Hi All,
I am new to gstreamer.I am using pipeline below mentioned for playing Video and Audio from Live source. gst-launch rtspsrc location=rtsp://192.168.1.168:8554/mpeg4 latency=10 name=demux demux. ! queue! rtpmp4vdepay ! mpeg4videoparse ! dmaidec_mpeg4 ! TIDmaiVideoSink sync=false videoOutput=LCD videoStd=320X240 demux. ! queue ! rtppcmudepay ! mulawdec ! alsasink Above Pipeline is working with proper Audio and Video sync.When I was testing the Video and Audio sync by running the above pipeline so many times,Sometimes Audio got delayed about half second. I am getting this warning mentioned below ../../../../src/gst-libs/gst/audio/gstbaseaudiosink.c(1449): gst_base_audio_sink_render (): /GstPipeline:pipeline0/GstAlsaSink:alsasink0: Unexpected discontinuity in audio timestamps of more than half a second (0:00:00.569375000), resyncing thanks and regards, Arasu.M ------------------------------------------------------------------------------ This SF.net email is sponsored by Make an app they can't live without Enter the BlackBerry Developer Challenge http://p.sf.net/sfu/RIM-dev2dev _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
Hi ,
Please use gst-launch rtspsrc location=rtsp://192.168.1.168:8554/mpeg4 latency=10 name=demux demux. ! queue! rtpmp4vdepay ! mpeg4videoparse ! dmaidec_mpeg4 ! TIDmaiVideoSink sync=true videoOutput=LCD videoStd=320X240 demux. ! queue ! rtppcmudepay ! mulawdec ! alsasink On Thu, Aug 12, 2010 at 7:55 PM, tamil arasu <[hidden email]> wrote: Hi All, -- Thanx & Regards Ajay Gautam +91-9717785580 ------------------------------------------------------------------------------ This SF.net email is sponsored by Make an app they can't live without Enter the BlackBerry Developer Challenge http://p.sf.net/sfu/RIM-dev2dev _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
Hi Ajay Gautam,
gst-launch rtspsrc location=rtsp://192.168.1.168:8554/mpeg4 latency=10 name=demux demux. ! queue! rtpmp4vdepay ! mpeg4videoparse ! dmaidec_mpeg4 ! TIDmaiVideoSink sync=true videoOutput=LCD videoStd=320X240 demux. ! queue ! rtppcmudepay ! mulawdec ! alsasink Above one is working with proper Video and Audio sync.But there was 3 to 4 second delay between live source and Destination end(where I am displaying video and audio). I increased latency to 500, I am getting the same delay. thanks and regards, Arasu.M
On Thu, Aug 12, 2010 at 9:00 PM, AJAY GAUTAM <[hidden email]> wrote: Hi , ------------------------------------------------------------------------------ This SF.net email is sponsored by Make an app they can't live without Enter the BlackBerry Developer Challenge http://p.sf.net/sfu/RIM-dev2dev _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
On Fri, Aug 13, 2010 at 11:10 AM, tamil arasu <[hidden email]> wrote: Hi Ajay Gautam, Hi , Please update your package to Revision 1.45 ,It will not sync properly if the first audio and video packets do not arrive in the receiver at the same time. -- Thanx & Regards Ajay Gautam +91-9717785580 ------------------------------------------------------------------------------ This SF.net email is sponsored by Make an app they can't live without Enter the BlackBerry Developer Challenge http://p.sf.net/sfu/RIM-dev2dev _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
Hi All,
Live Source =================> Capturing Module. (DM365ipnc) (DM365 leopard) My Live Source is DM365IPNC.I am capturing Video and audio coming from IPNC in Capturing Module Unit using other Board. There is no problem in Sync between Audio and Video in Capturing Module.Now Two or Three second delay between Live Source Module and Capturing Module. How to solve this delay problem between these two module. thanks and regards, Arasu.M On Fri, Aug 13, 2010 at 12:39 PM, AJAY GAUTAM <[hidden email]> wrote:
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