As per discussion in the bug, remove the drop state from transportreceivebin.
Dropping data is necessary, but for bundled config, needs to happen
further downstream after mixed flows have been separated.
Also support switching back to BLOCK from PASS state.
https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/1206
Using a GCond can easily lead to deadlocks and only duplicates the
waiting code from gstpad.c in the best case.
In this case it actually could lead to a deadlock if both RTP and RTCP
were waiting. Only one of them would be woken up because g_cond_signal()
was used instead of g_cond_broadcast().
Regression introduced by b4bdcf15b7
This commit prevents the handshake from reaching dtlsdec when
the receive state of the receive bin is set to DROP (for example
when transceivers are sendonly).
This preserves the intent of the commit, by blocking the bin
at its sinks until the receive state is no longer BLOCK, but
makes sure the handshake still goes through, by only dropping
data at the src pads, as was the case before.
SDP's are generated and consumed according to the W3C PeerConnection API
available from https://www.w3.org/TR/webrtc/
The SDP is either created initially from the connected
sink pads/attached transceivers as in the case of generating an offer or
intersected with the connected sink pads/attached transceivers as in
the case for creating an answer. In both cases, the rtp payloaded streams
sent by the peer are exposed as separate src pads.
The implementation supports trickle ICE, RTCP muxing, reduced size RTCP.
With contributions from:
Nirbheek Chauhan <nirbheek@centricular.com>
Mathieu Duponchelle <mathieu@centricular.com>
Edward Hervey <edward@centricular.com>
https://bugzilla.gnome.org/show_bug.cgi?id=792523
