gstrtpjitterbuffer.c: In function 'gst_rtp_jitter_buffer_loop':
gstrtpjitterbuffer.c:2978:3: error: 'result' may be used uninitialized in this function
while (result == GST_FLOW_OK);
^
Several conditional statements perform comparison on RTP sequence
numbers without taking the sequence number rollover into account.
Instead, use the gst_rtp_buffer_compare_seqnum function to perform the
comparison.
https://bugzilla.gnome.org/show_bug.cgi?id=725159
For each videoCdevice probe it input/output capabilities
if it match with video decoder requirement register a new element.
Signed-off-by: Benjamin Gaignard <benjamin.gaignard@linaro.org>
https://bugzilla.gnome.org/show_bug.cgi?id=722128
This fixes an issue with gst-rtsp-server where no sps and pps are
sent for the first intra frame, because the payloader starts working
already when receiving DESCRIBE but there is no transports so it tries
to send sps and pps, but that fails with a FLUSHING flow. But the time
for last sent sps and pps would still be set, so when PLAY arrives and
the first intra frame is to be sent there is no sps and pps sent due to
that time since last sps pps is less than spspps_interval.
https://bugzilla.gnome.org/show_bug.cgi?id=724213
In the case where we have no suitable candidate we post a warning and plug a
fake-element. Do the same when non of the candidate work.
This is more consistent and plugin the fakesink as a fallback is probably
helpful for running unit tests without requiring hardware src/sink elements.
Fixes#722981
Add log handlers for jack that write to the gst debug log. This avoids spamming
the console when e.g. using autoaudiosink, having the jack elements installed,
but not running jack.
We correctly indicate the field ordering on interlaced buffers, but fail to
flag them as containing interlaced video, which we need to do here because
we signal interlace-mode=mixed in our caps. This means that downstream
elements (like vaapipostproc from gstreamer-vaapi) don't recognise these
buffers as in need of deinterlacing.
Fix this by setting the interlaced flag on all interlaced buffers.
Signed-off-by: Simon Farnsworth <simon.farnsworth@onelan.co.uk>
https://bugzilla.gnome.org/show_bug.cgi?id=724899
Adds two extra checks:
- Sampling frequency on header can't be 15.
- Frame size should be at least 9 or 7, depending
on whether CRC protection is present.
https://bugzilla.gnome.org/show_bug.cgi?id=724638
Add support for a new property: tls-database. If the property is set,
the certificate database will be given to the rtsp connection if TLS
protocol is being used. If the server certificate can't be verified with
the default database, this additional database will be used.
https://bugzilla.gnome.org/show_bug.cgi?id=724396
Add fake audio/video sinks. Previously running the test might be flaky due to
the use of real elements (hardware in use), which we don't want to test here.
Add two more tests that check that the fakes are chosen.
Commit 46fd12ae5e introduced connection
recovery. But when server does not specify content-size,
souphttpsrc tries to reconnect even after regular end of stream.
Http server replies with SOUP_STATUS_REQUESTED_RANGE_NOT_SATISFIABLE
but souphttpsrc still emits error instead of EOS.
https://bugzilla.gnome.org/show_bug.cgi?id=724717
Signed-off-by: Branislav Katreniak <bkatreniak@nuvotechnologies.com>
It turns out that the change in gtk-doc-1.20 which wraps the |[]| content in
CDATA break xi:inlcude examples. As in a whole jhbuild checkout these where
the only 4, we're fixing them instead.
recv_rtp_sink: allow proxying of the allocation query.
send_rtp_sink: allow proxying of caps and allocation. This allows us to
query caps downstream as well as get an allocator from downstream.
send_rtp_src: allow proxy of caps, this makes the caps query do
upstream.
See https://bugzilla.gnome.org/show_bug.cgi?id=723850
