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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