On Mon, Oct 20, 2003, Paul Davis wrote:
> the output latency is always roughly the size of the hardware
> buffer. there is no way to avoid this. the best you could ever do
> would be one period, where there are two periods per buffer (i.e half
> the buffer size). when i say "the best you could do", the truth is
> that you probably can't do it. the very first sample you deliver to
> the hardware will have a delay of 1 period, each subsequent sample has
> a steadily increasing delay all the way up to the last sample, which
> is delayed by the length of the entire hardware buffer (2
> periods). this assumes that delivery is basically instaneous:
> obviously, if you take time to deliver it, the delay drops, but the
> total delay (from when you could have written to when the sound is
> audible) is the same.

I am working on a voice over IP application. We would like delays from
the write to soundcard to the actual playout lower than 50ms. Do you
think it is impossible with ALSA ? With OSS free the delay we got were
quite good, without any tweaking. But ALSA is said to have better
support for full-duplex.

Maybe using a small buffer on the sound card should then be a better
solution ? I tried with a buffer of 50 ms, and the playout is fine until
for some reason the buffer gets full. At that point I get an EPIPE error
on the next write, I then do a snd_pcm_prepare, can write again, but
after that both reading and writing give a lot of errors, and after a
very short while a write will block for more than one second. Are those
ALSA related problems, or hardware ones ?

Thanks,
-- 
Jobi
Please excuse me for the dupplicate previous e-mail


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