Fixes outputted frame sequence when performing a seek
i.e. when seeking backwards, the first frame after the seek was a frame
from the future. This would result in GstVideoDecoder essentially
marking all the timestamps as essentially bogus and the base class would
attempt to compensate. A visible indication of this was 'decreasing timestamp'
warning after a seek.
https://bugzilla.gnome.org/show_bug.cgi?id=790478
A deserialised timecode has a framerate of 0/1 by default. That breaks
it when comparing the frames field with another timecode (incoming from
the frame). We were setting the framerate when receiving the caps event,
but not when setting the timecode in set_property, so it was broken for
timecodes set after the caps event.
Also checking if the fps_n we got from the caps event is != 0 before
setting it - also at the caps event.
https://bugzilla.gnome.org/show_bug.cgi?id=790334
Now that timecodes support proper serialisation / deserialisation, a
timecode might have an invalid fps_n / fps_d even without using the
target-time-code-string property. Detect those cases and set fps_n/fps_d
properly.
The "fields" flag is ignored because currently GStreamer doesn't support
having only top or only bottom fields inside a frame. The "drop frame"
flag is ignored because some occurrences have been spotted where it
wasn't set while it should have been. In practice, when we have 29.97 or
59.94 FPS, it's always drop-frame.
https://bugzilla.gnome.org/show_bug.cgi?id=790112
SRT[0] is an open source transport technology[1] that optimizes
streaming performance across unpredictable networks.
Although SRT is based on UDP, it works like connection-oriented
protocol. However, it doesn't mean that the SRT server or client
is necessarily to link to a receiver or a sender so, here, the
pairs of source and sink elements are introduced.
- srtserversink: SRT server to feed SRT stream
- srtclientsrc: SRT client to get SRT stream from srtserversink
- srtclientsink: SRT client to send SRT stream
- srtserversrc: SRT server to listen from srtclientsink
[0] https://github.com/Haivision/srt
[1] http://www.srtalliance.org/https://bugzilla.gnome.org/show_bug.cgi?id=785730
If end_tc is NULL, it means that we don't want avwait to stop at any
timecode. When explicitly setting end_tc to NULL, there is no point in
comparing end_tc with start_tc (to see if we'll reject end_tc for being
before start_tc), so the check in question is completely disabled
instead of letting it crash.
Add support for parsing linear time code from
an audio source using libltc
https://github.com/x42/libltc
The user can now choose between 3 different and independently
running timecode sources. The old override-existing property
has been replaced by timecode-source.
https://bugzilla.gnome.org/show_bug.cgi?id=784295
This element can be configured to add jitter and/or drift to incoming
buffers' PTS, DTS, or both. Amplitude and average of jitter and drift
are configurable.
https://bugzilla.gnome.org/show_bug.cgi?id=787358
When we receive a video or audio buffer, we calculate the next stream
time based on the current stream time + buffer duration. If the next
buffer's stream time is after that, we issue a warning.
This happens because the stream time incoming from Decklink should be
really constant and without gaps. If there is a gap, it means that
something went wrong, e.g. the internal buffer pool is empty (too many
buffers queued up downstream).
https://bugzilla.gnome.org/show_bug.cgi?id=781776
