>To: [email protected] >From: [EMAIL PROTECTED] (Tony Mountifield) >Date: Mon, 14 Feb 2005 17:58:07 +0000 (UTC) >Subject: [Asterisk-Dev] Re: Audio delay in MeetMe using SIP when not 'q' mode > >In article <[EMAIL PROTECTED]>, >Tony Mountifield <[EMAIL PROTECTED]> wrote: >> In article <[EMAIL PROTECTED]>, >> Kevin P. Fleming <[EMAIL PROTECTED]> wrote: >> > Tony Mountifield wrote: >> > >> > > I am suspecting that the problem is something to do with the conf_play() >> > > of the enter and leave sounds. My guess is that by writing that raw data >> > > into the pseudo device fd, it causes a backlog that never drains, because >> > > the device is only getting emptied at the same rate as the conference is >> > > filling it. >> > >> > That's very interesting... it certainly seems possible that it could be >> > the case. It would be pretty simple to test, just replace the "conf >> > enter" sound with something quite a bit longer (4 or 5 seconds) and >> > determine if the delay increases accordingly. >> >> Well I've tried the next best thing: commented out the careful_write that >> is in conf_play(). The delay then went away, even on non-q confs. > >I've just had an idea how to fix it. > >a) Have conf_play() return the length of the data it wrote. > >b) Have a variable 'skiplen', and write the ENTER sound as: > > skiplen += conf_play(conf, ENTER); > >c) In the main loop, when processing the voice frames, do this: [code removed] >This would then throw away the voice bytes that would have occupied the >space now taken by the Enter sound. > >I haven't tested it yet. Will do shortly. > >Cheers >Tony >--
It would be better to overlay/merge the audio rather than just skip it. Someone could be saying something important at that moment... The problem has been around for a while. Open a bug report on it and post your patch there. That way there is a better record of the updates. Andrew _______________________________________________ Asterisk-Dev mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
