<html><head><style type='text/css'>p { margin: 0; }</style></head><body><div style='font-family: Times New Roman; font-size: 12pt; color: #000000'>I have my burst rate set to something low, 4096 right now. I also wrote a flash/flex app that has the same size buffer which results in the audio being heard immediately when connecting. As far as the audio being real time, the audio stream is about 6 seconds behind which I'm guessing is the result of the size of the lame buffers in the mod_shout modules (i'm using g.711 ulaw), I was going to look into that next week. Anyone have any thoughts about where else the delay may be happening? I hoping to get this down to around 2 seconds.<br><br>D-<br><br>----- Original Message -----<br>From: "Rupa Schomaker" <rupa@rupa.com><br>To: freeswitch-users@lists.freeswitch.org<br>Sent: Thursday, April 2, 2009 8:05:38 AM GMT -07:00 US/Canada Mountain<br>Subject: Re: [Freeswitch-users] mod_shout delay in trunk<br><br><br><br><div class="gmail_quote">2009/4/2 Anthony Minessale <span dir="ltr"><<a href="mailto:anthony.minessale@gmail.com" target="_blank">anthony.minessale@gmail.com</a>></span><br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Its the buffering and startup of the shout stream taking up the time,<br><br>HINT put the shoutcast stream into a local stream with a .loc file and then play that in the conference.<br></blockquote></div><br>Ah, that is easy enough! Though I think with icecast doing the burst_on_connect thingie there should be enough data (pushed much faster than real time) to fill FS's buffers. But that would require mod_shout to cooperate with that strategy. <br>
<br>ie: on connect, drain the socket as fast as it can filling it's own buffers. Once it's own buffers are full start streaming. I think right now it drains the socket only as fast as it needs to. Or maybe not. <shrug/><br>
<br>I'll go the local stream route for now....<br clear="all"><br>-- <br>-Rupa<br>
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