3 Jun
2010
3 Jun
'10
1:05 a.m.
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 !