[alsa-devel] Multiple codecs on one sound card for multi-channel sound card
Hi all, What's the current best practices to specify multiple codecs on one sound card, with all codecs sharing a single TDM bus?
I have a card with up to 32 TLV320AIC33 codecs on it. For the moment, we can limit the discussion to only 16 codecs, so we don't have to get extra serializers involved...
I currently have something like this in my am335x-boneblack.dts file:
(i.e. attempting to set tdm slot width and num in the i2c codec)
but it doesn't seem to be pickin up the fact that I want all the codecs linked together as one.
(FYI, the first codec, i.e. tlv32aic3x_a should be the master, and all others including the CPU will be slaves).
Thanks for any help!
BTW, I'm happy to use whatever kernel will support me with the minimal amount of effort!
Sincerely, -Caleb
&i2c1 { clock-frequency = <100000>; status = "okay"; pinctrl-names = "default"; pinctrl-0 = <&i2c1_pins_default>; status="okay";
tlv320aic3x_a: tlv320aic3x@18 { #sound-dai-cells = <0>; compatible = "ti,tlv320aic3x"; reg = <0x18>; status = "okay"; dai-tdm-slot-width = <16>; dai-tdm-slot-num = <0>; }; tlv320aic3x_b: tlv320aic3x@19 { #sound-dai-cells = <0>; compatible = "ti,tlv320aic3x"; reg = <0x19>; status = "okay"; dai-tdm-slot-width = <16>; dai-tdm-slot-num = <2>; }; tlv320aic3x_c: tlv320aic3x@1a { compatible = "ti,tlv320aic3x"; reg = <0x1a>; status = "okay"; dai-tdm-slot-width = <16>; dai-tdm-slot-num = <4>; }; tlv320aic3x_d: tlv320aic3x@1b { compatible = "ti,tlv320aic3x"; reg = <0x1b>; status = "okay"; dai-tdm-slot-width = <16>; dai-tdm-slot-num = <6>; }; };
sound { compatible = "simple-audio-card"; simple-audio-card,name = "puck audio"; // simple-audio-card,widgets= // simple-audio-card,routing= simple-audio-card,mclk-fs = <256>; status="okay"; simple-audio-card,bitclock-master = <&tlv320aic3x_a>; simple-audio-card,frame-master = <&tlv320aic3x_a>; simple-audio-card,dai-link@0 { format = "left_j"; cpu { sound-dai = <&mcasp0 0>; };
// *** Here's where I'm really confused -- this part doesn't seem to be supported to // have all the codecs on a single DAI. // The examples given seem to be for different codec DAIs // sharing a single CPU DAI, but not used at the same time.
codec { #sound-dai-cells = <0>; sound-dai = < &tlv320aic3x_a &tlv320aic3x_b &tlv320aic3x_c &tlv320aic3x_d >; }; }; };
Hi Mark Brown, Looking back on my email notes, you helped me with this back in 2011 with an old kernel version. I wonder if mutlti-codec-on-single-DAI configuration is still unsupported by the ASOC core. I don't see anything obvious.
If I were to implement it 'properly', what's the right DTS format to go with? Something like what I suggest below (i.e. in the simple-audio-card section, use multiple codecs in a single sound-dai? If so, where should the tdm-slot & clocking information go? In the codec DTS section itself, or in the sound-dai section?
Thanks again, -Caleb
On Tue, Sep 8, 2015 at 6:02 PM, Caleb Crome caleb@crome.org wrote:
Hi all, What's the current best practices to specify multiple codecs on one sound card, with all codecs sharing a single TDM bus?
I have a card with up to 32 TLV320AIC33 codecs on it. For the moment, we can limit the discussion to only 16 codecs, so we don't have to get extra serializers involved...
I currently have something like this in my am335x-boneblack.dts file:
(i.e. attempting to set tdm slot width and num in the i2c codec)
but it doesn't seem to be pickin up the fact that I want all the codecs linked together as one.
(FYI, the first codec, i.e. tlv32aic3x_a should be the master, and all others including the CPU will be slaves).
Thanks for any help!
BTW, I'm happy to use whatever kernel will support me with the minimal amount of effort!
Sincerely, -Caleb
&i2c1 { clock-frequency = <100000>; status = "okay"; pinctrl-names = "default"; pinctrl-0 = <&i2c1_pins_default>; status="okay";
tlv320aic3x_a: tlv320aic3x@18 { #sound-dai-cells = <0>; compatible = "ti,tlv320aic3x"; reg = <0x18>; status = "okay"; dai-tdm-slot-width = <16>; dai-tdm-slot-num = <0>; }; tlv320aic3x_b: tlv320aic3x@19 { #sound-dai-cells = <0>; compatible = "ti,tlv320aic3x"; reg = <0x19>; status = "okay"; dai-tdm-slot-width = <16>; dai-tdm-slot-num = <2>; }; tlv320aic3x_c: tlv320aic3x@1a { compatible = "ti,tlv320aic3x"; reg = <0x1a>; status = "okay"; dai-tdm-slot-width = <16>; dai-tdm-slot-num = <4>; }; tlv320aic3x_d: tlv320aic3x@1b { compatible = "ti,tlv320aic3x"; reg = <0x1b>; status = "okay"; dai-tdm-slot-width = <16>; dai-tdm-slot-num = <6>; }; };
sound { compatible = "simple-audio-card"; simple-audio-card,name = "puck audio"; // simple-audio-card,widgets= // simple-audio-card,routing= simple-audio-card,mclk-fs = <256>; status="okay"; simple-audio-card,bitclock-master = <&tlv320aic3x_a>; simple-audio-card,frame-master = <&tlv320aic3x_a>; simple-audio-card,dai-link@0 { format = "left_j"; cpu { sound-dai = <&mcasp0 0>; };
// *** Here's where I'm really confused -- this part doesn't seem to be supported to // have all the codecs on a single DAI. // The examples given seem to be for different codec DAIs // sharing a single CPU DAI, but not used at the same time.
codec { #sound-dai-cells = <0>; sound-dai = < &tlv320aic3x_a &tlv320aic3x_b &tlv320aic3x_c &tlv320aic3x_d >; }; }; };
(re-sending hope it's not a duplicate -- I think I must have had HTML in my previous email and it was ignored)
Hi all, What are the current best practices to specify multiple codecs on one sound card, with all codecs sharing a single TDM bus?
I have a card with up to 32 TLV320AIC33 codecs on it. For the moment, we can limit the discussion to only 16 codecs, so we don't have to get extra serializers involved...
I currently have something like this in my am335x-boneblack.dts file:
(i.e. attempting to set tdm slot width and num in the i2c codec)
but it doesn't seem to be picking up the fact that I want all the codecs linked together as one.
(FYI, the first codec, i.e. tlv32aic3x_a should be the master, and all others including the CPU will be slaves).
Thanks for any help!
BTW, I'm happy to use whatever kernel will support me with the minimal amount of effort, currently using 4.1
Thank you! -Caleb
&i2c1 { ... tlv320aic3x_a: tlv320aic3x@18 { #sound-dai-cells = <1>; compatible = "ti,tlv320aic3x"; reg = <0x18>; tdm-offset = <0>; status = "okay"; }; tlv320aic3x_b: tlv320aic3x@19 { #sound-dai-cells = <1>; compatible = "ti,tlv320aic3x"; reg = <0x19>; tdm-offset = <32>; status = "okay"; }; tlv320aic3x_c: tlv320aic3x@1a { #sound-dai-cells = <1>; compatible = "ti,tlv320aic3x"; reg = <0x1a>; tdm-offset = <64>; status = "okay"; }; tlv320aic3x_d: tlv320aic3x@1b { #sound-dai-cells = <1>; compatible = "ti,tlv320aic3x"; reg = <0x1b>; tdm-offset = <96>; status = "okay"; }; };
&mcasp0 { #sound-dai-cells = <0>; pinctrl-names = "default"; pinctrl-0 = <&mcasp_0_pins_default>; status = "okay";
op-mode = <0>; /* MCASP_IIS_MODE */ tdm-slots = <16>; num-serializer = <16>; serial-dir = < /* 0: INACTIVE, 1: TX, 2: RX */ 0 0 1 2 0 0 0 0 0 0 0 0 0 0 0 0
;
tx-num-evt = <1>; rx-num-evt = <1>; };
/ { sound { compatible = "simple-audio-card"; simple-audio-card,name = "puppy-audio"; simple-audio-card,mclk-fs = <256>; system-clock-frequency = <12288000>; status = "okay"; simple-audio-card,bitclock-master = <&tlv320aic3x_a>; simple-audio-card,frame-master = <&tlv320aic3x_a>; simple-audio-card,widgets = "Line", "Line Out", "Line", "Line In"; simple-audio-card,routing = "Line Out", "HPLOUT", "Line Out", "HPROUT", "Line In", "LINE1L", "Line In", "LINE1R";
simple-audio-card,dai-link@0 { format = "left_j"; cpu { sound-dai = <&mcasp0>; }; codec { sound-dai = <&tlv320aic3x_a 0>; dai-tdm-slot-num = <0>; dai-tdm-slot-width = <16>; }; }; /**** The stuff below doesn't work -- I can't figure out how to get the name_prefixes set on each codec... .How do I set up so that each codec gets its own name prfix and so that the soc core thinks all codecs are on the same DAI? **** */
// simple-audio-card,dai-link@1 { // format = "left_j"; // cpu { // sound-dai = <&mcasp0>; // }; // codec { // sound-dai = <&tlv320aic3x_b 0>; // dai-tdm-slot-num = <2>; // dai-tdm-slot-width = <16>; // }; // }; // simple-audio-card,dai-link@2 { // format = "left_j"; // cpu { // sound-dai = <&mcasp0>; // }; // codec { // sound-dai = <&tlv320aic3x_c>; // dai-tdm-slot-num = <4>; // dai-tdm-slot-width = <16>; // }; // }; // simple-audio-card,dai-link@3 { // format = "left_j"; // cpu { // sound-dai = <&mcasp0>; // }; // codec { // sound-dai = <&tlv320aic3x_d>; // dai-tdm-slot-num = <6>; // dai-tdm-slot-width = <16>; // }; // }; }; };
Hello Caleb,
Multi-codec support is now working fine the current linux releases. for previous releases, check if commit [PATCH] ASoC: dapm: Don't add prefix to widget stream name is included. I remember this is the last required patch.
After that, I don't know if the "simple-card" can be configured for multi codec support as you wish.
On my side, I cooked a particular sound card for the purpose. here is some tips:
struct snd_soc_dai_link_component codecs[2]; struct snd_soc_codec_conf codecs_conf[2]; [...] for (num_codecs=0; num_codecs<2; num_codecs++) { struct device_node *of_node; of_node = of_parse_phandle(pdev->dev.of_node, "audio-codec", num_codecs); if (!of_node) break; data->codecs[num_codecs].of_node = of_node; data->codecs[num_codecs].dai_name = "tlv320aic3x-foo";
/* add a "C2" name prefix for every control of the 2nd codec */ data->codecs_conf[num_codecs].of_node = of_node; if (num_codecs == 1) { data->codecs_conf[num_codecs].name_prefix = "C2"; } } [...] data->dai.codecs = data->codecs; data->dai.num_codecs = num_codecs; data->card.dai_link = &data->dai; data->card.codec_conf = data->codecs_conf; data->card.num_configs = num_codecs; [...]
On DTS side, I have something like
sound@0 { compatible = "fsl,imx-audio-foo"; model = "foo-audio"; ssi-controller = <&ssi1>;
/* * list phandles for the 2 codecs used in the same TDM network */ audio-codec = <&codec1>, <&codec2>;
[...]
};
You also need to have a particular hw_params() method to dispatch the TDM and sysclk configuration to your codecs, making the difference between your first codec (your bus master) and the others.
Arnaud
Le 15/09/2015 03:07, Caleb Crome a écrit :
(re-sending hope it's not a duplicate -- I think I must have had HTML in my previous email and it was ignored)
Hi all, What are the current best practices to specify multiple codecs on one sound card, with all codecs sharing a single TDM bus?
I have a card with up to 32 TLV320AIC33 codecs on it. For the moment, we can limit the discussion to only 16 codecs, so we don't have to get extra serializers involved...
I currently have something like this in my am335x-boneblack.dts file:
(i.e. attempting to set tdm slot width and num in the i2c codec)
but it doesn't seem to be picking up the fact that I want all the codecs linked together as one.
(FYI, the first codec, i.e. tlv32aic3x_a should be the master, and all others including the CPU will be slaves).
Thanks for any help!
BTW, I'm happy to use whatever kernel will support me with the minimal amount of effort, currently using 4.1
Thank you! -Caleb
&i2c1 { ... tlv320aic3x_a: tlv320aic3x@18 { #sound-dai-cells = <1>; compatible = "ti,tlv320aic3x"; reg = <0x18>; tdm-offset = <0>; status = "okay"; }; tlv320aic3x_b: tlv320aic3x@19 { #sound-dai-cells = <1>; compatible = "ti,tlv320aic3x"; reg = <0x19>; tdm-offset = <32>; status = "okay"; }; tlv320aic3x_c: tlv320aic3x@1a { #sound-dai-cells = <1>; compatible = "ti,tlv320aic3x"; reg = <0x1a>; tdm-offset = <64>; status = "okay"; }; tlv320aic3x_d: tlv320aic3x@1b { #sound-dai-cells = <1>; compatible = "ti,tlv320aic3x"; reg = <0x1b>; tdm-offset = <96>; status = "okay"; }; };
&mcasp0 { #sound-dai-cells = <0>; pinctrl-names = "default"; pinctrl-0 = <&mcasp_0_pins_default>; status = "okay";
op-mode = <0>; /* MCASP_IIS_MODE */ tdm-slots = <16>; num-serializer = <16>; serial-dir = < /* 0: INACTIVE, 1: TX, 2: RX */ 0 0 1 2 0 0 0 0 0 0 0 0 0 0 0 0
;
tx-num-evt = <1>; rx-num-evt = <1>; };
/ { sound { compatible = "simple-audio-card"; simple-audio-card,name = "puppy-audio"; simple-audio-card,mclk-fs = <256>; system-clock-frequency = <12288000>; status = "okay"; simple-audio-card,bitclock-master = <&tlv320aic3x_a>; simple-audio-card,frame-master = <&tlv320aic3x_a>; simple-audio-card,widgets = "Line", "Line Out", "Line", "Line In"; simple-audio-card,routing = "Line Out", "HPLOUT", "Line Out", "HPROUT", "Line In", "LINE1L", "Line In", "LINE1R";
simple-audio-card,dai-link@0 { format = "left_j"; cpu { sound-dai = <&mcasp0>; }; codec { sound-dai = <&tlv320aic3x_a 0>; dai-tdm-slot-num = <0>; dai-tdm-slot-width = <16>; }; }; /**** The stuff below doesn't work -- I can't figure out how to get the name_prefixes set on each codec... .How do I set up so that each codec gets its own name prfix and so that the soc core thinks all codecs are on the same DAI? **** */
// simple-audio-card,dai-link@1 { // format = "left_j"; // cpu { // sound-dai = <&mcasp0>; // }; // codec { // sound-dai = <&tlv320aic3x_b 0>; // dai-tdm-slot-num = <2>; // dai-tdm-slot-width = <16>; // }; // }; // simple-audio-card,dai-link@2 { // format = "left_j"; // cpu { // sound-dai = <&mcasp0>; // }; // codec { // sound-dai = <&tlv320aic3x_c>; // dai-tdm-slot-num = <4>; // dai-tdm-slot-width = <16>; // }; // }; // simple-audio-card,dai-link@3 { // format = "left_j"; // cpu { // sound-dai = <&mcasp0>; // }; // codec { // sound-dai = <&tlv320aic3x_d>; // dai-tdm-slot-num = <6>; // dai-tdm-slot-width = <16>; // }; // }; }; }; _______________________________________________ Alsa-devel mailing list Alsa-devel@alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
Ah Ha! thank you Arnaud! I'll understand this stuff eventually. I'll implement as you've shown and see how it goes.
I see that you have 'fsl,imx-audio-foo', using an SSI. from what I can see the SSIs only support channels_max=2. How do you get multi-channel TDM to work with the freescale SSI port? I need to get this TDM working on both TI & Freescale, and the freescale has been quite problematic because the multi-channel doesn't seem to be supported in the driver.
I have just checked out and verified that
http://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git/
sound/soc/fsl/*ssi*
all have channels_max = 2
We're very motivated to get this working on the freescale MX6, and getting it to work sure does not seem as simple as changing 2 to a bigger number :-)
Thanks, -Caleb
On Wed, Sep 16, 2015 at 2:57 AM, arnaud.mouiche@invoxia.com arnaud.mouiche@invoxia.com wrote:
Hello Caleb,
Multi-codec support is now working fine the current linux releases. for previous releases, check if commit [PATCH] ASoC: dapm: Don't add prefix to widget stream name is included. I remember this is the last required patch.
After that, I don't know if the "simple-card" can be configured for multi codec support as you wish.
On my side, I cooked a particular sound card for the purpose. here is some tips:
struct snd_soc_dai_link_component codecs[2]; struct snd_soc_codec_conf codecs_conf[2];
[...] for (num_codecs=0; num_codecs<2; num_codecs++) { struct device_node *of_node; of_node = of_parse_phandle(pdev->dev.of_node, "audio-codec", num_codecs); if (!of_node) break; data->codecs[num_codecs].of_node = of_node; data->codecs[num_codecs].dai_name = "tlv320aic3x-foo";
/* add a "C2" name prefix for every control of the 2nd codec */ data->codecs_conf[num_codecs].of_node = of_node; if (num_codecs == 1) { data->codecs_conf[num_codecs].name_prefix = "C2"; } }
[...] data->dai.codecs = data->codecs; data->dai.num_codecs = num_codecs; data->card.dai_link = &data->dai; data->card.codec_conf = data->codecs_conf; data->card.num_configs = num_codecs; [...]
On DTS side, I have something like
sound@0 { compatible = "fsl,imx-audio-foo"; model = "foo-audio"; ssi-controller = <&ssi1>; /* * list phandles for the 2 codecs used in the same TDM network */ audio-codec = <&codec1>, <&codec2>; [...] };
You also need to have a particular hw_params() method to dispatch the TDM and sysclk configuration to your codecs, making the difference between your first codec (your bus master) and the others.
Arnaud
Le 15/09/2015 03:07, Caleb Crome a écrit :
(re-sending hope it's not a duplicate -- I think I must have had HTML in my previous email and it was ignored)
Hi all, What are the current best practices to specify multiple codecs on one sound card, with all codecs sharing a single TDM bus?
I have a card with up to 32 TLV320AIC33 codecs on it. For the moment, we can limit the discussion to only 16 codecs, so we don't have to get extra serializers involved...
I currently have something like this in my am335x-boneblack.dts file:
(i.e. attempting to set tdm slot width and num in the i2c codec)
but it doesn't seem to be picking up the fact that I want all the codecs linked together as one.
(FYI, the first codec, i.e. tlv32aic3x_a should be the master, and all others including the CPU will be slaves).
Thanks for any help!
BTW, I'm happy to use whatever kernel will support me with the minimal amount of effort, currently using 4.1
Thank you! -Caleb
&i2c1 { ... tlv320aic3x_a: tlv320aic3x@18 { #sound-dai-cells = <1>; compatible = "ti,tlv320aic3x"; reg = <0x18>; tdm-offset = <0>; status = "okay"; }; tlv320aic3x_b: tlv320aic3x@19 { #sound-dai-cells = <1>; compatible = "ti,tlv320aic3x"; reg = <0x19>; tdm-offset = <32>; status = "okay"; }; tlv320aic3x_c: tlv320aic3x@1a { #sound-dai-cells = <1>; compatible = "ti,tlv320aic3x"; reg = <0x1a>; tdm-offset = <64>; status = "okay"; }; tlv320aic3x_d: tlv320aic3x@1b { #sound-dai-cells = <1>; compatible = "ti,tlv320aic3x"; reg = <0x1b>; tdm-offset = <96>; status = "okay"; }; };
&mcasp0 { #sound-dai-cells = <0>; pinctrl-names = "default"; pinctrl-0 = <&mcasp_0_pins_default>; status = "okay";
op-mode = <0>; /* MCASP_IIS_MODE */ tdm-slots = <16>; num-serializer = <16>; serial-dir = < /* 0: INACTIVE, 1: TX, 2: RX */ 0 0 1 2 0 0 0 0 0 0 0 0 0 0 0 0
;
tx-num-evt = <1>; rx-num-evt = <1>; };
/ { sound { compatible = "simple-audio-card"; simple-audio-card,name = "puppy-audio"; simple-audio-card,mclk-fs = <256>; system-clock-frequency = <12288000>; status = "okay"; simple-audio-card,bitclock-master = <&tlv320aic3x_a>; simple-audio-card,frame-master = <&tlv320aic3x_a>; simple-audio-card,widgets = "Line", "Line Out", "Line", "Line In"; simple-audio-card,routing = "Line Out", "HPLOUT", "Line Out", "HPROUT", "Line In", "LINE1L", "Line In", "LINE1R";
simple-audio-card,dai-link@0 { format = "left_j"; cpu { sound-dai = <&mcasp0>; }; codec { sound-dai = <&tlv320aic3x_a 0>; dai-tdm-slot-num = <0>; dai-tdm-slot-width = <16>; }; }; /**** The stuff below doesn't work -- I can't figure out how to get the name_prefixes set on each codec... .How do I set up so that each codec gets its own name prfix and so that the soc core thinks all codecs are on the same DAI? **** */
// simple-audio-card,dai-link@1 { // format = "left_j"; // cpu { // sound-dai = <&mcasp0>; // }; // codec { // sound-dai = <&tlv320aic3x_b 0>; // dai-tdm-slot-num = <2>; // dai-tdm-slot-width = <16>; // }; // }; // simple-audio-card,dai-link@2 { // format = "left_j"; // cpu { // sound-dai = <&mcasp0>; // }; // codec { // sound-dai = <&tlv320aic3x_c>; // dai-tdm-slot-num = <4>; // dai-tdm-slot-width = <16>; // }; // }; // simple-audio-card,dai-link@3 { // format = "left_j"; // cpu { // sound-dai = <&mcasp0>; // }; // codec { // sound-dai = <&tlv320aic3x_d>; // dai-tdm-slot-num = <6>; // dai-tdm-slot-width = <16>; // }; // }; }; }; _______________________________________________ Alsa-devel mailing list Alsa-devel@alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
Alsa-devel mailing list Alsa-devel@alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
Hello Caleb,
freescale SSI is fine with more than 2 channels. I have planed to publish a set of patch in this direction but we still have some corner cases to fix first (rare issues with channels alignment)
Yet, this following is far enough to have it working. And since the max/min rate is at the end the intersection of what the SSI, the codec and the card are declaring, it will not change anything until you connect the SSI to a codec with more than 2 channels, or a multi-codec solution.
Arnaud
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index c25a1e8..26e980b 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1082,14 +1082,14 @@ static struct snd_soc_dai_driver fsl_ssi_dai_template = { .playback = { .stream_name = "CPU-Playback", .channels_min = 1, - .channels_max = 2, + .channels_max = 4, .rates = FSLSSI_I2S_RATES, .formats = FSLSSI_I2S_FORMATS, }, .capture = { .stream_name = "CPU-Capture", .channels_min = 1, - .channels_max = 2, + .channels_max = 4, .rates = FSLSSI_I2S_RATES, .formats = FSLSSI_I2S_FORMATS, },
Le 17/09/2015 00:07, Caleb Crome a écrit :
Ah Ha! thank you Arnaud! I'll understand this stuff eventually. I'll implement as you've shown and see how it goes.
I see that you have 'fsl,imx-audio-foo', using an SSI. from what I can see the SSIs only support channels_max=2. How do you get multi-channel TDM to work with the freescale SSI port? I need to get this TDM working on both TI & Freescale, and the freescale has been quite problematic because the multi-channel doesn't seem to be supported in the driver.
I have just checked out and verified that
http://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git/
sound/soc/fsl/*ssi*
all have channels_max = 2
We're very motivated to get this working on the freescale MX6, and getting it to work sure does not seem as simple as changing 2 to a bigger number :-)
Thanks, -Caleb
On Wed, Sep 16, 2015 at 2:57 AM, arnaud.mouiche@invoxia.com arnaud.mouiche@invoxia.com wrote:
Hello Caleb,
Multi-codec support is now working fine the current linux releases. for previous releases, check if commit [PATCH] ASoC: dapm: Don't add prefix to widget stream name is included. I remember this is the last required patch.
After that, I don't know if the "simple-card" can be configured for multi codec support as you wish.
On my side, I cooked a particular sound card for the purpose. here is some tips:
struct snd_soc_dai_link_component codecs[2]; struct snd_soc_codec_conf codecs_conf[2];
[...] for (num_codecs=0; num_codecs<2; num_codecs++) { struct device_node *of_node; of_node = of_parse_phandle(pdev->dev.of_node, "audio-codec", num_codecs); if (!of_node) break; data->codecs[num_codecs].of_node = of_node; data->codecs[num_codecs].dai_name = "tlv320aic3x-foo";
/* add a "C2" name prefix for every control of the 2nd codec */ data->codecs_conf[num_codecs].of_node = of_node; if (num_codecs == 1) { data->codecs_conf[num_codecs].name_prefix = "C2"; } }
[...] data->dai.codecs = data->codecs; data->dai.num_codecs = num_codecs; data->card.dai_link = &data->dai; data->card.codec_conf = data->codecs_conf; data->card.num_configs = num_codecs; [...]
On DTS side, I have something like
sound@0 { compatible = "fsl,imx-audio-foo"; model = "foo-audio"; ssi-controller = <&ssi1>; /* * list phandles for the 2 codecs used in the same TDM network */ audio-codec = <&codec1>, <&codec2>; [...] };
You also need to have a particular hw_params() method to dispatch the TDM and sysclk configuration to your codecs, making the difference between your first codec (your bus master) and the others.
Arnaud
Le 15/09/2015 03:07, Caleb Crome a écrit :
(re-sending hope it's not a duplicate -- I think I must have had HTML in my previous email and it was ignored)
Hi all, What are the current best practices to specify multiple codecs on one sound card, with all codecs sharing a single TDM bus?
I have a card with up to 32 TLV320AIC33 codecs on it. For the moment, we can limit the discussion to only 16 codecs, so we don't have to get extra serializers involved...
I currently have something like this in my am335x-boneblack.dts file:
(i.e. attempting to set tdm slot width and num in the i2c codec)
but it doesn't seem to be picking up the fact that I want all the codecs linked together as one.
(FYI, the first codec, i.e. tlv32aic3x_a should be the master, and all others including the CPU will be slaves).
Thanks for any help!
BTW, I'm happy to use whatever kernel will support me with the minimal amount of effort, currently using 4.1
Thank you! -Caleb
&i2c1 { ... tlv320aic3x_a: tlv320aic3x@18 { #sound-dai-cells = <1>; compatible = "ti,tlv320aic3x"; reg = <0x18>; tdm-offset = <0>; status = "okay"; }; tlv320aic3x_b: tlv320aic3x@19 { #sound-dai-cells = <1>; compatible = "ti,tlv320aic3x"; reg = <0x19>; tdm-offset = <32>; status = "okay"; }; tlv320aic3x_c: tlv320aic3x@1a { #sound-dai-cells = <1>; compatible = "ti,tlv320aic3x"; reg = <0x1a>; tdm-offset = <64>; status = "okay"; }; tlv320aic3x_d: tlv320aic3x@1b { #sound-dai-cells = <1>; compatible = "ti,tlv320aic3x"; reg = <0x1b>; tdm-offset = <96>; status = "okay"; }; };
&mcasp0 { #sound-dai-cells = <0>; pinctrl-names = "default"; pinctrl-0 = <&mcasp_0_pins_default>; status = "okay";
op-mode = <0>; /* MCASP_IIS_MODE */ tdm-slots = <16>; num-serializer = <16>; serial-dir = < /* 0: INACTIVE, 1: TX, 2: RX */ 0 0 1 2 0 0 0 0 0 0 0 0 0 0 0 0
;
tx-num-evt = <1>; rx-num-evt = <1>; };
/ { sound { compatible = "simple-audio-card"; simple-audio-card,name = "puppy-audio"; simple-audio-card,mclk-fs = <256>; system-clock-frequency = <12288000>; status = "okay"; simple-audio-card,bitclock-master = <&tlv320aic3x_a>; simple-audio-card,frame-master = <&tlv320aic3x_a>; simple-audio-card,widgets = "Line", "Line Out", "Line", "Line In"; simple-audio-card,routing = "Line Out", "HPLOUT", "Line Out", "HPROUT", "Line In", "LINE1L", "Line In", "LINE1R";
simple-audio-card,dai-link@0 { format = "left_j"; cpu { sound-dai = <&mcasp0>; }; codec { sound-dai = <&tlv320aic3x_a 0>; dai-tdm-slot-num = <0>; dai-tdm-slot-width = <16>; }; }; /**** The stuff below doesn't work -- I can't figure out how to get the name_prefixes set on each codec... .How do I set up so that each codec gets its own name prfix and so that the soc core thinks all codecs are on the same DAI? **** */
// simple-audio-card,dai-link@1 { // format = "left_j"; // cpu { // sound-dai = <&mcasp0>; // }; // codec { // sound-dai = <&tlv320aic3x_b 0>; // dai-tdm-slot-num = <2>; // dai-tdm-slot-width = <16>; // }; // }; // simple-audio-card,dai-link@2 { // format = "left_j"; // cpu { // sound-dai = <&mcasp0>; // }; // codec { // sound-dai = <&tlv320aic3x_c>; // dai-tdm-slot-num = <4>; // dai-tdm-slot-width = <16>; // }; // }; // simple-audio-card,dai-link@3 { // format = "left_j"; // cpu { // sound-dai = <&mcasp0>; // }; // codec { // sound-dai = <&tlv320aic3x_d>; // dai-tdm-slot-num = <6>; // dai-tdm-slot-width = <16>; // }; // }; }; }; _______________________________________________ Alsa-devel mailing list Alsa-devel@alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
Alsa-devel mailing list Alsa-devel@alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
On Thu, Sep 17, 2015 at 1:51 AM, arnaud.mouiche@invoxia.com arnaud.mouiche@invoxia.com wrote:
Hello Caleb,
freescale SSI is fine with more than 2 channels. I have planed to publish a set of patch in this direction but we still have some corner cases to fix first (rare issues with channels alignment)
I'd love to see anything you're willing to show :-)
Yet, this following is far enough to have it working. And since the max/min rate is at the end the intersection of what the SSI, the codec and the card are declaring, it will not change anything until you connect the SSI to a codec with more than 2 channels, or a multi-codec solution.
What if I change channels_max to 16 for capture and playback? Last time I tried, it would not keep the channels synchronized on the TDM bus. I tried playing with the FIFO settings, which helped, but it still would not start each channel in the right slot reliably. Perhaps I was using an older kernel.
I'll give it a try again. Like I said, we'd love to get our 16-channel board up and running on the MX6, so I'm definitely motivated to put some work in to get it reliable.
Thanks, -Caleb
Arnaud
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index c25a1e8..26e980b 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1082,14 +1082,14 @@ static struct snd_soc_dai_driver fsl_ssi_dai_template = { .playback = { .stream_name = "CPU-Playback", .channels_min = 1,
.channels_max = 2,
.channels_max = 4, .rates = FSLSSI_I2S_RATES, .formats = FSLSSI_I2S_FORMATS, }, .capture = { .stream_name = "CPU-Capture", .channels_min = 1,
.channels_max = 2,
.channels_max = 4, .rates = FSLSSI_I2S_RATES, .formats = FSLSSI_I2S_FORMATS, },
Le 17/09/2015 00:07, Caleb Crome a écrit :
Ah Ha! thank you Arnaud! I'll understand this stuff eventually. I'll implement as you've shown and see how it goes.
I see that you have 'fsl,imx-audio-foo', using an SSI. from what I can see the SSIs only support channels_max=2. How do you get multi-channel TDM to work with the freescale SSI port? I need to get this TDM working on both TI & Freescale, and the freescale has been quite problematic because the multi-channel doesn't seem to be supported in the driver.
I have just checked out and verified that
http://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git/
sound/soc/fsl/*ssi*
all have channels_max = 2
We're very motivated to get this working on the freescale MX6, and getting it to work sure does not seem as simple as changing 2 to a bigger number :-)
Thanks, -Caleb
On Wed, Sep 16, 2015 at 2:57 AM, arnaud.mouiche@invoxia.com arnaud.mouiche@invoxia.com wrote:
Hello Caleb,
Multi-codec support is now working fine the current linux releases. for previous releases, check if commit [PATCH] ASoC: dapm: Don't add prefix to widget stream name is included. I remember this is the last required patch.
After that, I don't know if the "simple-card" can be configured for multi codec support as you wish.
On my side, I cooked a particular sound card for the purpose. here is some tips:
struct snd_soc_dai_link_component codecs[2]; struct snd_soc_codec_conf codecs_conf[2];
[...] for (num_codecs=0; num_codecs<2; num_codecs++) { struct device_node *of_node; of_node = of_parse_phandle(pdev->dev.of_node, "audio-codec", num_codecs); if (!of_node) break; data->codecs[num_codecs].of_node = of_node; data->codecs[num_codecs].dai_name = "tlv320aic3x-foo";
/* add a "C2" name prefix for every control of the 2nd codec */ data->codecs_conf[num_codecs].of_node = of_node; if (num_codecs == 1) { data->codecs_conf[num_codecs].name_prefix = "C2"; } }
[...] data->dai.codecs = data->codecs; data->dai.num_codecs = num_codecs; data->card.dai_link = &data->dai; data->card.codec_conf = data->codecs_conf; data->card.num_configs = num_codecs; [...]
On DTS side, I have something like
sound@0 { compatible = "fsl,imx-audio-foo"; model = "foo-audio"; ssi-controller = <&ssi1>; /* * list phandles for the 2 codecs used in the same TDM network */ audio-codec = <&codec1>, <&codec2>; [...] };
You also need to have a particular hw_params() method to dispatch the TDM and sysclk configuration to your codecs, making the difference between your first codec (your bus master) and the others.
Arnaud
Le 15/09/2015 03:07, Caleb Crome a écrit :
(re-sending hope it's not a duplicate -- I think I must have had HTML in my previous email and it was ignored)
Hi all, What are the current best practices to specify multiple codecs on one sound card, with all codecs sharing a single TDM bus?
I have a card with up to 32 TLV320AIC33 codecs on it. For the moment, we can limit the discussion to only 16 codecs, so we don't have to get extra serializers involved...
I currently have something like this in my am335x-boneblack.dts file:
(i.e. attempting to set tdm slot width and num in the i2c codec)
but it doesn't seem to be picking up the fact that I want all the codecs linked together as one.
(FYI, the first codec, i.e. tlv32aic3x_a should be the master, and all others including the CPU will be slaves).
Thanks for any help!
BTW, I'm happy to use whatever kernel will support me with the minimal amount of effort, currently using 4.1
Thank you! -Caleb
&i2c1 { ... tlv320aic3x_a: tlv320aic3x@18 { #sound-dai-cells = <1>; compatible = "ti,tlv320aic3x"; reg = <0x18>; tdm-offset = <0>; status = "okay"; }; tlv320aic3x_b: tlv320aic3x@19 { #sound-dai-cells = <1>; compatible = "ti,tlv320aic3x"; reg = <0x19>; tdm-offset = <32>; status = "okay"; }; tlv320aic3x_c: tlv320aic3x@1a { #sound-dai-cells = <1>; compatible = "ti,tlv320aic3x"; reg = <0x1a>; tdm-offset = <64>; status = "okay"; }; tlv320aic3x_d: tlv320aic3x@1b { #sound-dai-cells = <1>; compatible = "ti,tlv320aic3x"; reg = <0x1b>; tdm-offset = <96>; status = "okay"; }; };
&mcasp0 { #sound-dai-cells = <0>; pinctrl-names = "default"; pinctrl-0 = <&mcasp_0_pins_default>; status = "okay";
op-mode = <0>; /* MCASP_IIS_MODE */ tdm-slots = <16>; num-serializer = <16>; serial-dir = < /* 0: INACTIVE, 1: TX, 2: RX */ 0 0 1 2 0 0 0 0 0 0 0 0 0 0 0 0
;
tx-num-evt = <1>; rx-num-evt = <1>; };
/ { sound { compatible = "simple-audio-card"; simple-audio-card,name = "puppy-audio"; simple-audio-card,mclk-fs = <256>; system-clock-frequency = <12288000>; status = "okay"; simple-audio-card,bitclock-master = <&tlv320aic3x_a>; simple-audio-card,frame-master = <&tlv320aic3x_a>; simple-audio-card,widgets = "Line", "Line Out", "Line", "Line In"; simple-audio-card,routing = "Line Out", "HPLOUT", "Line Out", "HPROUT", "Line In", "LINE1L", "Line In", "LINE1R";
simple-audio-card,dai-link@0 { format = "left_j"; cpu { sound-dai = <&mcasp0>; }; codec { sound-dai = <&tlv320aic3x_a 0>; dai-tdm-slot-num = <0>; dai-tdm-slot-width = <16>; }; }; /**** The stuff below doesn't work -- I can't figure out how to get the name_prefixes set on each codec... .How do I set up so that each codec gets its own name prfix and so that the soc core thinks all codecs are on the same DAI? **** */
// simple-audio-card,dai-link@1 { // format = "left_j"; // cpu { // sound-dai = <&mcasp0>; // }; // codec { // sound-dai = <&tlv320aic3x_b 0>; // dai-tdm-slot-num = <2>; // dai-tdm-slot-width = <16>; // }; // }; // simple-audio-card,dai-link@2 { // format = "left_j"; // cpu { // sound-dai = <&mcasp0>; // }; // codec { // sound-dai = <&tlv320aic3x_c>; // dai-tdm-slot-num = <4>; // dai-tdm-slot-width = <16>; // }; // }; // simple-audio-card,dai-link@3 { // format = "left_j"; // cpu { // sound-dai = <&mcasp0>; // }; // codec { // sound-dai = <&tlv320aic3x_d>; // dai-tdm-slot-num = <6>; // dai-tdm-slot-width = <16>; // }; // }; }; }; _______________________________________________ Alsa-devel mailing list Alsa-devel@alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
Alsa-devel mailing list Alsa-devel@alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
hello,
Le 17/09/2015 15:38, Caleb Crome a écrit :
On Thu, Sep 17, 2015 at 1:51 AM, arnaud.mouiche@invoxia.com arnaud.mouiche@invoxia.com wrote:
Hello Caleb,
freescale SSI is fine with more than 2 channels. I have planed to publish a set of patch in this direction but we still have some corner cases to fix first (rare issues with channels alignment)
I'd love to see anything you're willing to show :-)
Yet, this following is far enough to have it working. And since the max/min rate is at the end the intersection of what the SSI, the codec and the card are declaring, it will not change anything until you connect the SSI to a codec with more than 2 channels, or a multi-codec solution.
What if I change channels_max to 16 for capture and playback? Last time I tried, it would not keep the channels synchronized on the TDM bus. I tried playing with the FIFO settings, which helped, but it still would not start each channel in the right slot reliably. Perhaps I was using an older kernel.
you put the finger on what's hurting ! ;)
In fact, we made 8 channels working in reliable way on imx50, with predictive latency ... with an old kernel (3.17) ... based on freescale BSP... on sound/soc/imx/imx-ssi.c (something that doesn't exist outside freescale BSP)
Well, just to say it is doable on any recent imx since the hardware is not the issue. But the job was a set of awful hacks.
My plan was to re-do all this job on an upstream kernel, in a way it can be accepted. Unfortunately you will have to wait, I'm under pressure for other stuff right now.
Regards, arnaud
I'll give it a try again. Like I said, we'd love to get our 16-channel board up and running on the MX6, so I'm definitely motivated to put some work in to get it reliable.
Thanks, -Caleb
On Thu, Sep 17, 2015 at 7:33 AM, arnaud.mouiche@invoxia.com arnaud.mouiche@invoxia.com wrote:
hello,
Le 17/09/2015 15:38, Caleb Crome a écrit :
On Thu, Sep 17, 2015 at 1:51 AM, arnaud.mouiche@invoxia.com arnaud.mouiche@invoxia.com wrote:
Hello Caleb,
freescale SSI is fine with more than 2 channels. I have planed to publish a set of patch in this direction but we still have some corner cases to fix first (rare issues with channels alignment)
I'd love to see anything you're willing to show :-)
Yet, this following is far enough to have it working. And since the max/min rate is at the end the intersection of what the SSI, the codec and the card are declaring, it will not change anything until you connect the SSI to a codec with more than 2 channels, or a multi-codec solution.
What if I change channels_max to 16 for capture and playback? Last time I tried, it would not keep the channels synchronized on the TDM bus. I tried playing with the FIFO settings, which helped, but it still would not start each channel in the right slot reliably. Perhaps I was using an older kernel.
you put the finger on what's hurting ! ;)
In fact, we made 8 channels working in reliable way on imx50, with predictive latency ... with an old kernel (3.17) ... based on freescale BSP... on sound/soc/imx/imx-ssi.c (something that doesn't exist outside freescale BSP)
Well, just to say it is doable on any recent imx since the hardware is not the issue. But the job was a set of awful hacks.
Sorry, I don't quite understand. Are you saying that on a recent kernel (say 4.1 or 4.2), you think it will just work. Or are yo saying that even on a modern kernel it's a set of awful hacks? And should I use fsl_ssi.c or imx-ssi.c (device tree/non device tree) version?
In short, what's your best current recommendation to get it working today?
My plan was to re-do all this job on an upstream kernel, in a way it can be accepted. Unfortunately you will have to wait, I'm under pressure for other stuff right now.
Heh, understood. If there's anything I can do to help, let me know.
Thanks, -Caleb
Regards, arnaud
I'll give it a try again. Like I said, we'd love to get our 16-channel board up and running on the MX6, so I'm definitely motivated to put some work in to get it reliable.
Thanks, -Caleb
Le 17/09/2015 17:34, Caleb Crome a écrit :
On Thu, Sep 17, 2015 at 7:33 AM, arnaud.mouiche@invoxia.com arnaud.mouiche@invoxia.com wrote:
hello,
Le 17/09/2015 15:38, Caleb Crome a écrit :
On Thu, Sep 17, 2015 at 1:51 AM, arnaud.mouiche@invoxia.com arnaud.mouiche@invoxia.com wrote:
Hello Caleb,
freescale SSI is fine with more than 2 channels. I have planed to publish a set of patch in this direction but we still have some corner cases to fix first (rare issues with channels alignment)
I'd love to see anything you're willing to show :-)
Yet, this following is far enough to have it working. And since the max/min rate is at the end the intersection of what the SSI, the codec and the card are declaring, it will not change anything until you connect the SSI to a codec with more than 2 channels, or a multi-codec solution.
What if I change channels_max to 16 for capture and playback? Last time I tried, it would not keep the channels synchronized on the TDM bus. I tried playing with the FIFO settings, which helped, but it still would not start each channel in the right slot reliably. Perhaps I was using an older kernel.
you put the finger on what's hurting ! ;)
In fact, we made 8 channels working in reliable way on imx50, with predictive latency ... with an old kernel (3.17) ... based on freescale BSP... on sound/soc/imx/imx-ssi.c (something that doesn't exist outside freescale BSP)
Well, just to say it is doable on any recent imx since the hardware is not the issue. But the job was a set of awful hacks.
Sorry, I don't quite understand. Are you saying that on a recent kernel (say 4.1 or 4.2), you think it will just work. Or are yo saying that even on a modern kernel it's a set of awful hacks? And should I use fsl_ssi.c or imx-ssi.c (device tree/non device tree) version?
- I'm saying it is working on recent upstream kernel, using sound/soc/fsl/fsl_ssi.c, but with some channel sync issues in rare conditions (at least 4 channels case, opening multiple SSI at the same time). So, you should test on your side with you setup, and simply be prepared to not see it working as a charm.
- we made it work perfectly, but on an old freescale BSP, where freescale rewrites a SSI driver different from the already 2 drivers (device tree/non device tree) available today in upstream + lot of awful hacks => so, this is not a hardware issue.
- when I will have time, I will start from the upstream driver, and make it work correctly with 8 channels (or more)
arnaud
In short, what's your best current recommendation to get it working today?
My plan was to re-do all this job on an upstream kernel, in a way it can be accepted. Unfortunately you will have to wait, I'm under pressure for other stuff right now.
Heh, understood. If there's anything I can do to help, let me know.
Thanks, -Caleb
Regards, arnaud
I'll give it a try again. Like I said, we'd love to get our 16-channel board up and running on the MX6, so I'm definitely motivated to put some work in to get it reliable.
Thanks, -Caleb
Got it. Thanks Arnaud.
-Caleb
On Thu, Sep 17, 2015 at 9:09 AM, arnaud.mouiche@invoxia.com arnaud.mouiche@invoxia.com wrote:
Le 17/09/2015 17:34, Caleb Crome a écrit :
On Thu, Sep 17, 2015 at 7:33 AM, arnaud.mouiche@invoxia.com arnaud.mouiche@invoxia.com wrote:
hello,
Le 17/09/2015 15:38, Caleb Crome a écrit :
On Thu, Sep 17, 2015 at 1:51 AM, arnaud.mouiche@invoxia.com arnaud.mouiche@invoxia.com wrote:
Hello Caleb,
freescale SSI is fine with more than 2 channels. I have planed to publish a set of patch in this direction but we still have some corner cases to fix first (rare issues with channels alignment)
I'd love to see anything you're willing to show :-)
Yet, this following is far enough to have it working. And since the max/min rate is at the end the intersection of what the SSI, the codec and the card are declaring, it will not change anything until you connect the SSI to a codec with more than 2 channels, or a multi-codec solution.
What if I change channels_max to 16 for capture and playback? Last time I tried, it would not keep the channels synchronized on the TDM bus. I tried playing with the FIFO settings, which helped, but it still would not start each channel in the right slot reliably. Perhaps I was using an older kernel.
you put the finger on what's hurting ! ;)
In fact, we made 8 channels working in reliable way on imx50, with predictive latency ... with an old kernel (3.17) ... based on freescale BSP... on sound/soc/imx/imx-ssi.c (something that doesn't exist outside freescale BSP)
Well, just to say it is doable on any recent imx since the hardware is not the issue. But the job was a set of awful hacks.
Sorry, I don't quite understand. Are you saying that on a recent kernel (say 4.1 or 4.2), you think it will just work. Or are yo saying that even on a modern kernel it's a set of awful hacks? And should I use fsl_ssi.c or imx-ssi.c (device tree/non device tree) version?
- I'm saying it is working on recent upstream kernel, using
sound/soc/fsl/fsl_ssi.c, but with some channel sync issues in rare conditions (at least 4 channels case, opening multiple SSI at the same time). So, you should test on your side with you setup, and simply be prepared to not see it working as a charm.
- we made it work perfectly, but on an old freescale BSP, where freescale
rewrites a SSI driver different from the already 2 drivers (device tree/non device tree) available today in upstream + lot of awful hacks => so, this is not a hardware issue.
- when I will have time, I will start from the upstream driver, and make it
work correctly with 8 channels (or more)
arnaud
In short, what's your best current recommendation to get it working today?
My plan was to re-do all this job on an upstream kernel, in a way it can be accepted. Unfortunately you will have to wait, I'm under pressure for other stuff right now.
Heh, understood. If there's anything I can do to help, let me know.
Thanks, -Caleb
Regards, arnaud
I'll give it a try again. Like I said, we'd love to get our 16-channel board up and running on the MX6, so I'm definitely motivated to put some work in to get it reliable.
Thanks, -Caleb
participants (2)
-
arnaud.mouiche@invoxia.com
-
Caleb Crome