Call gst_rtspsrc_connection_flush (src, FALSE) to reset connections as
non-flushing before sending PAUSE and PLAY with the new npt range. Without this
patch, those commands would fail with EINTR as the connections were still
flushing.
It is placed inside a 'vids' struct, so it was being exposed on
a pad named video_%d. XSUB are subtitles and this patch adds
an special case for it to be exposed in a subpicture_%d pad
A media stream can have multiple payload types. Parse all the payload
types and collect the caps information. We then have to store the
pt<->caps mapping instead of 1 pt and 1 caps.
Parse the profile from the SDP and use that to negotiate the transport
instead of always using AVP.
Rework how we do some tweaks for ASF and Realmedia.
It is possible that the DTS is invalid (when we receive RTP packets from
TCP, for example). As a fallback, use the reconstructed PTS value to
calculate the buffer level.
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
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
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
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
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.
