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 Linux-audio-dev@email-addr-hidden http://lists.linuxaudio.org/listinfo/linux-audio-devReceived on Thu Jun 3 04:15:01 2010
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