[alsa-devel] Programming ALSA sound device.

[Bankim Bhavsar]

Producer thread writes sound bytes worth 10 ms every 10 ms to a ring buffer.
Consumer thread is responsible for writing sound bytes to ALSA sound
device in a strictly non-blocking fashion.
Consumer thread checks snd_pcm_avail/snd_pcm_avail_update, fetches
minimum of sound bytes available in ring buffer and
availFrames(returned by snd_pcm_avail_update) and writes to sound
device using snd_pcm_writei().
Consumer thread polls every 20 ms. However on consumer thread
snd_pcm_avail/snd_pcm_avail_update() returns 0 at least 3-4 times
consecutively (total 60-80 ms) and on every 5th-6th poll
snd_pcm_avail/snd_pcm_avail_update returns full buffer size
occasionally reporting an under-run. On an average, under-run is
reported at least 10-15 times a minute which is not desirable.

Following are the hardware and software parameter settings.

hardware params:
   16-bit 44100 stereo
   buffer time 200 ms
   period time 25 ms

software params:
   start_threshold: full buffer size
   no setting for avail_min

Alternatively I've tried using wait_for_poll() technique as mentioned
in the example.
http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2pcm_8c-example.html
However poll() unblocks very rapidly and producer ring buffer doesn't
have enough bytes.

What's the best technique for PCM playback for such use cases? Is it
okay to skip calling snd_pcm_avail/snd_pcm_avail_update and directly
attempt writing to ALSA sound device instead?
Let me know if more information is required.

Thanks,
Bankim.