }; c_snd_codec_o3: codec@2 { sound-dai = <&i2c_codec3>; clocks=<&adau1761_mclk>; clock-names = "mclk"; clock-frequency = <12288000>; name-prefix = "c3"; dai-tdm-slot-tx-mask = <0 0 0 0 0 0 0 0>; dai-tdm-slot-rx-mask = <0 0 0 0 0 1 0 0>; };
};
On 3/9/20 9:33 AM, Rolf Peder Klemetsen wrote:
Hello,
I am working on bringup of a custom board that has three ADAU1761 codecs, and a NXP LS1028A SoC. The codecs are supposed to be configured in TDM8 mode and share a common playback I2S bus connected to SAI4, and a common capture I2S bus connected to SAI3. What is the recommended way of modelling this setup in the Linux device tree ?
So far I have tried the following:
&i2c0 { status = "okay";
i2c_codec1: adau1761@38 { compatible = "adi,adau1761"; #sound-dai-cells=<0>; reg = <0x38>; tdm-offset = <0>; clocks = <&adau1761_mclk>; clock-names = "mclk"; }; i2c_codec2: adau1761@39 { compatible = "adi,adau1761"; #sound-dai-cells=<0>; reg = <0x39>; tdm-offset = <16>; clocks = <&adau1761_mclk>; clock-names = "mclk"; }; i2c_codec3: adau1761@3a { compatible = "adi,adau1761"; #sound-dai-cells=<0>; reg = <0x3a>; tdm-offset = <32>; clocks = <&adau1761_mclk>; clock-names = "mclk"; };};
At first I attempted creating three simple-audio-card's. That attempted failed since sharing the same cpu (SAI3 and SAI4) failed with no ALSA device as a result.
Then I attempted creating a single simple-audio-card with two dai-links (one for playback and one for capture), each containing three codec sub-nodes. That doesn't really work either. Can someone please point me in the right direction? Many of the parameters that I have put in the dts have no effect and are assumed not supported.
Hi,
One physical sound card is definitely the right direction. If you need multiple sound cards exposed to userspace you can use the ALSA plugin architecture to create multiple virtual sound cards on top of the one sound card.
Do you know why your approach with a single sound card does not work, do you get any error messages? Is the sound card registered?
The primary use case of the simple-audio-card is as the name suggests simple setups and your setup is not that simple. It is possible that your particular setup has never been tested with the simple-sound-card and there are still some hidden bugs. It might also be the case that you might have to create a custom sound card driver for your setup if you can't make it work with the simple card driver.
I believe I can see at least on issue with your DT snipped. For the widgets that are registered by the CODECs the prefix will be added to the name. So for example "LOUT" has to be "c1 LOUT", in addition you'll have to add one route for each sound card.
- Lars
Regards Rolf
sound { status = "okay"; model = "Super Sound System"; compatible = "simple-audio-card"; simple-audio-card,name = "sound"; simple-audio-card,format = "i2s"; simple-audio-card,bitclock-master = <&p_snd_codec_o1>; simple-audio-card,frame-master = <&p_snd_codec_o1>; simple-audio-card,widgets = "Microphone", "Mic In", "Headphone", "Headphone Out", "Line", "Line In", "Line", "Line Out";
simple-audio-card,routing = "Line Out", "LOUT", "Line Out", "ROUT", "Headphone Out", "LHP", "Headphone Out", "RHP", "Mic In", "MICBIAS", "LINN", "Mic In", "RINN", "Mic In", "LINP", "Mic In", "RINP", "Mic In", "LAUX", "Line In", "RAUX", "Line In"; playback_link: simple-audio-card,dai-link@0 { format = "i2s"; convert-channels = <8>; bitclock-master = <&p_snd_codec_o1>; frame-master = <&p_snd_codec_o1>; name-prefix = "p"; snd_dai_playback: cpu@0 { cpu-dai-name = "sai4_0"; sound-dai-name = "sai4_0"; sound-dai = <&sai4 0>; dai-tdm-slot-num = <8>; dai-tdm-slot-width = <16>; dai-tdm-slot-tx-mask = <1 0 1 0 1 0 0 0>; dai-tdm-slot-rx-mask = <0 0 0 0 0 0 0 0>; }; p_snd_codec_o1: codec@0 { sound-dai = <&i2c_codec1>; frame-master; bitclock-master; clocks=<&adau1761_mclk>; clock-frequency = <12288000>; clock-names = "mclk"; name-prefix = "c1"; dai-tdm-slot-tx-mask = <1 0 0 0 0 0 0 0>; dai-tdm-slot-rx-mask = <0 0 0 0 0 0 0 0>; }; p_snd_codec_o2: codec@1 { sound-dai = <&i2c_codec2>; clocks=<&adau1761_mclk>; clock-frequency = <12288000>; clock-names = "mclk"; name-prefix = "c2"; dai-tdm-slot-tx-mask = <0 0 1 0 0 0 0 0>; dai-tdm-slot-rx-mask = <0 0 0 0 0 0 0 0>; }; p_snd_codec_o3: codec@2 { sound-dai = <&i2c_codec3>; clocks=<&adau1761_mclk>; clock-frequency = <12288000>; clock-names = "mclk"; name-prefix = "c3"; dai-tdm-slot-tx-mask = <0 0 0 0 1 0 0 0>; dai-tdm-slot-rx-mask = <0 0 0 0 0 0 0 0>; }; }; capture_link: simple-audio-card,dai-link@1 { format = "i2s"; convert-channels = <8>; bitclock-master = <&c_snd_codec_o1>; frame-master = <&c_snd_codec_o1>; name-prefix = "c"; snd_dai_capture: cpu@0 { sound-dai = <&sai3>; dai-tdm-slot-num = <8>; dai-tdm-slot-width = <16>; dai-tdm-slot-tx-mask = <0 0 0 0 0 0 0 0>; dai-tdm-slot-rx-mask = <0 1 0 1 0 1 0 0>; }; c_snd_codec_o1: codec@0 { sound-dai = <&i2c_codec1>; frame-master; bitclock-master; clocks=<&adau1761_mclk>; clock-names = "mclk"; clock-frequency = <12288000>; name-prefix = "c1"; dai-tdm-slot-tx-mask = <0 0 0 0 0 0 0 0>; dai-tdm-slot-rx-mask = <0 1 0 0 0 0 0 0>; }; c_snd_codec_o2: codec@1 { sound-dai = <&i2c_codec2>; clocks=<&adau1761_mclk>; clock-names = "mclk"; clock-frequency = <12288000>; name-prefix = "c2"; dai-tdm-slot-tx-mask = <0 0 0 0 0 0 0 0>; dai-tdm-slot-rx-mask = <0 0 0 1 0 0 0 0>; }; c_snd_codec_o3: codec@2 { sound-dai = <&i2c_codec3>; clocks=<&adau1761_mclk>; clock-names = "mclk"; clock-frequency = <12288000>; name-prefix = "c3"; dai-tdm-slot-tx-mask = <0 0 0 0 0 0 0 0>; dai-tdm-slot-rx-mask = <0 0 0 0 0 1 0 0>; }; };
Hi,
Thanks Lars. I finally got prefix working by adding "sound-name-prefix" to the adau1761 definitions under &i2c0. And as you stated the controls are now name "c1 LOUT", etc. So that is good! But I still haven't got multi-codec to work; only the first codec in each dailink is enumerated. The others are ignored.
To me it seems that Asoc simple card supports multi-codec. See for example here: https://www.alsa-project.org/pipermail/alsa-devel/2018-March/133104.html and "Example 2 - many DAI links and multi-CODECs". I am unable to spot what is wrong with my dailink definition compared to the above example.
/Rolf
tir. 10. mar. 2020 kl. 14:28 skrev Lars-Peter Clausen lars@metafoo.de:
On 3/9/20 9:33 AM, Rolf Peder Klemetsen wrote:
Hello,
I am working on bringup of a custom board that has three ADAU1761 codecs, and a NXP LS1028A SoC. The codecs are supposed to be configured in TDM8 mode and share a common playback I2S bus connected to SAI4, and a common capture I2S bus connected to SAI3. What is the recommended way of modelling this setup in the Linux device tree ?
So far I have tried the following:
&i2c0 { status = "okay";
i2c_codec1: adau1761@38 { compatible = "adi,adau1761"; #sound-dai-cells=<0>; reg = <0x38>; tdm-offset = <0>; clocks = <&adau1761_mclk>; clock-names = "mclk"; }; i2c_codec2: adau1761@39 { compatible = "adi,adau1761"; #sound-dai-cells=<0>; reg = <0x39>; tdm-offset = <16>; clocks = <&adau1761_mclk>; clock-names = "mclk"; }; i2c_codec3: adau1761@3a { compatible = "adi,adau1761"; #sound-dai-cells=<0>; reg = <0x3a>; tdm-offset = <32>; clocks = <&adau1761_mclk>; clock-names = "mclk"; };};
At first I attempted creating three simple-audio-card's. That attempted failed since sharing the same cpu (SAI3 and SAI4) failed with no ALSA device as a result.
Then I attempted creating a single simple-audio-card with two dai-links (one for playback and one for capture), each containing three codec sub-nodes. That doesn't really work either. Can someone please point me in the right direction? Many of the parameters that I have put in the dts have no effect and are assumed not supported.
Hi,
One physical sound card is definitely the right direction. If you need multiple sound cards exposed to userspace you can use the ALSA plugin architecture to create multiple virtual sound cards on top of the one sound card.
Do you know why your approach with a single sound card does not work, do you get any error messages? Is the sound card registered?
The primary use case of the simple-audio-card is as the name suggests simple setups and your setup is not that simple. It is possible that your particular setup has never been tested with the simple-sound-card and there are still some hidden bugs. It might also be the case that you might have to create a custom sound card driver for your setup if you can't make it work with the simple card driver.
I believe I can see at least on issue with your DT snipped. For the widgets that are registered by the CODECs the prefix will be added to the name. So for example "LOUT" has to be "c1 LOUT", in addition you'll have to add one route for each sound card.
- Lars
Regards Rolf
sound { status = "okay"; model = "Super Sound System"; compatible = "simple-audio-card"; simple-audio-card,name = "sound"; simple-audio-card,format = "i2s"; simple-audio-card,bitclock-master = <&p_snd_codec_o1>; simple-audio-card,frame-master = <&p_snd_codec_o1>; simple-audio-card,widgets = "Microphone", "Mic In", "Headphone", "Headphone Out", "Line", "Line In", "Line", "Line Out";
simple-audio-card,routing = "Line Out", "LOUT", "Line Out", "ROUT", "Headphone Out", "LHP", "Headphone Out", "RHP", "Mic In", "MICBIAS", "LINN", "Mic In", "RINN", "Mic In", "LINP", "Mic In", "RINP", "Mic In", "LAUX", "Line In", "RAUX", "Line In"; playback_link: simple-audio-card,dai-link@0 { format = "i2s"; convert-channels = <8>; bitclock-master = <&p_snd_codec_o1>; frame-master = <&p_snd_codec_o1>; name-prefix = "p"; snd_dai_playback: cpu@0 { cpu-dai-name = "sai4_0"; sound-dai-name = "sai4_0"; sound-dai = <&sai4 0>; dai-tdm-slot-num = <8>; dai-tdm-slot-width = <16>; dai-tdm-slot-tx-mask = <1 0 1 0 1 0 0 0>; dai-tdm-slot-rx-mask = <0 0 0 0 0 0 0 0>; }; p_snd_codec_o1: codec@0 { sound-dai = <&i2c_codec1>; frame-master; bitclock-master; clocks=<&adau1761_mclk>; clock-frequency = <12288000>; clock-names = "mclk"; name-prefix = "c1"; dai-tdm-slot-tx-mask = <1 0 0 0 0 0 0 0>; dai-tdm-slot-rx-mask = <0 0 0 0 0 0 0 0>; }; p_snd_codec_o2: codec@1 { sound-dai = <&i2c_codec2>; clocks=<&adau1761_mclk>; clock-frequency = <12288000>; clock-names = "mclk"; name-prefix = "c2"; dai-tdm-slot-tx-mask = <0 0 1 0 0 0 0 0>; dai-tdm-slot-rx-mask = <0 0 0 0 0 0 0 0>; }; p_snd_codec_o3: codec@2 { sound-dai = <&i2c_codec3>; clocks=<&adau1761_mclk>; clock-frequency = <12288000>; clock-names = "mclk"; name-prefix = "c3"; dai-tdm-slot-tx-mask = <0 0 0 0 1 0 0 0>; dai-tdm-slot-rx-mask = <0 0 0 0 0 0 0 0>; }; }; capture_link: simple-audio-card,dai-link@1 { format = "i2s"; convert-channels = <8>; bitclock-master = <&c_snd_codec_o1>; frame-master = <&c_snd_codec_o1>; name-prefix = "c"; snd_dai_capture: cpu@0 { sound-dai = <&sai3>; dai-tdm-slot-num = <8>; dai-tdm-slot-width = <16>; dai-tdm-slot-tx-mask = <0 0 0 0 0 0 0 0>; dai-tdm-slot-rx-mask = <0 1 0 1 0 1 0 0>; }; c_snd_codec_o1: codec@0 { sound-dai = <&i2c_codec1>; frame-master; bitclock-master; clocks=<&adau1761_mclk>; clock-names = "mclk"; clock-frequency = <12288000>; name-prefix = "c1"; dai-tdm-slot-tx-mask = <0 0 0 0 0 0 0 0>; dai-tdm-slot-rx-mask = <0 1 0 0 0 0 0 0>; }; c_snd_codec_o2: codec@1 { sound-dai = <&i2c_codec2>; clocks=<&adau1761_mclk>; clock-names = "mclk"; clock-frequency = <12288000>; name-prefix = "c2"; dai-tdm-slot-tx-mask = <0 0 0 0 0 0 0 0>; dai-tdm-slot-rx-mask = <0 0 0 1 0 0 0 0>; }; c_snd_codec_o3: codec@2 { sound-dai = <&i2c_codec3>; clocks=<&adau1761_mclk>; clock-names = "mclk"; clock-frequency = <12288000>; name-prefix = "c3"; dai-tdm-slot-tx-mask = <0 0 0 0 0 0 0 0>; dai-tdm-slot-rx-mask = <0 0 0 0 0 1 0 0>; }; };
On 3/10/20 8:21 PM, Rolf Peder Klemetsen wrote:
Hi,
Thanks Lars. I finally got prefix working by adding "sound-name-prefix" to the adau1761 definitions under &i2c0. And as you stated the controls are now name "c1 LOUT", etc. So that is good! But I still haven't got multi-codec to work; only the first codec in each dailink is enumerated. The others are ignored.
To me it seems that Asoc simple card supports multi-codec. See for example here: https://www.alsa-project.org/pipermail/alsa-devel/2018-March/133104.html and "Example 2 - many DAI links and multi-CODECs". I am unable to spot what is wrong with my dailink definition compared to the above example.
Hi,
I just had a quick look at the code and it seems that at the moment the simple-card only supports a single CODEC.
https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/tree/soun...
Your options are to extend it to support multiple CODECs or create a custom card driver that handles your case.
On a first look it seems it wouldn't be that difficult to extend it, but there might be some hidden pitfalls.
- Lars
tir. 10. mar. 2020 kl. 14:28 skrev Lars-Peter Clausen lars@metafoo.de:
On 3/9/20 9:33 AM, Rolf Peder Klemetsen wrote:
Hello,
I am working on bringup of a custom board that has three ADAU1761 codecs, and a NXP LS1028A SoC. The codecs are supposed to be configured in TDM8 mode and share a common playback I2S bus connected to SAI4, and a common capture I2S bus connected to SAI3. What is the recommended way of modelling this setup in the Linux device tree ?
So far I have tried the following:
&i2c0 { status = "okay";
i2c_codec1: adau1761@38 { compatible = "adi,adau1761"; #sound-dai-cells=<0>; reg = <0x38>; tdm-offset = <0>; clocks = <&adau1761_mclk>; clock-names = "mclk"; }; i2c_codec2: adau1761@39 { compatible = "adi,adau1761"; #sound-dai-cells=<0>; reg = <0x39>; tdm-offset = <16>; clocks = <&adau1761_mclk>; clock-names = "mclk"; }; i2c_codec3: adau1761@3a { compatible = "adi,adau1761"; #sound-dai-cells=<0>; reg = <0x3a>; tdm-offset = <32>; clocks = <&adau1761_mclk>; clock-names = "mclk"; };};
At first I attempted creating three simple-audio-card's. That attempted failed since sharing the same cpu (SAI3 and SAI4) failed with no ALSA device as a result.
Then I attempted creating a single simple-audio-card with two dai-links (one for playback and one for capture), each containing three codec sub-nodes. That doesn't really work either. Can someone please point me in the right direction? Many of the parameters that I have put in the dts have no effect and are assumed not supported.
Hi,
One physical sound card is definitely the right direction. If you need multiple sound cards exposed to userspace you can use the ALSA plugin architecture to create multiple virtual sound cards on top of the one sound card.
Do you know why your approach with a single sound card does not work, do you get any error messages? Is the sound card registered?
The primary use case of the simple-audio-card is as the name suggests simple setups and your setup is not that simple. It is possible that your particular setup has never been tested with the simple-sound-card and there are still some hidden bugs. It might also be the case that you might have to create a custom sound card driver for your setup if you can't make it work with the simple card driver.
I believe I can see at least on issue with your DT snipped. For the widgets that are registered by the CODECs the prefix will be added to the name. So for example "LOUT" has to be "c1 LOUT", in addition you'll have to add one route for each sound card.
- Lars
Regards Rolf
sound { status = "okay"; model = "Super Sound System"; compatible = "simple-audio-card"; simple-audio-card,name = "sound"; simple-audio-card,format = "i2s"; simple-audio-card,bitclock-master = <&p_snd_codec_o1>; simple-audio-card,frame-master = <&p_snd_codec_o1>; simple-audio-card,widgets = "Microphone", "Mic In", "Headphone", "Headphone Out", "Line", "Line In", "Line", "Line Out";
simple-audio-card,routing = "Line Out", "LOUT", "Line Out", "ROUT", "Headphone Out", "LHP", "Headphone Out", "RHP", "Mic In", "MICBIAS", "LINN", "Mic In", "RINN", "Mic In", "LINP", "Mic In", "RINP", "Mic In", "LAUX", "Line In", "RAUX", "Line In"; playback_link: simple-audio-card,dai-link@0 { format = "i2s"; convert-channels = <8>; bitclock-master = <&p_snd_codec_o1>; frame-master = <&p_snd_codec_o1>; name-prefix = "p"; snd_dai_playback: cpu@0 { cpu-dai-name = "sai4_0"; sound-dai-name = "sai4_0"; sound-dai = <&sai4 0>; dai-tdm-slot-num = <8>; dai-tdm-slot-width = <16>; dai-tdm-slot-tx-mask = <1 0 1 0 1 0 0 0>; dai-tdm-slot-rx-mask = <0 0 0 0 0 0 0 0>; }; p_snd_codec_o1: codec@0 { sound-dai = <&i2c_codec1>; frame-master; bitclock-master; clocks=<&adau1761_mclk>; clock-frequency = <12288000>; clock-names = "mclk"; name-prefix = "c1"; dai-tdm-slot-tx-mask = <1 0 0 0 0 0 0 0>; dai-tdm-slot-rx-mask = <0 0 0 0 0 0 0 0>; }; p_snd_codec_o2: codec@1 { sound-dai = <&i2c_codec2>; clocks=<&adau1761_mclk>; clock-frequency = <12288000>; clock-names = "mclk"; name-prefix = "c2"; dai-tdm-slot-tx-mask = <0 0 1 0 0 0 0 0>; dai-tdm-slot-rx-mask = <0 0 0 0 0 0 0 0>; }; p_snd_codec_o3: codec@2 { sound-dai = <&i2c_codec3>; clocks=<&adau1761_mclk>; clock-frequency = <12288000>; clock-names = "mclk"; name-prefix = "c3"; dai-tdm-slot-tx-mask = <0 0 0 0 1 0 0 0>; dai-tdm-slot-rx-mask = <0 0 0 0 0 0 0 0>; }; }; capture_link: simple-audio-card,dai-link@1 { format = "i2s"; convert-channels = <8>; bitclock-master = <&c_snd_codec_o1>; frame-master = <&c_snd_codec_o1>; name-prefix = "c"; snd_dai_capture: cpu@0 { sound-dai = <&sai3>; dai-tdm-slot-num = <8>; dai-tdm-slot-width = <16>; dai-tdm-slot-tx-mask = <0 0 0 0 0 0 0 0>; dai-tdm-slot-rx-mask = <0 1 0 1 0 1 0 0>; }; c_snd_codec_o1: codec@0 { sound-dai = <&i2c_codec1>; frame-master; bitclock-master; clocks=<&adau1761_mclk>; clock-names = "mclk"; clock-frequency = <12288000>; name-prefix = "c1"; dai-tdm-slot-tx-mask = <0 0 0 0 0 0 0 0>; dai-tdm-slot-rx-mask = <0 1 0 0 0 0 0 0>; }; c_snd_codec_o2: codec@1 { sound-dai = <&i2c_codec2>; clocks=<&adau1761_mclk>; clock-names = "mclk"; clock-frequency = <12288000>; name-prefix = "c2"; dai-tdm-slot-tx-mask = <0 0 0 0 0 0 0 0>; dai-tdm-slot-rx-mask = <0 0 0 1 0 0 0 0>; }; c_snd_codec_o3: codec@2 { sound-dai = <&i2c_codec3>; clocks=<&adau1761_mclk>; clock-names = "mclk"; clock-frequency = <12288000>; name-prefix = "c3"; dai-tdm-slot-tx-mask = <0 0 0 0 0 0 0 0>; dai-tdm-slot-rx-mask = <0 0 0 0 0 1 0 0>; }; };
participants (2)
-
Lars-Peter Clausen -
Rolf Peder Klemetsen