linux-kernel - Re: [PATCH] ASoC: pcm: allow changing the playback/capture rates for symmetric links

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Date:	Mon, 2 May 2016 09:43:53 +0200
From:	Peter Rosin <peda@...ntia.se>
To:	Mark Brown <broonie@...nel.org>
CC:	<alsa-devel@...a-project.org>, Liam Girdwood <lgirdwood@...il.com>,
	Jaroslav Kysela <perex@...ex.cz>, Takashi Iwai <tiwai@...e.de>,
	<linux-kernel@...r.kernel.org>
Subject: Re: [PATCH] ASoC: pcm: allow changing the playback/capture rates for
 symmetric links

On 2016-04-28 12:38, Mark Brown wrote:
> On Wed, Apr 27, 2016 at 10:45:48PM +0200, Peter Rosin wrote:
>> I haven't looked really closely at the userspace side of this, but the
>> big picture is that we're using a (closed source) library that in its
>> documentation has an example where they open /dev/dsp like this and
>> feeds the fd to the lib. We are simply replicating that in our code.
>> I don't know if the library does anything OSSy with the fd, or if it
>> would be equally happy with an ALSA fd.
> 
> If it's using OSS it's unlikely to work with ALSA, the two ABIs are
> completely different.  You need to talk to your proprietary software
> vendor about updating their software.  There are also some userspace
> utilities that try to do the remapping by wrapping things there but I
> can't remember the names off the top of my head, it's been a long time.

Ok, I managed to hook it up with ALSA instead but am now bitten by
a completely different problem.

The code now does this (error checks elided, there are no errors
reported in the real code which does check for errors):

snd_pcm_open(&pcm, "default", SND_PCM_STREAM_PLAYBACK, 0);
snd_pcm_set_params(pcm,
	SND_PCM_FORMAT_S16_LE,
	SND_PCM_ACCESS_RW_INTERLEAVED,
	2, 22050, 0, 1000000);
while (!end) {
	/* code filling in stereo_buf elided */
	snd_pcm_writei(pcm, stereo_buf, sample_count);
}
snd_pcm_drain(pcm);
snd_pcm_close(pcm);

There is no rate control in this, the code relies on snd_pcm_writei
to block if there is no room. This part appears to work as expected.
sample_count is not constant, but it is mostly 4096 and never bigger
than that.

The problem is in the drain/close part. If I leave out snd_pcm_drain
I naturally lose the tail of the output, but if I have it I get what
I guess is a repeat of the content in some circular buffer. I.e. the
very last part of the played sound is played twice(ice).

If I instead write the generated samples to a file and play them
with aplay, all is fine, so I appear to be misunderstanding the
alsa api?

Anything obviously wrong in the above code?

Cheers,
Peter