Follow-up on 97d83056b3, only check
for intersection with the current srccaps when checking if a sinkpad
can accept caps.
I must have been lucky in my firefox testing then, and always entered
the code path with audio getting negotiated first, thus not failing
the is_subset check when srccaps had been negotiated as
application/x-rtp, and an accept-caps query was made for the video
caps with a defined extmap.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1384>
This is a requirement for GstPlayer when using the default overlay interface
provided by the pipeline. The GstPlayerWrappedVideoRenderer requires a valid
pipeline, but that's available only after the GstPlay thread has successfully
started.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1345>
Dynamically registered elements (hardware element in most cases)
may or may not be available on a system and properties may be different
per system.
This new API will make documentation skipping possible in programmable way.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1360>
ShowWindow() could be blocked while doing gst_d3d11_window_win32_unprepare
when external window handle provided to d3d11videosink in multi-threaded
environment.
The condition that issue happened is, UI thread is waiting for a
background thread that changes d3d11videosink state to NULL, and the
background thread would try to send a window message to the queue.
The queue is already occupied by the UI thread, so the background
thread will be blocked.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1366>
Instead of defining a sized array for function signature, use it
unsized (a pointer alias, basically). In this way clang warning is
silenced:
warning: ‘fill_profiles’ accessing 64 bytes in a region of size 12 [-Wstringop-overflow=]
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1357>
If qt asks us to redraw before we have both set a buffer and caps we
would attempt to use the new caps with the old buffer which could result
in bad things happening.
Only update caps from new_caps once the buffer has actually been set.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1346>
Make sure the EGLImage we're rendering to the GL memory stays alive long enough,
until the the GL memory has been destroyed.
This change fixes tearing and black flashes artefacts that were happening
because the EGLImage was sometimes destroyed before the sink actually rendered
the associated texture.
Fixes#889
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1354>
This takes a plain message string and not a format string, and as a
result doesn't have to be passed through vasprintf() and lead to further
unnecessary allocations. It can also contain literal `%` because of
that.
The new function is mostly useful for bindings that would have to pass a
full string to GStreamer anyway and would do formatting themselves with
language-specific functionality.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1356>
There's a potential race condition with this sort of pipelines on
certain systems (depends on the processing load):
GST_DEBUG_DUMP_DOT_DIR=/tmp \
gst-launch-1.0 uridecodebin3 uri=file://stream.mp4 ! glupload ! \
glimagesink --gst-debug=*:4
Right after the pipeline passes from PAUSED to READY, bin_to_dot_file
dumps uridecodebin3 properties, but current uri and suburi might be
already freed, causing a potential use-after-freed.
This patch makes NULL the current item right after all the play items
are freed.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1353>
A previous patch has caused rtpfunnel to output twcc-related
information downstream, however this leaked into upstream
negotiation (through funnel->srccaps), causing payloader to
negotiate twcc caps even when not prompted to do so by the user.
Fix this by only enforcing that upstream sends us application/x-rtp
caps as was the case originally.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1278>
Negative composition time offsets are only allowed with version 1 of the
box, however we parse it as a signed value also for version 0 boxes as
unfortunately there are such files out there and it's unlikely to have
(valid) huge composition offsets.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1294>
Sometimes the resampler has enough space to store all the incoming
samples without outputting anything. When this happens,
gst_audio_resampler_get_out_frames() returns 0.
In that case, the resampler should consume samples and just return.
Otherwise, we get a segfault when gst_audio_resampler_resample() tries
to resample into a NULL 'out' pointer.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1343>
g_sequence_remove_range's end iter is exclusive, so if one
wants to remove that item as well, it should be called with
the next iter.
This could in theory fix an issue where:
* The sequence isn't entirely trimmed, with an old item lingering
* Following FEC packets are immediately discarded because they
arrived later than corresponding media packets, long enough for
seqnums to wrap around
* We now try to reconstruct a media packet with a completely obsolete
FEC packet, chaos ensues.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1341>
We need to hold onto the last buffer until the next buffer arrives.
Before, if a caps change comes we would remove the currently rendering
buffer. if Qt asks use to render something, we would render the dummy
black texture.
Fixes a period of black output when upstream is e.g. changing resolution
as in hls adaptive bitrate scenarios.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1338>
The `gst_v4l2_buffer_pool_dqbuf` function contains this ominous comment:
/* get our GstBuffer with that index from the pool, if the buffer was
* outstanding we have a serious problem.
*/
outbuf = pool->buffers[group->buffer.index];
Unfortunately it is common for buffers in _output_ buffer pools to be
both queued and outstanding at the same time. This can happen if the
upstream element keeps a reference to the buffer, or in an encoder
element itself when it keeps a reference to the input buffer for each
frame.
Since the current code doesn't handle this case properly we can end up
with crashes in other elements such as:
(gst-launch-1.0:32559): CRITICAL **: 17:33:35.740: gst_video_frame_map_id: assertion 'GST_IS_BUFFER (buffer)' failed
and:
(gst-launch-1.0:231): GStreamer-CRITICAL **: 00:16:20.882: write map requested on non-writable buffer
Both these crashes are caused by a race condition related to releasing
the same buffer twice from two different threads. If a buffer is queued
and outstanding this situation is possible:
**Thread 1**
- Calls `gst_buffer_unref` decrementing the reference count to zero.
- The core GstBufferPool object marks the buffer non-outstanding.
- Calls the V4L2 release buffer function.
- If the buffer is _not_ queued:
- Release it back to the free pool (containing non-queued buffers).
**Thread 2**
- Dequeues the queued output buffer.
- Marks the buffer as not queued.
- If the buffer is _not_ outstanding:
- Calls the V4L2 release buffer function.
- Release it back to the free pool (containing non-queued buffers).
If both of these threads run at exactly the same time there is a small
window where the buffer is marked both not outstanding and not queued
but before it has been released. In this case the buffer will be freed
twice causing the above crashes.
Unfortunately the variable recording whether a buffer is outstanding is
part of the core `GstBuffer` object and is managed by `GstBufferPool` so
it's not as straightforward as adding a mutex. Instead we can fix this
by additionally recording the buffer state in `GstV4l2BufferPool`, and
handle "internal" and "external" buffer release separately so we can
detect when a buffer becomes not outstanding.
In the new solution:
- The "external" buffer pool release and the "dqbuf" functions
atomically update the buffer state and determine if a buffer is still
queued or outstanding.
- Subsequent code and a new
`gst_v4l2_buffer_pool_complete_release_buffer` function can proceed to
release (or not) a buffer knowing that it's not racing with another
thread.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1010>
The pipeline flow for receiving looks like this:
rtpsession ! rtpssrcdemux ! session_fec_decoder ! rtpjitterbuffer ! \
rtpptdemux ! stream_fec_decoder ! ...
There are two places where a fec decoder could be placed.
1. As requested from the 'request-fec-decoder' signal: after rtpptdemux
for each ssrc/pt produced
2. after rtpssrcdemux but before rtpjitterbuffer: added for the
rtpst2022-1-fecenc/dec elements,
However, there was some cross-contamination of the elements involved and
the request-fec-decoder signal was also being used to request the fec
decoder for the session_fec_decoder which would then be cached and
re-used for subsequent fec decoder requests. This would cause the same
element to be attempted to be linked to multiple elements in different
places in the pipeline. This would fail and cause all kinds of havoc
usually resulting in a not-linked error being returned upstream and an
error message being posted by the source.
Fix by not using the request-fec-decoder signal for requesting the
session_fec_decoder and instead solely rely on the added properties for
that case.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1300>
The average_period should always represent the time between two
events. The specification defines the event time as the time
between audio samples, video frame sync, video line sync, etc.
In case of one timestamp per PDU the timestamp_interval identifies
the amount of events between the timestamp of one PDU and the
timestamp of the next PDU.
As described in IEEE 1722-2016 chapter
"10.4.12 timestamp_interval field" timestamp_interval shall be
nonzero.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1076>
The failure conditions can be overidden by subclasses, and a boolean
return value is provided to the caller whether adding/removing the child
element has actually worked. The caller can then handle this
accordingly but flooding stderr with this is not very useful.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1320>
