underrun problems after setting parameters with snd_pcm_set_params()

[robert rozee]

   hi Kai,
       well, that's embarrassing! the problem was indeed largely with the
   latency setting.

   the min(500000, sample_length) was to allow for samples of as short as
   50ms.
   i had found that short clips they were not playing at all - but that
   was before i added in
   snd_pcm_drain(handle).

   with latency fixed at 500000 and snd_pcm_drain(handle) at the end,
   everything
   now seems to be working perfectly!

   much appreciate your help   :-)


   cheers,
   rob   :-)


   robert rozee
   p.o. box 25-232
   christchurch, new zealand
   phone: +64 21 123-2603
   email: rozee at mail.com


   Sent: Wednesday, April 29, 2020 at 3:52 AM
   From: "Kai Vehmanen" <kai.vehmanen at linux.intel.com>
   To: "robert rozee" <rozee at mail.com>
   Cc: alsa-devel at alsa-project.org
   Subject: Re: underrun problems after setting parameters with
   snd_pcm_set_params()
   Hey,
   On Tue, 28 Apr 2020, robert rozee wrote:
   > the problem only occurs with SHORT sound clips. i found that clips
   longer than
   > 1111ms played without flaw. clips less that 500ms generally failed to
   play to
   > the end, but there was no error code returned.
   in general, given you have sources available to all the popular apps,
   it's
   good to check how other apps use the API. I.e. aplay.c may be useful
   simple additional reference for you to see how to use the interfaces.
   I think you are missing draining the samples at the end, and then
   your latency is setting is incorrect. I.e.
   > snd_pcm_set_params(handle, SND_PCM_FORMAT_U8,
   > SND_PCM_ACCESS_RW_INTERLEAVED,
   > 1, // number of channels
   > 48000, // bitrate (bps)
   > 1, // resampling on/off
   > min(500000, sample_length)); // latency
   That sample_length does not look, the unit is usecs here. Please try
   just putting latency of 500000 (i.e. 0.5sec).
   > // send out current buffer content
   >   frames=snd_pcm_writei(handle, @buffer, frames_to_write_now);
   >   if frames<0 then frames=snd_pcm_recover(handle, frames, 0);
   >   if frames<0 then break;
   > }
   When you have finished writing the audio samples to the ALSA device,
   you
   need to wait until ALSA has a chance to play all samples out. If you
   look
   at playback_go() in aplay.c, you'll see:
   snd_pcm_nonblock(handle, 0);
   snd_pcm_drain(handle);
   snd_pcm_nonblock(handle, nonblock);
   ... at the end.
   Br, Kai