On Sat, Oct 17, 2009 at 2:26 PM, Richard Kenner <[email protected]> wrote: > Is this patch correct? The "&&" doesn't make logical sense to me. I think > it should be "||" and making this change fixes the problem I have with SIP > phones in MeetMe conferences. If it's correct, is there someplace more > formal that I should submit it to? > > *** app_meetme.c.old 2009-10-11 17:56:44.000000000 -0400 > --- app_meetme.c 2009-10-17 14:22:29.000000000 -0400 > *************** > *** 2901,2905 **** > to write out all the > samples. > */ > ! if (user->talking && > !(confflags & CONFFLAG_OPTIMIZETALKER)) { > careful_write(fd, > f->data.ptr, f->datalen, 0); > } > --- 2901,2905 ---- > to write out all the > samples. > */ > ! if (user->talking || > !(confflags & CONFFLAG_OPTIMIZETALKER)) { > careful_write(fd, > f->data.ptr, f->datalen, 0); > }
I have a limited acquaintance with the app_meetme.c code, having also modified it for my own purposes in the past. For some historical perspective, you should know that originally in 1.6, talker optimization was forced on, with no way to turn it off. Many people reported that it sounded awful, and patches were released to make optimization be off by default, with a flag to turn it on, the way it was in 1.4 There are more details, especially as regards versions were various patches from trunk were merged into the releases, but that's the very quick version. If you want bug numbers, please check out http://issues.asterisk.org and search for talker optimization and meetme. When I first tried to use MeetMe with 1.6, I had a lot of voice-cut-out problems, which I was able to verify were caused by over-aggressive talker optimization, resulting in me going to 1.6.1.* series as that series at that time had the 'optimization off by default' feature that I wanted. >From a quick glance at your patch, I would say that it probably tries to address the audio quality problems I and others were experiencing. It's possible, and probably likely that somebody else already fixed this. Please be sure to check through the issues, and make sure you compare your work against app_meetme.c in trunk, as your work will NOT make it into asterisk by posting it to asterisk-users. If you didn't already know, you should know that SVN is often way ahead of the 1.6.0. series for many reasons. For the latest versions for each 1.6 series, you should check through: http://svnview.digium.com/svn/asterisk/branches/ _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
