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

russell at digium.com russell at digium.com
Thu Oct 22 09:47:23 CDT 2009

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
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