On Tue, Oct 25, 2011 at 9:31 AM, Pierre-Louis Bossart <pierre-louis.bossart at linux.intel.com> wrote: > > Hi Dylan, > please use plain text in your messages, HTML makes it hard to quote your > text... > > Your idea of having fewer wakes makes sense. Just technically I think you > are confusing latency with frame size. if you want to use 10ms frames for > speech processing, you will have a 20ms latency, be that with ALSA or > PulseAudio. Understood, I wasn't worrying about frame sizing yet. Just passing through unprocessed samples to start. If I was to have 10ms frames (likely) I'd want to have an audio wake from the output every 10ms. > Also I think you've hit an issue with PulseAudio's inner details. The idea > is that there's a server side buffer that has the same length than the ALSA > ring buffer. This makes sense for low-power audio, so that you can wake-up > at the last moment and quickly fill the ring buffer. For low-latency, this > might not be such a good idea, since it entails many useless wakes and the > client does have the data available. This behavior is enforced in > pulsecore/protocol-native.c, it may be possible to patch this code to reduce > the server side buffer to zero (or minreq). Thanks for the pointer, I'll take a look in that area and see if it's something easy. > This might be a good point to bring to Lennart, if he still remembers what > he wrote a couple of years ago. Lennart are you still with us? > -Pierre > > > _______________________________________________ > pulseaudio-discuss mailing list > pulseaudio-discuss at lists.freedesktop.org > http://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss
