gstrtspsrc question

[Federico Zamperini]

Try with something like:

gst-launch-0.10 playbin2 uri=rtsp://your_url_here:port/path

with enough debug trace (--gst-debug-level=3 or 4?) to see what playbin2 
does (beware: I read in the changelog that playbin2 dropped the "2" and 
became playbin in gstreamer-1.0).
Or you can see a graph of the same pipeline setting 
GST_DEBUG_DUMP_DOT_DIR and then looking at the dot files (see 
http://gstreamer.freedesktop.org/wiki/DumpingPipelineGraphs for details).

On this site (http://hughesbennett.co.uk/Graphviz) you can even paste 
the content of your dot files and see the graphs in the browser without 
installing anything.

Strictly talking about your question have a look at gstrtpbin:

http://www.freedesktop.org/software/gstreamer-sdk/data/docs/latest/gst-plugins-good-plugins-0.10/gst-plugins-good-plugins-gstrtpbin.html

and at the examples here:

http://cgit.freedesktop.org/gstreamer/gst-plugins-good/tree/tests/examples/rtp


Federico


Il 11/02/2013 21:22, Chuck Crisler ha scritto:
> I am trying to add support for rtsp input into my app using gstreamer
> 0.10.30 with good-plugins-0.10.25. The initial test file is an MP4 that
> has both audio and video. It gets pretty far in the negotiation and then
> fails with 'not linked'. Since this has both audio and video, I need to
> specify decoders and display devices (or NULL sinks) for both streams,
> somehow using gstrtpssrcdemux (I think). Is there an example somewhere
> of how to do this or would someone sketch out what I need to do? I am
> currently working with gst-launch but will implement the app in C++ when
> I get the test working.
>
> Thank you very much,
> Chuck Crisler
>
>
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