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, pablo7fonov...@gmail.com 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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