"do-retransmission" was previously set when rtx-time != 0,
which made no sense as do-retransmission is used to enable
the sending of retransmission requests, where as rtx-time
is used by the peer to enable storing of buffers in order
to respond to retransmission requests.
rtsp-media now also provides a callback for the
request-aux-receiver signal.
https://bugzilla.gnome.org/show_bug.cgi?id=794822
For NTP and PTP clocks we signal the actual clock that is used and signal
the direct media clock offset.
For all other clocks we at least signal that it's the local sender clock.
This allows receivers to know which clock was used to generate the media and
its RTP timestamps. Receivers can then implement network synchronization,
either absolute or at least relative by getting the sender clock rate directly
via NTP/PTP instead of estimating it from RTP timestamps and packet receive
times.
https://bugzilla.gnome.org/show_bug.cgi?id=760005
Without TEARDOWN it might be desireable to keep the media running and continue
sending data to the client, even if the RTSP connection itself is
disconnected.
Only do this for session medias that have only UDP transports. If there's at
least on TCP transport, it will stop working and cause problems when the
connection is disconnected.
https://bugzilla.gnome.org/show_bug.cgi?id=758999
Allow specifying the GType of a GstRtspMedia subclass to create
as a simpler way to get the factory to create a custom
GstRtspMedia sub-class, without subclassing GstRtspMediaFactory.
Add support for different suspend modes. The stream is suspended right after
producing the SDP and after PAUSE. Different suspend modes are available that
affect the state of the pipeline. NONE leaves the pipeline state unchanged and
is the current and old behaviour, PAUSE will set the pipeline to the PAUSED
state and RESET will bring the pipeline to the NULL state.
A stream is also unsuspended when it goes back to PLAYING, for RESET streams,
this means that the pipeline needs to be prerolled again.
Base on patches by Ognyan Tonchev <ognyan@axis.com>
See https://bugzilla.gnome.org/show_bug.cgi?id=711257
Remove the auth object from media and factory. We want to have the RTSPClient
authenticate and authorize resources, there is no need to place another auth
manager on the media/factory.
Make GObjects from the remaining simple structures.
Remove GstRTSPSessionStream, it's not needed.
Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter
Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how
a GstRTSPStream should be transported to a client.
Rename GstRTSPMediaFactory::get_element -> create_element because that
more accurately describes what it does.
Make nice methods instead of poking in the structures.
Move some methods inside the relevant object source code.
Use GPtrArray to store objects instead of plain arrays, it is more
natural and allows us to more easily clean up.
Move the allocation of udp ports to the Stream object. The Stream object
contains the elements needed to stream the media to a client.
Improve the prepare and unprepare methods. Unprepare should now undo
everything prepare did. Improve also async unprepare when doing EOS on
shutdown. Make sure we always unprepare correctly.
