Otherwise it can happen that we regularly switch back and forth between
clocks under certain circumstances for no good reason.
Also remove redundant comparison when comparing the steps removed between two
clocks.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5520>
When the property "start-time-selection" is set to "first", it
calculates the start time of the output from the buffer pts
(converting it to running time of the segment), but if the
rate is negative, the real start is not the pts, but the
pts + duration, because it plays from the end of the buffer
to it's start.
As a result of this bug, in the negative rate, when the
start-time-selection=first, the first frame is dropped
by the videoaggregator (reproduced on d3d11compositor).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5276>
unlock_stop() is expected to be run while the streaming thread is idle. To
guaranty this is the case, we should take the streamlock, but its not
possible to take this lock during state transitions from PAUSED to
PLAYING as the wait function that we want to terminate is holding it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4961>
After a create() call, which may have returned FLUSHING or a filled buffer,
if it possible that we detect that we are now in pause. As live sourced
don't produce data in pause, drop the buffer is any and later retry creating
a buffer. This will ensure that we resume from pause while avoiding displaying
ancient frame.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4961>
When the pipeline goes from Playing to Paused, this change will invoke
unlock in the derived class. When the pipeline goes from Paused to
Playing, this change will invoke unlock_stop in the derived class.
This feature was removed in commit 523de1a9 and is now being restored.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4961>
Adds gst_queue_array_sort for sorting and gst_queue_array_push_sorted{,struct} for pushing in a sorted order.
All three functions accept a comparison GCompareDataFunc along with optional user_data to pass to it.
In gst_queue_array_sort a small workaround was needed to correctly sort non-struct arrays. Like what _find() already
does, we need to dereference our pointers first, to make sure we can use the same comparison functions everywhere.
This is done via a small wrapper around the provided comparison function.
The array can also wrap around (tail ends up 'before' the head), in which case we have to reorder the array (similar to
what do_expand() does) to then be able to use an existing sorting function, like g_qsort_with_data().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5112>
It's possible and normal to tear down a harness while the pipeline is
running. At the same time, it's desired for the
`gst_harness_pad_link_tear_down()` function to be synchronous.
This has created the conflict where the main thread may request a
harness to be torn down while it's in use or about to be used by a pad
in the streaming thread.
The previous implementation of `gst_harness_pad_link_tear_down()` tried
to handle this by taking the stream lock of the harnessed pad and
resetting all the pad functions while holding it. That approach was
however insufficient to handle the case where a non-serialized event
or query is being handled or about to be handled in a different thread.
This edge case was one race condition behind the flakes in the flvmux
check tests -- the rest being covered by https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2803.
This patch fixes the problem by adding an intermediate ref-counted
object, GstHarnessLink, which replaces the usage of the HARNESS_KEY
association. GstHarnessLink allows the pad functions such as event,
query and chain to borrow a reference to GstHarness and more
importantly, to lock the GstHarnessLink during their usage to block
(delay) its destruction until no users are left, and guarantee that any
future user will not receive an invalid GstHarness handle past its
destruction.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5017>
Race condition without this patch:
- srcpad task is being stopped in gst_aggregator_stop_srcpad_task()
- at that moment, in pre-queue event handler, gst_pad_get_task_state()
returned GST_TASK_PAUSED
- then in srcpad task got stopped in gst_aggregator_stop_srcpad_task()
- finally srcpad task got resumed in pre-queue event handler
To address it, checks "running" flag in pre-queue event handler.
Both pre-queue stream-start event handler and "running" flag
are protected by SRC_LOCK already.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4847>
If the time server is restarted with a time in the past the net client
clock will not report the new time anymore as this would mean that the
clock moves back in time which it does not do.
Now the clock will be kept alive but marked as corrupted and will not
be re-used from the cache.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4802>
Otherwise it only works if GStreamer is binding the first socket on this
port.
Unfortunately this requires duplicating a bit more of Rust std because
`UdpSocket` can only be created already bound without allowing to set
any options between socket creation and binding.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4807>
According to the documentation this should never happen but apparently
does under certain circumstances. As the sockets are set non-blocking,
trying to read from them regardless should not cause any problems.
In all cases that were observed so far, the socket in question actually
has a packet queued up for reading.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4748>
While this doesn't yet use any OS provided times from the actual network
stack, this still gets rid of any IPC jitter between the helper process
and the main process as part of the PTP time calculations and should
improve accuracy.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4665>
ptpd is defaulting to the hybrid mode, and was sending invalid multicast
PTP messages in that configuration until ce96c742a88792a8d92deebaf03927e1b367f4a9.
While this commit was made in 2015 there was no release in the meantime.
Work around this by detecting this case and defaulting to the default
values for the given intervals as given by the PTP standard.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4654>
The generated gir file marks the size parameter as "out" by default.
This is wrong in the context of a caller allocated buffer with a given size.
Explicitly marking the size parameter as (in) fixes the issue.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4399>
This works on Linux, Android, Windows, macOS, FreeBSD, NetBSD, OpenBSD,
DragonFlyBSD, Solaris and Illumos.
Newly supported compared to the C version is Windows.
Compared to the C version various error paths are handled more correctly
and a couple of memory leaks are fixed. Otherwise it should work identically.
The minimum required Rust version for compiling this is 1.48, i.e. the
version currently in Debian stable. On Windows, Rust 1.54 is needed at
least.
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1259
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3889>
gst_base_src_new_segment() does not send the segment right away, which
may break events ordering if subclass sends other events after
calling it.
Introducing a variant pushing the segment right away to preserve
ordering in such cases.
Will be used by appsrc which has its own internal queue where we need to
preserve events order.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4297>
