Hi Pablo,
I took a look at your code.
> static int pcm3168_hw_params(struct snd_pcm_substream *substream,
> struct snd_pcm_hw_params *params)
> {
> struct snd_soc_pcm_runtime *rtd = substream->private_data;
> struct snd_soc_dai *codec_dai = rtd->codec_dai;
> struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
> struct snd_soc_codec *codec = rtd->codec;
> struct snd_soc_card *soc_card = rtd->card;
> struct platform_device *pdev = to_platform_device(soc_card->dev);
> int ret = 0;
> unsigned sysclk = ((struct snd_soc_card_drvdata_davinci *)
> snd_soc_card_get_drvdata(soc_card))->sysclk;
>
> /* set the codec system clock */
> ret = snd_soc_dai_set_sysclk(codec_dai, 0, sysclk, SND_SOC_CLOCK_OUT);
> if (ret < 0)
> return ret;
>
> /* set the CPU system clock */
> ret = snd_soc_dai_set_sysclk(cpu_dai, 0, sysclk, SND_SOC_CLOCK_OUT);
> if (ret < 0)
> return ret;
>
> return 0;
> }
I see that you are calling snd_soc_dai_set_sysclk for both codec_dai and
cpu_dai. I think that is the error. In your case codec_dai is the PCM3168 chip
and cpu_dai is the processor McASP module. Sysclk only makes sense for McASP as
far as I understand. Try commenting out
> /* set the codec system clock */
> ret = snd_soc_dai_set_sysclk(codec_dai, 0, sysclk, SND_SOC_CLOCK_OUT);
> if (ret < 0)
> return ret;
I would also suggest you read up about the McASP clock generation structure in
the sitara documentation (~7000 pages document available from TI). This helped
me understand how the McASP module works.
Another resource I used is the linux cross-reference
http://lxr.free-electrons.com/
I used it to look up where functions are defined and what they do.
Cheers,
Robin
> On 10 janv. 2016, at 18:02, Pablo Fonovich <[email protected]> wrote:
>
> Hi: im using 24Mhz, that is provided by de beaglebone... i used printk to see
> that sysclk value and it is correct... i get the value as is stated in the
> evm_hw_params function.
> I dont see why the function returns error... but is there a list of error
> codes to see what -524 means?
> Thanks very much
>
> El ene 10, 2016 1:55 PM, "Robin Scheibler" <[email protected]
> <mailto:[email protected]>> escribió:
> Hi Pablo,
>
>> On 10 janv. 2016, at 01:48, Pablo Fonovich <[email protected]
>> <mailto:[email protected]>> wrote:
>>
>> Hi Robin:
>> with printk statements, i've isolated the error to this instruction:
>>
>> ret = snd_soc_dai_set_sysclk(codec_dai, 0, sysclk, SND_SOC_CLOCK_OUT);
>>
>> but i don't know what is wrong... the sysclk variable is:
>>
>> unsigned int sysclk = ((struct snd_soc_card_drvdata_davinci *)
>> snd_soc_card_get_drvdata(soc_card))->sysclk;
>> Just linke in the evm_hw_params
>
> What value of sysclk are you providing ?
>
>>
>> I've noticed that you used:
>>
>> ret = snd_soc_dai_set_sysclk(cpu_dai, 0, 0, SND_SOC_CLOCK_IN);
>>
>> If i'm getting it right, this is to set the cpu_dai sysclock, but i dont get
>> why you used 0 in the frequency parameter. Also you never used the function
>> to set the codec_dai sysclk... Is it not necessary? I'll try commenting this
>> line
>
> In my case, MCLK is provided to the chipset by an external crystal. Thus
> there is no sysclk for me. BCLK and LRCLK are also provided by the chipset to
> McASP.
>
> Best,
> Robin
>
>>
>>
>>
>>
>> 2016-01-09 11:55 GMT-03:00 Pablo Fonovich <[email protected]
>> <mailto:[email protected]>>:
>> Hi Robin:
>> I intend to implement a portable multitrack recorder, that's why i choose
>> this codec, it has many inputs and outputs, good sample rate and bit
>> resolution, and all in a single chip.
>> I can't find wendelin's mail direction, so i think i'll just write to that
>> posts, but they are a bit old i think. I have already looked at them, but i
>> was more confused by that time, perhaps now i can understand a bit more.
>>
>> If i find out anything i will continue posting here, so if someone is
>> interested in this project can look at ti...
>>
>> Please if you find any documentation tell me!
>>
>> Thanks
>>
>> 2016-01-08 19:51 GMT-03:00 Robin Scheibler <[email protected]
>> <mailto:[email protected]>>:
>> Maybe also this can help:
>> http://mailman.alsa-project.org/pipermail/alsa-devel/2013-February/059433.html
>>
>> <http://mailman.alsa-project.org/pipermail/alsa-devel/2013-February/059433.html>
>>
>> R
>>
>>> On 8 janv. 2016, at 23:49, Robin Scheibler <[email protected]
>>> <mailto:[email protected]>> wrote:
>>>
>>> Hi Pablo,
>>>
>>> I don’t know for 4.2. Try to read the
>>> sound/alsa/soc/davinci/davinci-mcasp.c file to understand which
>>> functionalities are implemented. I used the kernel from Robert C Nelson’s
>>> repo on github.
>>>
>>> After a quick search on google, I saw you are not the first person to
>>> attempt to connect this codec to the BBB.
>>> http://www.spinics.net/lists/alsa-devel/msg41291.html
>>> <http://www.spinics.net/lists/alsa-devel/msg41291.html>
>>> Maybe you can try to contact this Wendelin Klimann.
>>>
>>> BTW, this PCM3168 seems like a very neat chip. What exactly are you
>>> building ?
>>>
>>> Cheers,
>>> Robin
>>>
>>>> On 8 janv. 2016, at 23:17, Pablo Fonovich <[email protected]
>>>> <mailto:[email protected]>> wrote:
>>>>
>>>> Thanks Robin,
>>>> Im using latest stable kernel, think is 4.2, i can check when i get
>>>> home... i will add printk statements, as you suggested...
>>>> Another question. I inted to use one mcasp serializer for 4 channels... in
>>>> the am335x datasheet it seems possible, so i configured 4 tdm slots in the
>>>> device tree instead of 2, but if i try to record with -c 6 it says again
>>>> the "channels count non available". Do you know if its possible? its quite
>>>> frustrating to work without propper documentation, but in need this
>>>> project to get my title...
>>>>
>>>> And last thing, where you able to get alsamixer working?
>>>> Thanks very much
>>>>
>>>> 2016-01-08 19:03 GMT-03:00 Robin Scheibler <[email protected]
>>>> <mailto:[email protected]>>:
>>>> Hi Pablo,
>>>>
>>>> What is the kernel version you are using ? I have finally used kernel 3.14
>>>> because in 3.8 it seems that the mcasp driver implementation could not
>>>> support more than two channels.
>>>>
>>>> I was in the same position as you. I mostly dug as much as possible from
>>>> TI documentation and also read the drivers for other chipsets in the
>>>> kernel code. Then, I also used
>>>> the sitara am355xx data sheet to understand how McASP works. It took me
>>>> quite some time to piece together how things work. Still now I do not
>>>> fully understand. The problem is that there is no proper documentation for
>>>> the ALSA SoC wrt embedded platforms such as the BBB.
>>>>
>>>> Note that the TI example code might not be for exactly the same kernel
>>>> version you are using. There might be discrepancies.
>>>>
>>>> I suggest you run a search on the source code to find where the error
>>>> “ASoC: machine hw_params failed” is generated. You can add printk
>>>> statements in your code so that you can later check in the kernel log that
>>>> things are happening as you expect.
>>>>
>>>> Cheers,
>>>> Robin
>>>>
>>>>> On 8 janv. 2016, at 01:25, Pablo Fonovich <[email protected]
>>>>> <mailto:[email protected]>> wrote:
>>>>>
>>>>> sorry, the message also states:
>>>>>
>>>>> arecord: set_params:1297: Unable to install hw params:
>>>>> ACCESS: RW_INTERLEAVED
>>>>> FORMAT: S24_LE
>>>>> SUBFORMAT: STD
>>>>> SAMPLE_BITS: 32
>>>>> FRAME_BITS: 128
>>>>> CHANNELS: 4
>>>>> RATE: 96000
>>>>> PERIOD_TIME: (21333 21334)
>>>>> PERIOD_SIZE: 2048
>>>>> PERIOD_BYTES: 32768
>>>>> PERIODS: 4
>>>>> BUFFER_TIME: (85333 85334)
>>>>> BUFFER_SIZE: 8192
>>>>> BUFFER_BYTES: 131072
>>>>> TICK_TIME: 0
>>>>> # aplay -f S24_LE -r 96000 ../Yam*wav
>>>>> Warning: format is changed to S16_LE
>>>>> Playing WAVE '../Yamaha-TG100-Ocarina-C5.wav' : Signed 16 bit Little
>>>>> Endian, Rate 44100 Hz, Stereo
>>>>> [ 102.291787] davinci_evm sound: ASoC: machine hw_params failed: -524
>>>>> aplay: set_params:1297: Unable to install hw params:
>>>>> ACCESS: RW_INTERLEAVED
>>>>> FORMAT: S16_LE
>>>>> SUBFORMAT: STD
>>>>> SAMPLE_BITS: 16
>>>>> FRAME_BITS: 32
>>>>> CHANNELS: 2
>>>>> RATE: 44100
>>>>> PERIOD_TIME: (125011 125012)
>>>>> PERIOD_SIZE: 5513
>>>>> PERIOD_BYTES: 22052
>>>>> PERIODS: 4
>>>>> BUFFER_TIME: (500045 500046)
>>>>> BUFFER_SIZE: 22052
>>>>> BUFFER_BYTES: 88208
>>>>> TICK_TIME: 0
>>>>>
>>>>>
>>>>> 2016-01-07 21:24 GMT-03:00 Pablo Fonovich <[email protected]
>>>>> <mailto:[email protected]>>:
>>>>> Thanks very much, that was a problem. But now another error appears:
>>>>> "davinci_evm sound: ASoC: machine hw_params failed: -524". Do you know
>>>>> what it means, or where can i find the meaning? I assume this means there
>>>>> is something wrong when i created the pcm3168_hw_params in the alsa
>>>>> machine layer... but i don't know the problem... where did you look for
>>>>> info? only in that TI guide? I attach my files, and if is not much
>>>>> trouble for you i would be grateful if you help me...
>>>>>
>>>>> i've noticed that some functions you used in davinci_evm.c where not
>>>>> explained in that guide... and it's very difficult for a noob like me in
>>>>> this subject for understainding what to do... I've just based my
>>>>> pcm3168.c in another driver, and i really don't quite understand much of
>>>>> what is going on, nor found any simple documentation about it. If you
>>>>> could give me some hints it would be great!
>>>>>
>>>>> Thanks very much..
>>>>>
>>>>> 2016-01-07 17:30 GMT-03:00 Robin Scheibler <[email protected]
>>>>> <mailto:[email protected]>>:
>>>>> Hi Pablo,
>>>>>
>>>>> With McASP it is not possible to capture less than 2 channels. Try with
>>>>> -c 2 argument.
>>>>>
>>>>> Best,
>>>>> Robin
>>>>>
>>>>>> On 7 janv. 2016, at 20:16, [email protected]
>>>>>> <mailto:[email protected]> wrote:
>>>>>>
>>>>>> Hi Robin:
>>>>>>
>>>>>> I'm doing a similar project, but with a PCM3168 codec... I also followed
>>>>>> the guide
>>>>>> http://processors.wiki.ti.com/index.php/Sitara_Linux_Audio_DAC_Example
>>>>>> <http://processors.wiki.ti.com/index.php/Sitara_Linux_Audio_DAC_Example>
>>>>>> Anyway there are lots of things that are not very clear for me...
>>>>>> <http://processors.wiki.ti.com/index.php/Sitara_Linux_Audio_DAC_Example>
>>>>>>
>>>>>> In my case, i use codec as clock slave, and synch, so the problem you
>>>>>> had should not affect to me... But when i try to record it says:
>>>>>>
>>>>>> arecord -D hw:0,0 -f S24_LE -c 1 -r 96000 -t wav test.wav
>>>>>> Recording WAVE 'test.wav' : Signed 24 bit Little Endian, Rate 96000 Hz,
>>>>>> Mono
>>>>>> arecord: set_params:1239: Channels count non available
>>>>>>
>>>>>> Could you help me?
>>>>>>
>>>>>> On Sunday, May 24, 2015 at 4:38:43 AM UTC-3, Robin Scheibler wrote:
>>>>>> Dear all,
>>>>>>
>>>>>> I have created a custom cape around the STA321MP
>>>>>> <http://www.st.com/web/en/catalog/sense_power/FM125/CL935/SC534/PF251500d.d24>
>>>>>> chip from ST Microelectronics. It has 6 MEMS microphones and outputs
>>>>>> them on an I2S bus with 3 data lines. I am trying to use mcasp0 on the
>>>>>> boneblack side to receive the data. I am now to the point where the
>>>>>> codec configuration happens correctly (I can see the i2s signals with a
>>>>>> logic analyzer) and a soundcard is detected. However, when I try to
>>>>>> record sound with arecord, I get the following error:
>>>>>>
>>>>>> root@beaglebone:~# arecord -l
>>>>>> **** List of CAPTURE Hardware Devices ****
>>>>>> card 0: EVM [STA321MP EVM], device 0: STA321MP sta321mp-audio-0 []
>>>>>> Subdevices: 1/1
>>>>>> Subdevice #0: subdevice #0
>>>>>> root@beaglebone:~# arecord -D hw:0,0 -f S24_LE -c 6 -r 44100 -t wav
>>>>>> test.wav
>>>>>> Recording WAVE 'test.wav' : Signed 24 bit Little Endian, Rate 44100 Hz,
>>>>>> Channels 6
>>>>>> arecord: pcm_read:1801: read error: Input/output error
>>>>>>
>>>>>> Let me know provide more details.
>>>>>> The codec is frame and bit clocks master
>>>>>> The connection to mcasp0 is the following:
>>>>>> Bit clock <=> P9_12 (mcasp0_aclkr_mux3)
>>>>>> Frame sync clock (L/R) <=> P9_27 (mcasp0_fsr)
>>>>>> Data <=> P9_28 (mcasp0_axr2)
>>>>>> Data <=> P9_30 (mcasp0_axr0)
>>>>>> Data <=> P9_25 (mcasp0_axr3)
>>>>>> I am using kernel 3.14.41 (because 3.8 doesn't have support for mcasp
>>>>>> with >2 channels)
>>>>>> I am using dtb-builder to add my device to the device tree
>>>>>> All the code I have added to the kernel (codec and machine code) and to
>>>>>> the device tree is on github. Here are the links to the specific changes
>>>>>> I have made.
>>>>>>
>>>>>> kernel:
>>>>>> https://github.com/fakufaku/linux/commit/3a6680960412ef0fc3fbcc6f2c0021350039cc92
>>>>>>
>>>>>> <https://github.com/fakufaku/linux/commit/3a6680960412ef0fc3fbcc6f2c0021350039cc92>
>>>>>> device tree:
>>>>>> https://github.com/fakufaku/dtb-rebuilder/commit/127dbf3c82c78cc602058308ed72978b170c99a6
>>>>>>
>>>>>> <https://github.com/fakufaku/dtb-rebuilder/commit/127dbf3c82c78cc602058308ed72978b170c99a6>
>>>>>> It is my first time trying to code audio driver so please be indulgent
>>>>>> :) I have spent a lot of time on this but it seems I am now completely
>>>>>> stuck. I'd really appreciate some help. I'd be happy to provide more
>>>>>> clarifications on anything.
>>>>>>
>>>>>> One thing I do not understand is if I need to add mixer bindings. And
>>>>>> also the audio routing (jack vs codec pins, etc).
>>>>>>
>>>>>> Thank you in advance!
>>>>>> Robin
>>>>>>
>>>>>> --
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>>>>>> <http://beagleboard.org/discuss>
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>>>>>
>>>>>
>>>>> --
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>>>>
>>>>
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>>
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