New conference mixing algorithm

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On Jan 14, 2008, at 11:51 AM, Benny Prijono wrote:

> On 1/11/08, Norman Franke <norman at myasd.com> wrote:
> Great! I've temporarily worked around this by ensuring only a  
> single output is active at the start of the call (when we play a  
> recorded message.) Other playbacks are still a problem, but happen   
> much less often.
>
> I've then set the gain of the caller to 250% which makes the  
> recordings sound good.
>
> However, I am anxious for a better fix to the conference mixing  
> code. If you need more info, let me know.
>
> It's here, finally. The SVN trunk now has a brand new conference  
> algorithm ( http://www.pjsip.org/trac/ticket/449). The main feature  
> of the new algorithm is it's no longer uses multiplication to mix  
> signal, but rather it just sums it. Which means there shouldn't be  
> any reduction in audio level (unlike the old one), and also it  
> should be more efficient. But the drawback is when there are too  
> many sources talking at the same time, the audio will be clipped to  
> avoid overflow. There is a simple AGC to smoothen the transition,  
> but nevertheless some audio clicks can't be avoided when signal has  
> to be clipped immediately.
>
> We've spent couple of days testing the new algorithm, and I think  
> it should be good now. But since this is major change, surely there  
> must be some bugs that slip away, so it would be great if you could  
> give it a spin.

Great! One issue immediately, if I set the volume of a playback to  
0.6 all I get it static.


> Also http://www.pjsip.org/trac/ticket/447 may be related to your  
> problem too (you mentioned something about playing WAV file without  
> the LOOP mode).

Not sure. I'm using a simple port callback to play audio, so I know  
I'm feeding in all the data. Even if I pad the last buffer with zeros  
and tell PJ it's a full buffer, I still get that semi-random burst of  
noise.

Norman Franke
Answering Service for Directors, Inc.
www.myasd.com

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