[Bug 740683] rtspsrc: add retransmission handling for rtp

[GStreamer (bugzilla.gnome.org)]

https://bugzilla.gnome.org/show_bug.cgi?id=740683
  GStreamer | gst-plugins-good | git

Olivier Crête <olivier.crete> changed:

           What    |Removed                     |Added
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                 CC|                            |olivier.crete at ocrete.ca

--- Comment #4 from Olivier Crête <olivier.crete at ocrete.ca> 2014-11-29 22:07:28 UTC ---
(In reply to comment #3)
> (In reply to comment #2)
> > ::: gst/rtsp/gstrtspsrc.c
> > @@ +198,3 @@
> >  #define DEFAULT_TLS_VALIDATION_FLAGS     G_TLS_CERTIFICATE_VALIDATE_ALL
> >  #define DEFAULT_TLS_DATABASE     NULL
> > +#define DEFAULT_DO_RETRANSMISSION        FALSE
> > 
> > Any reason to default to false?
> 
> Because we override the request-aux-receiver signal and the application might
> want to do its own thing with that signal.

Maybe we could use g_signal_connect_after(), so it will use any external
override first and the internal one only if no external one is connected?

Making is the detault would be a big advantage for all of the playbin users.

> There's also rdt in -ugly that might be used which doesn't support the
> request-aux-receiver signal.

But that would trigger the "transport->trans != GST_RTSP_TRANS_RTP" on the
previous line.

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