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.
Allow for adding a GstRTSPAuth on the factory and media level and check
permissions when accessing the factory.
Add hints to the auth methods for future more fine grained authorisation.
Add example application for per factory authentication.
Make a 'prepared' signal and emit it when we successfully prepared the element.
This signal can be used to configure the media object after it has been prepared
for streaming.
When we are dynamically adding pads, the addition of the udpsrc elements will
trigger an ASYNC_DONE. We have to ignore this because we only want to react to
the real ASYNC_DONE when everything is prerolled.
Add an eos-shutdown property that will send an EOS to the pipeline before
shutting it down. This allows for nice cleanup in case of a muxer.
Fixes#625597
If we have a new enough multiudpsink with the send-duplicates property, use this
instead of doing our own filtering. Our custom filtering code should eventually
be removed when we can depend on a released -good.
Keep track of how the client connected to the server and setup the udp ports
with the same protocol.
Copy the server ip address in the SDP so that clients can send RTCP back to
us.
When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait
until the media is prerolled or in error. This avoids doing a blocking call of
gst_element_get_state() that can cause lockups when there is an error.
Fixes#611899
Keep track of which transport is active to avoid closing the connection too
soon.
Remove the destination transport also when going to NULL.
Print some stats about the SDES and other RTCP messages we receive from the
clients.
Always perform the state actions even if the target state of the pipeline is
already correct, we still want to add/remove the transports when we are dealing
with shared media.
Keep a counter of the number of active transports for a media so that we can use
this to perform a state change when needed.
Perform a state change of the pipeline only when the first transport was added
or when there are no active transports.
Fix the refcounting of media and sessions in the client. Properly clean up the
session data when the client performs a teardown.
Add Server header to responses.
Allow for multiple uri setups in one session.
Add Range header to the PLAY response and add the range attribute to the SDP
message.
Fix the session pool remove method, it used the wrong key in the hashtable. Also
give the ownership of the sessionid to the session object.
Make the media accept an array of transports for the streams that we have
configured for the play/pause requests.
Implement server states for a client and its media.
Require 0.10.22.1 (git HEAD) of gstreamer.
