I naively tried, in one xterm:
% sh gst-plugins-good/tests/examples/rtp/server-decodebin-H263p-AMR.sh file:///home/prlw1/optflow.avi and in another xterm: % sh gst-plugins-good/tests/examples/rtp/client-H263p-AMR.sh Plenty of output. On the server side, packets-sent=52, packets-received=52, packets-lost=-52, and the 52 doesn't change. On the client side s/52/0/g. Isn't that meant to "just work"? (gst-play-1.0 ...optflow.avi works fine) Cheers, Patrick _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
On Fri, 2017-10-27 at 15:30 +0100, Patrick Welche wrote:
Hi Patrick, I suspect these scripts may be a bit fragile or bitrotten. Does it work with the VTS server script? If I run the server one on a file I get this in the output: WARNING: from element /GstPipeline:pipeline0/GstURIDecodeBin:decode: Delayed linking failed. Additional debug info: ./grammar.y(510): gst_parse_no_more_pads (): /GstPipeline:pipeline0/GstURIDecodeBin:decode: failed delayed linking some pad of GstURIDecodeBin named decode to some pad of GstVideoConvert named videoconvert0 which isn't a good sign. Would need to investigate what the cause of that is. The simplest way to get going with RTP streaming is usually by using gst-rtsp-server. The source code has a test-uri example, so then you can just do $ gst-rtsp-server/examples/test-uri file:///home/prlw1/optflow.avi and in another terminal or on another machine (with right ip ofc): $ gst-play-1.0 rtsp://127.0.0.1:8554/test Cheers -Tim -- Tim Müller, Centricular Ltd - http://www.centricular.com _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
On Sat, Oct 28, 2017 at 11:46:06AM +0100, Tim M?ller wrote:
> On Fri, 2017-10-27 at 15:30 +0100, Patrick Welche wrote: > I suspect these scripts may be a bit fragile or bitrotten. > > Does it work with the VTS server script? I tried server-VTS-H264-rtx.sh and client-H264-rtx.sh which gave me a test video which drifts to the left (circular shift). > The simplest way to get going with RTP streaming is usually by using > gst-rtsp-server. The source code has a test-uri example, so then you > can just do > > $ gst-rtsp-server/examples/test-uri file:///home/prlw1/optflow.avi > > and in another terminal or on another machine (with right ip ofc): > > $ gst-play-1.0 rtsp://127.0.0.1:8554/test For simplicity, I tried test-video-rtx, which worked. gst-rtsp-server does rather more than the rtpbin stage of the example scripts? Anything in particular? (not just better packaging?) Thanks, Patrick _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
On Tue, 2017-10-31 at 17:06 +0000, Patrick Welche wrote:
> gst-rtsp-server does rather more than the rtpbin stage of the > example scripts? Anything in particular? (not just better packaging?) It has a session connection and can negotiate certain things, like what ports to use, and can transmit setup information that needs to be signalled out of band such as configuration data (some codecs may require that or at least benefit from that, esp. video codecs). Also things like encryption or retransmission support, and it can send the data interleaved via TCP as fallback if needed. It also supports extra features such as pausing and seeking, for example. Cheers -Tim -- Tim Müller, Centricular Ltd - http://www.centricular.com _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
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