Re: start_threshold having no effect

[Paul Davis <paul@linuxaudiosystems.com>]

>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.

JACK routinely works with 5ms latency on low latency kernels, and 20ms
on regular kernels. its easy, as long as you understand how to write
software correctly.

>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 ?

OSS has *NO* xrun detection. ALSA has it, and its turned on by
default. you need to set the stop threshold to -1, so that ALSA never
stops for xruns and never reports them. 

But please note: if you get EPIPE, it means there was an xrun, which
in turn means that your application was *not keeping up with the audio
interface*. you will normally never know this under OSS, whereas ALSA
offers you the chance to find out.


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