Re: Sound from/to custom ADC/DAC

[pepijn de vos <pepijndevos@xxxxxxxxx>]

Thanks for the advice so far.

Based on this post, I managed to make my own I2S overlay.
https://www.raspberrypi.org/forums/viewtopic.php?t=189152

However, still only two channels.

I found that the Pi can in fact understand TDM, but the SoC can only get 2 of the channels for some reason.
https://github.com/raspberrypi/linux/pull/1982

For the Beagle Bone, I also found someone who makes a cape with a lot of inputs and outputs.

There are some forum post scattered here and there with references to McASP, as you mentioned.

This overlay seems a good starting point, potentially:
https://github.com/ctag-fh-kiel/ctag-face-2-4/blob/master/device-tree-overlays/BB-CTAG-SW-8CH-00A0.dts

So to sum it up, I2S works on the Pi for 2 channels, Beagle Bone is something to look at more closely.

Pepijn

On Wed, Mar 14, 2018 at 3:35 PM, Chris Caudle <chris@xxxxxxxxxxxxxxx> wrote:

Server rejected this the first time for some reason, resending

On Sun, March 11, 2018 8:35 am, pepijn de vos wrote:
> The problem with that seems to be that JACK is in control of the sampling
> rate.

I think you are missing a fundamental aspect of how audio hardware works.
The device which converts to and from analog is always in control of the
sampling rate.  When you say that jackd is in control of the sampling
rate, what that really means is that jackd is following the sampling rate
dictated by the device which you have told jackd to use as the backend.

> It would make sense to write an ALSA driver for what is pretty much a
> custom sound card.

Either an ALSA driver or a custom jackd backend.
The only way that makes sense for audio data transfer to work is that the
audio hardware fills a buffer then signals to software that the data is
ready to be read.  In the other direction the audio hardware converts a
buffer of data to analog then signals to the software that it is ready to
have another buffer filled.  Unless the software is capable of filling a
buffer of data in less time than one sample period (hint: it is not) that
implies at least two buffers, one which is being read by the hardware and
converted to analog a sample at a time, and another buffer which the
software can fill while the first buffer is being converted to analog.

--
Chris Caudle
_______________________________________________
Linux-audio-user mailing list
Linux-audio-user@lists.linuxaudio.org
https://lists.linuxaudio.org/listinfo/linux-audio-user

_______________________________________________
Linux-audio-user mailing list
Linux-audio-user@xxxxxxxxxxxxxxxxxxxx
https://lists.linuxaudio.org/listinfo/linux-audio-user