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]>:

> Maybe also this can help:
>
> http://mailman.alsa-project.org/pipermail/alsa-devel/2013-February/059433.html
>
> R
>
> On 8 janv. 2016, at 23:49, Robin Scheibler <[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
> 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]>
> 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]>:
>
>> 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]>
>> 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]>:
>>
>>> 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]>:
>>>
>>>> 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] 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
>>>> 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
>>>>>
>>>>> device tree:
>>>>> 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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>>>
>>>
>>
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