<html> <head> <style></style> </head> <body class='hmmessage'> Hi Viraj,<BR>  <BR> I was able to play the the filesrc and aslasink in the same pipeline - the problem was when they were separated by an rtp link. However I have things going now with a customised file src which seems to send things out at a rate which vlc recognises. I don't need to receive it in gst-inspect anymore so I'm not going to persevere with this line of enquiry. But I appreciate your help on this.<BR>  <BR> Thanks,<BR> John<BR> <BR> <HR id=stopSpelling> Date: Mon, 5 Oct 2009 23:50:18 +0530<BR>From: virajk@gmail.com<BR>To: gstreamer-devel@lists.sourceforge.net<BR>Subject: Re: [gst-devel] alsasrc ! alawenc ! rtppcmapay ! udpsink - can receive, record then playback, but not playback live.<BR><BR>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=ecxgmail_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 style="PADDING-LEFT: 1ex" class=ecxgmail_quote> <DIV><BR><BR> <HR> Date: Mon, 5 Oct 2009 19:56:50 +0530<BR>From: <A href="mailto:virajk@gmail.com">virajk@gmail.com</A> <DIV class=ecxim><BR>To: <A href="mailto:gstreamer-devel@lists.sourceforge.net">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=ecxim><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">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">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> </body> </html>