On Wed, Jun 02, 2010 at 01:37:40PM +0200, Julien Claassen wrote: > Hello everyone! > I've just scanned the code of app_jack.c as best I could. It > seems, that the functionality is rather simple: > create a jack_client, if demanded, connect it to somewhere. > > Get Audioframes from a telephone channel (8kHz, signed 16 bit - I believe) > > If necessary resample them to jack_sampl_rate > > Get input from JACK, if necessary resample them to 8kHz > > Write 20ms frames back to the telephone channel.
One problem I can imagine is that the two sample clocks - the one used by Jack and and the 8 kHz one used by the telephone interface) are independent and resampling will have to adapt to any errors. But you mentioned a resampling library before, maybe this takes care of this. The same problem would exist with any audio interface (ALSA, OSS) used by Asterisk. Ciao, -- FA O tu, che porte, correndo si ? E guerra e morte ! _______________________________________________ Linux-audio-dev mailing list [email protected] http://lists.linuxaudio.org/listinfo/linux-audio-dev
