change_ssrc field of RTPSession should be set before calling
rtp_session_schedule_bye_locked () as this function will call reconsider function
that will wake up rtcp_thread which will call rtp_session_on_timeout () that will
check change_ssrc to change the ssrc.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=694184
Depending on the order g_object_set() calls aare made, the
target r/g/b settings will override the method if set to
green/blue. Change that so we do not use the target-r/g/b values
unless the method is set to custom.
https://bugzilla.gnome.org/show_bug.cgi?id=694374
Only delay the RTCP thread when we are a sender, which we can know because we
have a send_rtp_src pad. Otherwise we might delay the RTCP thread if we
are only a receiver and then there is no code path that wakes up the
RTCP thread and we end up without RTCP packets.
v4l has add a new IOCTL to export a buffer by using dmabuf.
This patch allow to use this new IOTCL if it has been defined in videodev2.h
I introduce a new IO mode (GST_V4L2_IO_DMABUF) to enable this way of working.
https://bugzilla.gnome.org/show_bug.cgi?id=693826
Don't unmap short MOOV atom buffer twice, which happened
in the case where we don't fix up the MOOV atom.
Fixes crashes when thumbnailing partial mp4 file where
the MOOV atom is still incomplete.
https://bugzilla.gnome.org/show_bug.cgi?id=694010
Apparently there's no reason to use it any longer. Drop libsoup-gnome
dependency while at it, now that we don't need anything from it any
more (it only consists entirely of deprecated API now anyways).
https://bugzilla.gnome.org/show_bug.cgi?id=693911
Now that the subset check actually works, this breaks
things with demuxers that don't put a "sof-marker"
in their jpeg caps, and we don't have a good parser
to plug either yet.
We're testing with an http server on localhost, but don't support
an exception list for the http_proxy, so just unset the environment
variable to make sure we can run this test properly even if the
environment has http_proxy set.
Also, don't skip all tests if there is an issue with the SSL server,
just run the non-SSL tests then.
https://jenkins.qa.ubuntu.com/view/Raring/view/JHBuild%20Gnome/job/jhbuild-amd64-gst-plugins-good/
Delay sending the first RTCP packet until we have sent the first RTP packet.
Otherwise we will send out a Receiver Report instead of a sender report.
See https://bugzilla.gnome.org/show_bug.cgi?id=691400
Move the work of cleaning up the client streams in the free_stream
function. This allows us to properly clean up the client streams when we
remove an RTP stream as well.
Based on patch by Sujay <sdatar@cisco.com>
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=660156
Only run the skew estimation code when we have a new RTP timestamp. If we have
the same RTP timestamp, we simply use the previous estimation. This works
because the new observation with the same RTP timestamp has to have a bigger
receiver time and is thus not going to influence the estimation except for
causing more jitter.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=640023
