On Mon, 2012-05-21 at 21:09 +0600, Alexander E. Patrakov wrote: > 2012/5/21 Pierre-Louis Bossart <pierre-louis.bossart at linux.intel.com>: > > I am not sure if there's really a problem. Increasing the audio latency > > doesn't necessarily result in A/V sync issues. As long as the A/V sync is > > done by querying how many samples are queued instead of using the number of > > samples pushed into PulseAudio, you should be able to use pretty much > > whatever buffer size you want. It's the same issue with ALSA, if you use > > large buffers and base the A/V on the number of samples written to the ring > > buffer, A/V sync will be off. Use snd_pcm_delay() and you'll be fine. > > Dear Pierre-Louis, > > this thread is based on a misunderstanding. The original poster, > instead of asking someone to fix the original problem (see below), > asks you about workarounds, and you don't see the big picture. Thanks for that explanation. Yes, the issue is that tvtime does not handle A/V sync and yes, admittedly I'm trying to fix the problem somewhere other than where it should be fixed ideally. But I thought this might be a case where it would be easiest to fix the problem in PulseAudio. -- ------------------------------------------------------------------------ | Steven Elliott | http://selliott.org | selliott4 at austin.rr.com | ------------------------------------------------------------------------
