Previously we only handled one event at a time, which could lead to the
following two suboptimal situations:
- frame 0 at 20ms, frame 1 at 40ms and two force-keyunit events at 10ms
and 15ms. We would create a new keyframe for both of the frames.
- 100 force-keyunit events with running-time NONE would cause all
following 100 frames to be made into a keyframe.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/684>
When linking source pads to decodebin, make sure we use the *specified* new
source pad and not some random one.
This avoids ending up with source pads being unlinked.
Main cause of random timeouts with rtsp change_state_intensive validate tests
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/687>
Also not optimal but at least simplifies the code a bit and doesn't
require g_list_length() and g_list_append() in a few places.
For 2.0 there are some more candidates to change but unfortunately
they're currently part of the API.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/683>
This means we cannot access [view layer] or view.bounds from the OpenGL
thread. This also means that we need to call the main thread when
setting the window handle. However, we cannot perform that
synchronously as that may deadlock with the application performing the
set_window_handle() call.
We need to defer the actual update and run it asynchronously and wait
for the window handle update internally at each point it is needed.
Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/372
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/681>
Otherwise there is a mismatch between the QoS values and what upstream
would expect, leading to too much buffer dropping in video decoders in
case rate < 1.0 or not enough buffer dropping in case rate > 1.0
Adding validate tests with and without decoders.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/679>
We need to take into account the base_ts to compute next_ts and it needs
to be updated on rate change.
This introduces `pending_rate` so that change rate is properly handled
in the streaming thread in a safe way.
Added tests
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/679>
Currently, videoscale just drops all metas that have other tags
besides video. However videoscale wont change the colorspace or
the orientation of the video so metas tagged as such may be
copied safely. Additionaly, given that videoscale will change
the frame size, we invoke the meta transform implementation
to give it the opportunity to scale accordingly.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/548>
Especially when changing the sample rate our timestamp tracking will be
completely off, but even otherwise we would usually lose the last few
samples if we don't drain here as the resampler gets reset if anything
but the sample rate changes.
This is usually not a problem as the first buffer after a caps event
usually has the discont flag set, but can cause problems if
- the caps event is followed by a segment event, which then causes
draining according to the new sample rate
- the caps were changed because of rengotiation due to a reconfigure
event and there is not discontinuity from upstream
In both cases we would output buffers with completely wrong timestamps.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/670>
Stop comparing all timestamps from buffers that are before the segment
with the segment.stop and compare with the actual end times.
Comparing to segment.stop for all the buffers that where before
the segment.stop was incorrect and leading to consuming wrong buffers
and not respecting segment.stop, this is now properly tested.
Expectations for `reverse.10_to_1fps.validatetest` have been fixed to
take that into account and comparing the checksums of the sinkpad and
srcpad expectations makes pretty clear how wrong that was.
(we can see in the expectations that videotestsrc outputs an extra
buffer with pts == segment.stop and this one is now properly dropped
by videorate as bec7f4ad5e aimed at
doing)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/668>
In reverse playback we were not taking into account the current buffer
samples to check if we had reached EOS which was leading to a buffer
with PTS = CLOCK_TIME_NONE containing too many frames followed by a
useless buffer with pts=0 duration=0, and a g_critical issue in
gst_object_sync_values.
Also add a validate based test case.
Without that patch this is how the expectation fails:
``` diff
--- log-asink-sink-expected 2020-05-22 23:22:42.654384579 -0400
+++ log-asink-sink-actual 2020-05-22 23:29:35.671586380 -0400
@@ -27,5 +27,6 @@
buffer: pts=0:00:00.058820861, due=0:00:00.023219955, flags=discont
buffer: pts=0:00:00.035600907, due=0:00:00.023219954, flags=discont
buffer: pts=0:00:00.012380952, due=0:00:00.023219955, flags=discont
-buffer: pts=0:00:00.000000000, due=0:00:00.012380952, flags=discont
+buffer: due=0:00:00.012380953, flags=discont
+buffer: pts=0:00:00.000000000, flags=discont
event eos: (no structure)
```
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/667>
The gir generator wrongly assume that the vfunc
GstGLFilterClass.filter() and the method gst_gl_filter_filter_texture()
are related. As a result it complains about not matching argument names.
Workaround this by naming both of their arguments input and output.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/664>
