[pulseaudio-discuss] My attempt to reduce latency with pacat and tvtime

Steven Elliott selliott4 at austin.rr.com
Mon May 21 18:10:06 PDT 2012


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.

-- 
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