In reverse playback, buffers are played back from buffer.stop
(buffer.pts + buffer.duration) to buffer.pts, which means that the
position after the buffer is consumed is buffer.pts, not buffer.pts -
buffer.duration.
Without that change, and when `automatic_eos` feature is on,
we were dropping the last buffers as marking the stream EOS one buffer
too soon.
This is now being tested extensively by GstValidate in the
`validate.test.clock_sync.*` set of tests.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/450>
* Making sure that `static inline` function are in the GIR (by first
defining them, and make sure to mark as skiped)
* Do not try to link to unexisting symbols
* Also generate GIR information about gst_tracers
Fixes flaky appsrc unit test where depending on scheduling
the submitted list might not be writable if submitted via
an action signal from the application thread.
Fixes gst-plugins-base#522
Otherwise it's possible that we won't be able to start again
depending the implementation. We do start/stop in normal use cases
whenever GST_QUERY_SCHEDULING happens before we are started.
https://bugzilla.gnome.org/show_bug.cgi?id=794149
The flushing state is handled a bit differently, there is no need
to stop flushing in start_complete. This would other result in
unlock_stop being called without unlock_start.
Unlike what the old comment says, there is no need to take the live
lock here, we are still single threaded at this point (app thread
or the state change thread). Also, we will wait for playing state
in create/getrange, no need to do that twice.
https://bugzilla.gnome.org/show_bug.cgi?id=794149
Add a gst_base_src_submit_buffer_list() function that allows subclasses
to produce a bufferlist containing multiple buffers in the ::create()
function. The buffers in the buffer list will then also be pushed out
in one go as a GstBufferList. This can reduce push overhead
significantly for sources with packetised inputs (such as udpsrc)
in high-throughput scenarios.
The _submit_buffer_list() approach was chosen because it is fairly
straight-forward, backwards-compatible, bindings-friendly (as opposed
to e.g. making the create function return a mini object instead),
and it allows the subclass maximum control: the subclass can decide
dynamically at runtime whether to return a list or a single buffer
(which would be messier if we added a create_list virtual method).
https://bugzilla.gnome.org/show_bug.cgi?id=750241
Holding this lock on live source prevents the source from changing
the caps in ::create() without risking a deadlock. This has consequences
as the LIVE_LOCK was replacing the STREAM_LOCK in many situation. As a
side effect:
- We no longer need to unlock when doing play/pause as the LIVE_LOCK
isn't held. We then let the create() call finish, but will block if
the state have changed meanwhile. This has the benefit that
wait_preroll() calls in subclass is no longer needed.
- We no longer need to change the state to unlock, simplifying the
set_flushing() interface
- We need different handling for EOS depending if we are in push or pull
mode.
This patch also document the locking of each private class member and
the locking order.
https://bugzilla.gnome.org/show_bug.cgi?id=783301
If segment.stop was given, and the subclass provides a size that might be
smaller than segment.stop and also smaller than the actual size, we would
already stop there.
Instead try reading up to segment.stop, the goal is to ignore the (possibly
inaccurate) size the subclass gives and finish until segment.stop or when the
subclass tells us to stop.
The subclass should do that already, but just in case do it ourselves too as a
fallback. Without this, e.g. playbin will just wait forever if this fails
because it is triggered as part of an ASYNC state change.
Currently, the query values are being set even if the query itself was
determined to have failed. Fix this to ensure the values are only set in
case of a query success.
https://bugzilla.gnome.org/show_bug.cgi?id=760479
Otherwise we're going to set a rather arbitrary DTS of segment.start (usually
0) for live sources, which confuses synchronization if the source started
capturing at a later time. And it's especially wrong for raw media, for which
we should not set any DTS at all.
https://bugzilla.gnome.org/show_bug.cgi?id=747731
It could be triggered by:
gst-launch-1.0 videotestsrc num-buffers=20 ! videcrop bottom=214748364 ! videoconvert ! autovideosink
Spotted while testing:
https://bugzilla.gnome.org/show_bug.cgi?id=743910
The flush-stop event should not restart the task for live sources unless
the element is playing. This was breaking seeks in pause with the rtpsrc.
https://bugzilla.gnome.org/show_bug.cgi?id=635701
