John,<br>When you play from filesrc, the alsasink will make sure that the pipeline runs realtime.<br>So actually no hacking is required. Did you face any issues without hacking filesrc?<br><br>Reason for sporadic bursts could be loss of packets while pipeline is busy in playing out.<br> Try putting a queue before alsasink. That will decouple the udp receive and alsasink playout and should avoid packets loss. But I could work without any queue.<br><br>- Viraj<br><br><div class="gmail_quote">On Mon, Oct 5, 2009 at 8:38 PM, john oliver <span dir="ltr"><<a href="mailto:johnwpoliver@hotmail.com">johnwpoliver@hotmail.com</a>></span> wrote:<br> <blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"> <div> <br><br><hr>Date: Mon, 5 Oct 2009 19:56:50 +0530<br>From: <a href="mailto:virajk@gmail.com" target="_blank">virajk@gmail.com</a><div class="im"><br>To: <a href="mailto:gstreamer-devel@lists.sourceforge.net" target="_blank">gstreamer-devel@lists.sourceforge.net</a><br> </div>Subject: Re: [gst-devel] alsasrc ! alawenc ! rtppcmapay ! udpsink - can receive, record then playback, but not playback live.<div class="im"><br><br><div>Hi,</div> <div>I dont think there is any bug in gstreamer.</div> <div>I have successfully run "audio capture - UDP send" on one board and "UDP receive - audio playout" on other board. </div> <div>But we didnt specify ssrc, clock-base, seqnum-base etc.. parameters for udpsrc.</div> <div>Try this once (your command with minimal udpsrc parameters)</div> <div> </div> <div>gst-launch-0.10 -v udpsrc port=1234 caps="application/x-rtp, media=(string)audio, clock-rate=(int)8000, encoding-name=(string)PCMA" ! rtppcmadepay ! audio/x-alaw, rate=8000, channels=1 ! alawdec ! alsasink </div> <br><br></div>Thanks Viraj, however I am still only hearing sporadic bursts with that pipeline at the client end. I should have also mentioned that when I successfully streamed to vlc, on the server end I was time-stamping from a filesrc, but I had hacked gstfilesrc.c to sleep between frames for the sample length equivelent to my frame size.<br> <br><br>John<br> </div> <br>------------------------------------------------------------------------------<br> Come build with us! The BlackBerry® Developer Conference in SF, CA<br> is the only developer event you need to attend this year. Jumpstart your<br> developing skills, take BlackBerry mobile applications to market and stay<br> ahead of the curve. Join us from November 9-12, 2009. Register now!<br> <a href="http://p.sf.net/sfu/devconf" target="_blank">http://p.sf.net/sfu/devconf</a><br>_______________________________________________<br> gstreamer-devel mailing list<br> <a href="mailto:gstreamer-devel@lists.sourceforge.net">gstreamer-devel@lists.sourceforge.net</a><br> <a href="https://lists.sourceforge.net/lists/listinfo/gstreamer-devel" target="_blank">https://lists.sourceforge.net/lists/listinfo/gstreamer-devel</a><br> <br></blockquote></div><br><br clear="all"><br>-- <br>- Viraj<br>Reality is merely an illusion, albeit a very persistent one.<br>