> >> So now i am a but confused how to get the same kind of latency in my >> own code (using alsa directly rather than through jack) >> >> Basically what happens is when i have a buffersize of 8192 and a >> period of 32 is that snd_pcm_avail_update(capture_handle), will keep >> returning 0 until i have played back the full 8192 samples. After >> that I will start receiving input samples, but obviously the latency >> is now more than 8182 samples… >> >> How can i make alsa not wait until the entire buffer is full? > > maybe that's $GOD's way of telling you to use jack? > jokes aside, why not profit from the flexibility of jack when it doesn't have > any real disadvantages?
Well i might have to go that way if i can’t work this out. But i would prefer to use alsa directly if possible. The reason for me not to use jack, is not that there is anything wrong with it, on the contrary. The reason is just that it doesn’t make sense in my case, i don’t need any inter-app routing for this project, and it would just be another dependency. Adding jack to the mix doesn’t seem like the right solution to the actual problem... > with that kind of i/o, i doubt you're doing some tightly constrained embedded > project :-] > it makes for a very nice delay actually :-) fokke > -- > Jörn Nettingsmeier > Lortzingstr. 11, 45128 Essen, Tel. +49 177 7937487 > > Meister für Veranstaltungstechnik (Bühne/Studio) > Tonmeister VDT > > http://stackingdwarves.net > _______________________________________________ Linux-audio-dev mailing list Linux-audio-dev@lists.linuxaudio.org http://lists.linuxaudio.org/listinfo/linux-audio-dev
