This causes rtspsrc to send a teardown and wait on
PAUSED->READY transition, with a configurable delay.
Otherwise, typically teardown never gets sent in
playbin / uridecodebin where the transition back to NULL
happens too quickly.
The timeout is set to 100ms default.
https://bugzilla.gnome.org/show_bug.cgi?id=751994
Just remove the code. It's not doing anything useful anyways. The modified
caps are the result of a caps query, so either not used afterwards of a
reference to some internal caps of another element that should not be
modified.
https://bugzilla.gnome.org/show_bug.cgi?id=796837
When it is trivial to pass-through a timecode, by only removing the
"interlaced" flag, do pass-through. Otherwise, double the fps_n and
adjust the "frames" field.
https://bugzilla.gnome.org/show_bug.cgi?id=796818
When handling input with timestamps that repeat, sometimes
splitmuxsink would get confused and ignore a keyframe.
The logic in question is a holdover from before the cmd queue
moved the file cutting to the multiqueue output side and made
it deterministic, so it's no longer needed on the input
here.
https://bugzilla.gnome.org/show_bug.cgi?id=796773
This reverts commit 3ac5430311.
There's no need to make a freshly created event writable,
and the other half of this patch was already fixed
and pushed in f2f15a1
Always wait with starting the RTCP thread until either a RTP or RTCP
packet is sent or received. Special handling is needed to make sure the
RTCP thread is started when requesting an early RTCP packet.
We want to wait with starting the RTCP thread until it's needed in order
to not send RTCP packets for an inactive source.
https://bugzilla.gnome.org/show_bug.cgi?id=795139
* When receiving a segment in TIME, use that seqnum
* Only reset the stored sequence number when doing HARD reset
(and not when we get a FLUSH event from upstream)
This patch aims at fixing the recent regressions in the adaptive test
suite.
All segment pushing in push mode is now done with
gst_qtdemux_check_send_pending_segment(), which is idempotent and
handles both edit lists cases and cases where the upstream TIME segments
have to be sent directly.
Fragmented files that start with a non-zero tfdt are also taken into
account, but their handling has been vastly simplified: now they are
handled as implicit default seeks so there is no need to extend the
GstSegment formulas as was being done before.
qtdemux->segment.duration is no longer modified when
upstream_format_is_time, respecting in this way the durations provided
by dashdemux and fixing bugs in reverse playback tests where mangled
durations appeared in the emitted segments.
https://bugzilla.gnome.org/show_bug.cgi?id=752603
Upstream driving elements such as dashdemux often do reverse playback by
feeding qtdemux with the fragments containing the requested playback
range in reverse order.
But the requested playback range stop may be somewhere in the
middle of a fragment. In that case, a naive pts >= segment.stop
condition may declare end of segment prematurely when demuxing this
first fragment.
This used not to happen because there were places in moov parsing where
segment.stop was overwritten to GST_CLOCK_TIME_NONE even if
upstream_format_is_time -- resulting in this case in a segment with rate
< 0 and stop == -1 and hence not triggering the EOS check, but that was
likely an accident.
This patch modifies the EOS check to take this case into account, not
sending EOS when upstream_format_is_time if rate < 0.
This fixes adaptive.dash.playback.seek_end_live.DASHIF_livestream_testpic_2s
https://bugzilla.gnome.org/show_bug.cgi?id=752603
Sample table based segment event (genereted by qtdemux) could break
presentation timeline. For example, qtdemux should not modify upstream
time format segment (e.g., adaptivedemux use case)
https://bugzilla.gnome.org/show_bug.cgi?id=796480
This field is actually only informatory and the user can potentially
choose something else. EME tests in WebKit testsuite actually doesn't
take it into and force another encryption system to be used, and expects
to be given the occasion to do so.
This basically also reverts 3e063703b3.
Instead of always keeping a safe segment (start=0) event from the beginning,
delay the creation of this event to when we really know the timestamp of the
first sample. This is important to properly start fragmented streams that
we might join in the middle or to play isolated fragment files that might
have an advanced tfdt.
https://bugzilla.gnome.org/show_bug.cgi?id=752603
Fragmented files often use elst.duration=0 which before
ee78825eae was wrongly interpreted as
having no frames.
Since that issue has now been fixed, there is no reason to disable edit
lists in fragmented files. This commit enables them, therefore producing
correct stream time for files containing edit lists.
https://bugzilla.gnome.org/show_bug.cgi?id=793058
Since ca068865c3 the duration of the first
frame is not used for estimating the frame rate.
For this purpose, stream->first_duration was initialized with the
duration of the first frame. In fragmented files, this was previously
done by peeking the first moof, but that can only be done in pull mode.
Fortunately, we don't really need to do that, at least with the current
design: When we are estimating the frame rate we already have the
sample table, regardless of the scheduling mode and whether the file is
fragmented or not, so we can obtain first_duration there much more
reliably.
This fixes frame rate estimation for fragmented files in push mode.
https://bugzilla.gnome.org/show_bug.cgi?id=796384
This mode is useful for muxers that can take a long time to finalize a
file. Instead of blocking the whole upstream pipeline while the muxer is
doing its stuff, we can unlink it and spawn a new muxer+sink combination
to continue running normally.
This requires us to receive the muxer and sink (if needed) as factories,
optionally accompanied by their respective properties structures. Also
added the muxer-added and sink-added signals, in case custom code has to
be called for them.
https://bugzilla.gnome.org/show_bug.cgi?id=783754
This is a straightforward translation of 5dd39d8, can be trivially
checked by running:
gst-launch-1.0 -v videotestsrc ! video/x-raw, colorimetry=2:4:7:1 ! \
matroskamux ! matroskademux ! fakesink
and verifying that the colorimetry is correctly preserved.
https://bugzilla.gnome.org/show_bug.cgi?id=796344
The code before copied GstStructure twice. The first time inside
gst_value_set_structure and the second time in g_value_array_append.
Optimized version does no copies, just transfers ownership to
GValueArray. It takes advantage of the fact that array has already
enough elements preallocated and the memory is zero initialized.
https://bugzilla.gnome.org/show_bug.cgi?id=795139
If obtain_internal_source() returns a source that is not internal it
means there exists a non-internal source with the same ssrc. Such an
ssrc collision should be handled by sending a GstRTPCollision event
upstream and choose a new ssrc, but for now we simply drop the packet.
Trying to process the packet further will cause it to be pushed
usptream (!) since the source is not internal (see source_push_rtp()).
https://bugzilla.gnome.org/show_bug.cgi?id=795139
If there is an external source which is about to timeout and be removed
from the source hashtable and we receive feedback RTCP packet with the
media ssrc of the source, we unlock the session in
rtp_session_process_feedback before emitting 'on-feedback-rtcp' signal
allowing rtcp timer to kick in and grab the lock. It will get rid of
the source and rtp_session_process_feedback will be left with RTPSource
with ref count 0.
The fix is to grab the ref to the RTPSource object in
rtp_session_process_feedback.
https://bugzilla.gnome.org/show_bug.cgi?id=795139
