[asterisk-dev] Choppy Audio with JACK (app_jack)

[russell at digium.com]

It looks to be as simple as needing to increase the buffer size in  
between Asterisk and JACK.  I never tested with a sample rate anywhere  
close to 96 kHz.  I'm not at a computer so I can't point to the exact  
code right now.

I wonder if I made the buffers a static size.  We should have them as  
a dynamic size based on the sample rate in use.  I am traveling this  
weekend, but I will take a look when I get to a computer.

--
Russell Bryant
(Sent from mobile device)

On Oct 22, 2009, at 8:44 AM, Esben Stien <b0ef at esben-stien.name> wrote:

> I'm playing with the app_jack module and experience very noisy choppy
> audio.
>
> It seems to setup the JACK channels fine, but the audio has lots of
> dropouts.
>
> In the terminal, I get repeatedly this message:
>
> [Oct 22 16:07:06] ERROR[29065]: app_jack.c:559 queue_voice_frame:  
> Output
> buffer filled ... need to increase its size
>
> ..while the conversation is active.
>
> The sample rate conversion seems fine. I inserted a log call after the
> resample_factor was set:
>
>    *resample_factor = to_srate / from_srate;
>    ast_log(LOG_ERROR,"factor=%f \n",*resample_factor);
>
> ..which outputs:
>
> [Oct 22 16:07:06] ERROR[29065]: app_jack.c:221 alloc_resampler:  
> factor=12.000000
> [Oct 22 16:07:06] ERROR[29068]: app_jack.c:221 alloc_resampler:  
> factor=0.083333
>
> I run JACK at 96kHz, which factor here confirms to have found.  
> 8kHz*12=96kHz.
>
> I'm running on GNU/Linux with asterisk-SVN from today. I'm running
> jack-1.9.4 (SVN).
>
> I have quite an extensive setup of JACK apps, so I know that JACK  
> itself
> is not a problem.
>
> Any pointers as to what I can try?.
>
> -- 
> Esben Stien is b0ef at e     s      a
>         http://www. s     t    n m
>          irc://irc.  b  -  i  .   e/%23contact
>           sip:b0ef@   e     e
>           jid:b0ef@    n     n
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-dev mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-dev