>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. ------------------------------------------------------- This SF.net email is sponsored by OSDN developer relations Here's your chance to show off your extensive product knowledge We want to know what you know. Tell us and you have a chance to win $100 http://www.zoomerang.com/survey.zgi?HRPT1X3RYQNC5V4MLNSV3E54 _______________________________________________ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
