<html><head><meta http-equiv="Content-Type" content="text/html; charset=UTF-8"></head><body><div>Why so complicated? If you have only video source, why not source ! queue ! tcpserversink.</div><div>The rest could be done on client side.</div><div><br></div><div><br></div><div><br></div><div id="composer_signature"><div style="font-size:85%;color:#575757" dir="auto">Von meinem Samsung Galaxy Smartphone gesendet.</div></div><div><br></div><div style="font-size:100%;color:#000000"><div>-------- Ursprüngliche Nachricht --------</div><div>Von: Andres Gonzalez <andres.agoralabs@gmail.com> </div><div>Datum: 04.08.16 20:24 (GMT+01:00) </div><div>An: gstreamer-devel@lists.freedesktop.org </div><div>Betreff: Re: AW: the humble videotestsrc element is magical !!! </div><div><br></div></div>Thank you Bernhard for your response.<br><br>This pipeline appears to work correctly on the server but causes the client<br>to exit in error:<br><br>gst-launch-1.0 -vv decklinkvideosrc device-number=4 mode=10 ! queue !<br>videoconvert ! x264enc byte-stream=true !<br>"video/x-h264,stream-format=byte-stream" ! h264parse config-interval=1 !<br>queue ! matroskamux streamable=true ! queue leaky=2 ! tcpserversink<br>port=50010 host=0.0.0.0 recover-policy=keyframe sync-method=latest-keyframe<br>sync=false<br><br>This pipeline also appears to work correctly on the server but causes the<br>client to exit in error:<br><br>gst-launch-1.0 -vv v4l2src device=/dev/video0 typefind=true ! queue !<br>videoconvert ! x264enc byte-stream=true ! h264parse config-interval=1 !<br>queue ! matroskamux streamable=true ! queue leaky=2 ! tcpserversink<br>port=50010 host=0.0.0.0 recover-policy=keyframe sync-method=latest-keyframe<br>sync=false<br><br>Actually, these are just temporary sources for me. I am working on a native<br>app that uses the API and not gst-launch. I have several GStreamer RTP/RTCP<br>streams working correctly using appsrc for the sources, and I am currently<br>in the process of adding TCP streaming for these appsrc sources. But I could<br>not get my appsrc-based pipelines to work correctly when I used those same<br>basic appsrc-based pipelines (but modified the sink elements for<br>tcpserversink instead of the RTP/RTCP sinks). <br><br>So I concluded that TCP streaming and tcpserversink was more complicated<br>than I originally thought, or at least I am missing some key concepts. So I<br>went back to playing with these gst-launch pipelines so I could figure out<br>what I was doing wrong. Debugging GStreamer gst-launch scripts is much<br>easier than debugging GStreamer API routines in my native C++ app. :-)<br><br>Thanks for your help,<br><br>-Andres<br><br><br><br>--<br>View this message in context: http://gstreamer-devel.966125.n4.nabble.com/the-humble-videotestsrc-element-is-magical-tp4678965p4678969.html<br>Sent from the GStreamer-devel mailing list archive at Nabble.com.<br>_______________________________________________<br>gstreamer-devel mailing list<br>gstreamer-devel@lists.freedesktop.org<br>https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel<br></body></html>