If the server URL is not an RTMP(S) URL, the protocol field becomes
required.
The output field becomes required on non-RTMP(S) services to keep
backward compatibility.
Also skip service if the protocol is not available.
Since 65eb3c0815 we tried to get as close
a sync between audio and video tracks as we could before starting to
send frames to the output.
But in be717dbb2c when multi-track audio
was considered for synchronization it continued to try and line ALL
audio packets up with one video packet. While audio and video packets
are similar size this will work out. But once video packets duration is
smaller than 1/2 audio packet duration this may fail forever. In
practice with AAC's 20-23ms frame duration we can often get tracks
producing frames 5-10ms out of phase. For 60fps video this mostly fine
as 16ms frame duration will cover the gap, but for framerates as low as
100 its possible to fail to synchronize. In practice this is ~50%
for 6 audio tracks and 120fps video on my system, or 100% at 240fps.
The play, pause and restart shortcuts were being called as global
shortcuts, being triggered no matter what. This fixes this by calling
the shortcut actions only when the media controls widget has focus.
The Pulseaudio implementation of audio monitoring was actually listing audio sources.
This change corrects that, so all possible outputs are available.
Some send() errors are not treated as fatal but the connection gets shut
down regardless. When this happens, librtmp may send an FCunpublish
message which various services interpret as an "end of stream" message
and disable features like "disconnect protection". Instead, let's
explicitly close the socket so that the remote end is aware that this is
an unclean disconnect.
When creating a group, a newly allocated group `sub_scene` is released
just after calling `obs_scene_add_internal`.
If another thread released the scene-item, which is the sub_scene,
use-after-free might happen.
