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
We want to notice ourselves that we're EOS. Otherwise we will
always cancel requests in the very end and confuse the server...
and also make it impossible to use persistent connections.
gstsouputils.c:35:25: error: comparison of constant 9 with expression of type
'SoupLoggerLogLevel' is always false
[-Werror,-Wtautological-constant-out-of-range-compare]
It was used in the past in 0.10 when there was no explicit DTS
field in buffers, now we have it in 1.x series and we can
check it directly with GST_BUFFER_DTS_IS_VALID
Do not try to use subsequent buffer timestamps to calculate
sparse streams durations because the stream is sparse and
the buffers might not be 'time adjacent'. So rely on the
duration and give the option to the pad to provide
custom 'empty' buffers to represent the gaps in the
stream, this can vary on how the data is represented.
Right now, the only sparse stream supported is tx3g subtitles.
