Fixes ffeb09e4ab
if (sscanf(...)) { // != 0
error;
}
Is not correct where != 0 indicates some kind of success.
Check instead that the correct number of elements were slurped.
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
By removing the indirection to the main loop completely when receiving
the peer certificate. For reference, the on-decoder-key signal does not
have a redirection.
We call the base class first as this will remove the pad from
the aggregator, thus stopping misc callbacks from being called,
one of which (process_textures) will recreate the vertex_buffer
if it is destroyed
https://bugzilla.gnome.org/show_bug.cgi?id=760873
For libsrtp 1, add defines that translate the new namespaced identifiers
to the old unnamespaced ones. Also move the code for setting and getting
a stream's ROC into two compat functions that match libsrtp2's API.
It seems that libsrtp2 properly supports changing the ROC without having
to touch the sequence numbers afterwards, given that srtp_set_stream_roc
sets a pending_roc field, so the entire roc_changed dance should not be
needed anymore. The compat functions for libsrtp 1 just contain our
preexisting hacks, however, so it's still needed there.
libsrtp2 has no means of discovering the streams in the session, so to
create the stats structure we need to iterate over our own set of SSRCs.
For this we also need to re-add the previously removed ssrcs_set to the
encoder.
https://bugzilla.gnome.org/show_bug.cgi?id=776901
Fix regression when used in combination with new flvmux which was
ported to GstAggregator, and which sends plain video/x-flv caps
before sending full caps that include streamheaders.
Instead of a massive if/else/if/else/if/else/...:
* Use a common cleanup path for allocated items just before leaving
the function (which will be free-d only if we're not dealing with
a delayed SPU).
* "goto" that cleanup path wherever needed
CID #1427096
CID #1427114
In file included from ../../../gst-plugins-bad/ext/gl/gstopengl.c:47:0:
../../../gst-plugins-bad/ext/gl/gstglmixerbin.h:25:29: fatal error: gst/video/video.h: No such file or directory
