* Trying to disconnect a stream from a running splitmuxsink by flushing
it results in the FLUSH_START blocking in the stream queue's
gst_pad_pause_task because the flush did not unblock
complete_or_wait_on_out, so add a check for ctx->flushing there.
* Add a GST_SPLITMUX_BROADCAST_INPUT so check_completed_gop notices
flushing changed and the incoming push is unblocked.
* Pass the FLUSH_STOP along to the muxer without waiting.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/687>
GCC 10 was complaining like following. It really is complaining about default cases returning
with potentially unitialized *desval, but those cases in the switch should never be hit.
```
../subprojects/gst-plugins-good/gst/auparse/gstauparse.c: In function 'gst_au_parse_chain':
../subprojects/gst-plugins-good/gst/auparse/gstauparse.c:481:37: error: 'timestamp' may be used uninitialized in this function [-Werror=maybe-uninitialized]
481 | GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
../subprojects/gst-plugins-good/gst/auparse/gstauparse.c:482:36: error: 'duration' may be used uninitialized in this function [-Werror=maybe-uninitialized]
482 | GST_BUFFER_DURATION (outbuf) = duration;
../subprojects/gst-plugins-good/gst/auparse/gstauparse.c:480:34: error: 'offset' may be used uninitialized in this function [-Werror=maybe-uninitialized]
480 | GST_BUFFER_OFFSET (outbuf) = offset;
cc1: all warnings being treated as errors
```
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/671>
This identifiers are registered in the MPEG-RA and defined
to be used by the Dolby Vision AVC/HEVC streams.
This is a first step to present the stream to the decoder.
Additional box parsing of DOVIConfigurationBox is necessary
to complete the media presentation with proper Dolby Vision
enhancements.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/658>
Previously, the user input for stsd entries is trusted completely, and
so a maliciously crafted file could choose the length of the stsd
entries arbitrarily and cause qtdemux to try to allocate up to 2GB of
memory (half of a 32 bit max int).
This patch fixes this by sanity checking the stsd input against the
size of the entire stsd atom.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/670>
During trak parsing, we need to check for the existence of stsd_entries,
otherwise, we end up with a NULL pointer to them. It is entirely
possible for the stsd to exist, but for it to have no entries, which the
previous checks did not take into account.
This patch adds a simply check to ensure that all files that do not
contain a stsd entry are deemed corrupt, and adds a test case to prevent
a regression.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/670>
Previously imagefreeze would always operate as non-live element and
output frames as fast as possible according to the configured segment
(via SEEK events) and the negotiated framerate from start to stop or the
other way around.
With the new live mode (enabled via the is-live property) it would only
output frames in PLAYING. Frames would be output according to the
negotiated framerate unless it would be too late, in which case it would
jump ahead and skip over the requirement amount of frames.
This makes it possible to actually use imagefreeze in live pipelines
without having to manually ensure somehow that it would start outputting
at the current running time and without still risking to fall behind
without recovery.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/653>
We never run as a live element, even if upstream is live, and never
output any buffers with latency but immediately generate buffers as
fast as we can according to the negotiated framerate.
Passing the query upstream would proxy whatever mode of operation
upstream has, which has nothing to do with how we produce buffers.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/653>
Move the SVMI stereoscopic atom parsing out to a helper
function to shrink qtdemux_parse_trak a bit.
Add a bounds check that the received atom is large enough
before parsing it.
Add a note to the atom parser that svmi comes from the
MPEG-A spec 23000-11.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/634>
Until now, do_expected_timeout() was shortly dropping the JBUF_LOCK in order
to push RTX event event without causing deadlock. As a side effect, some
CPU hung would happen as the timerqueue would get filled while looping over
the due timers. To mitigate this, we were processing the lost timer first and
placing into a queue the remainign to be processed later.
In the gap caused by an unlock, we could endup receiving one of the seqnum
present in the pending timers. In that case, the timer would not be found and
a new one was created. When we then update the expected timer, the seqnum
would already exist and the updated timer would be lost.
In this patch we remove the unlock from do_expected_timeout() and place all
pending RTX event into a queue (instead of pending timer). Then, as soon as
we have selected a timer to wait (or if there is no timer to wait for) we send
all the upstream RTX events. As we no longer unlock, we no longer need to pop
more then one timer from the queue, and we do so with the lock held, which
blocks any new colliding timers from being created.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/616>
When a GST_EVENT_FLUSH_START reaches the jitterbuffer, there is a chance that
our task is currently blocking waiting for a timer.
There was two problems:
* That wait wasn't checking for flushing situations
* The flushing handling wasn't waking up that conditional (to check whether it
should abort)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/608>
