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. 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). cheers, -benny Norman Franke > Answering Service for Directors, Inc. > www.myasd.com > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20080114/d2ca6d59/attachment.html
