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>
Upstream caps might for example be
application/x-rtp,media=audio,encoding-name={OPUS, X-GST-OPUS-DRAFT-SPITTKA-00, multiopus}
and while that is not fixed caps it is enough to match it with a media.
Only caps structures that have the correct structure name and that have
the media and encoding-name field are preserved, but if both are present
then these caps are used as "codec preferences".
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1291>
Currently reading extension relies on the fact that everything after the
last"." character is a file extension. Whereas that works fine for most
of the cases, it breaks when the URI contains a query part.
E.g.: `http://url.com/file.mp4?param=value` returns `mp4?param=value`
instead of `mp4`.
In this commit we use URI parser to read the path of the URI (in the example
above, that is `/file.mp4`) and read extension from that path.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1305>
Some decoding APIs support delayed output for performance reasons.
One example would be to request decoding for multiple frames and
then query for the oldest frame in the output queue.
This also increases throughput for transcoding and improves seek
performance when supported by the underlying backend.
Introduce support in the mpeg2 base class, so that backends that
support render delays can actually implement it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1013>
Downstream might need the start code offset when decoding.
Previously this computation would be scattered in multiple sites. This
is error prone, so move it to the base class. Subclasses can access
slice->sc_offset directly without computing the address themselves
knowing that the size will also take the start code into account.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1013>
The GstV4l2CodecAllocator dispose function clears `self->decoder` but
the finalize function then tries to use it if the allocator has no been
detached yet.
Fix by detaching in the dispose function before we clear
`self->decoder`.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1220>
Standard interlace handling:
* If we have interlace-mode=interleaved and the field order, we just
set it when creating the session
* If we have interlace-mode=(interleaved|mixed) and no field order, we
set the field order on the first buffer
The encoder session does not support changing the FieldDetail after it
has started encoding frames, so we cannot support mixed streams
correctly.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1214>
Delay decoders downstream negotiation just before an output frame
needs to be allocated.
This is required, are least for H.264 and H.265 decoders, since
codec_data might trigger a new sequence before finishing upstream
negotiation, and sink pad caps need to set before setting source pad
caps, particularly to forward HDR fields. The other decoders are
changed too in order to keep the same structure among them.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1257>
Using GstBaseDec hack to access the parent_object of each element in
the element itself is a bit fragile. It would be better to keep its
own parent object as the usual global variable. It would make it
resistant to code changes.
The GstBaseDec macro to access the parent object now it's internal to
base decoder.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1257>
We were checking if the start time of the gap event was
GST_CLOCK_TIME_NONE, which is superfluous because that cannot happen,
and then not checking if it was NONE after gst_segment_to_running_time,
which caused a crash if an identity received a gap event fully or
partially outside the current segment.
This patch was done in cooperation with:
Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1269>
.. if a current direction has already been set
When `webrtcbin` has created an offer based on codec_preferences,
it might not have received caps on its sinkpads by the time a
remote description is set, in which case we want to connect the
input stream upon actual reception of the caps instead.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1233>
With mpeg4videoparse drop=false config-interval=N|-1 we might be
trying to insert a config before we have actually received one,
in which case we'll try to map a NULL buffer which will generate
lots of criticals.
Fixes#855
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1265>
gst_va_fixate_format() will iterate all othercaps' structures to find
the one with less information lost at color conversion. If a structure
with same color format is found, the iteration stops. It's like a
smart truncation. Then, this function also will choose the caps
feature.
Later this structure is used fixate its size and no further truncation
is needed.
Don't intersect at fixate, since it kills possible resizing.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1261>
If the pad does not have a current caps, get_pad() returns the query
caps which can be ANY. In such case the caps does not have any structure
resulting in a critical warning when calling gst_caps_get_structure().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1254>
Detected while reading the code, cccombiner must set
self->current_video_buffer to NULL *after* emitting selected-samples
in order for the application to get a useful return when peeking
the next video sample.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1252>
When schedule is true (as is the case by default), we insert padding
when no caption data is present in the schedule queue, and previously
weren't checking whether the caption pad had gone EOS, leading to
infinite scheduling of padding after EOS on the caption pad.
Rectify that by adding a "drain" parameter to dequeue_caption()
In addition, update the captions_and_eos test to push valid cc_data
in: without this cccombiner was attaching padding buffers it had
generated itself, and with that patch would now stop attaching
said padding to the second buffer. By pushing valid, non-padding
cc_data we ensure a caption buffer is indeed attached to the first
and second video buffers.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1252>
Also refactor various internals of the monitor code:
- Don't allow starting twice but just return directly when starting a
second time.
- Don't end up in an inconsistent state if call start() a second time
while the monitor is starting up.
- Remove complicated cookie code: it was not possible to add/remove
filters while the monitor was started anyway so this was only useful
in the very small time-window while starting the monitor or while
getting the devices. Instead disallow adding/removing filters while
the monitor is starting, and when getting devices work on a snapshot
of providers/filters.
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/667
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1189>
At gst_va_dmabuf_allocator_setup_buffer_full, static code analysis tool
does not know number of objects in descriptor is always larger than 0 if
export_surface_to_dmabuf succeeds. Thus, the tool will assume buf is
allocated with mem but not released when desc.num_objects equals to 0
and raise a mem leak issue.
For gst_va_dambuf_memories_setup, we should also inform the tool that
n_planes will be larger than 0 by checking the value at very beginning.
Then, the defect similar to above will not be raised during static analysis.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1241>
gst_base_parse_reset() does not reset data_bytecount to 0, so
gst_base_parse_update_bitrates() uses a wrong value to calculate
the average bitrate on subsequent pipeline starts. This leads to an
excessive amount of "tag" events being pushed. These events include
very high "bitrate" values that diminish over time, and are produced
until the average bitrate is back to sane values.
Fixes#840
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1228>
This will only affect individual/tarball module builds, as the
options yield to the parent project which was set to gpl=disabled
by default already. We kept it as auto in the original commit
to accommodate the need to update cerbero as well, which had to
be done separately after the initial commit.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1217>
Previously gst_structure_has_name was used to get a string to compare with supported mimetypes.
This is incorrect as above function returns a user defined structure name which is
not the structure mimetype value.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1206>
Trying to reset before the pads have been deactivated races with the
streaming thread. There was also a buggy buffer clear leaving a dangling
`stored_frame` pointer around. Use `gst_interlace_reset` so this happens
properly.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1039>
We don't support D3D11 interop for UWP because some APIs
(specifically MFTEnum2) are desktop application only.
However, the code for symbol loading is commonly used by both UWP and WIN32.
Just link GModule unconditionally which is UWP compatible, and simply don't
try to load any library/symbol dynamically when D3D11 interop is unavailable.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1216>
... in favour of dep.get_variable('foo', ..) which in some
cases allows for further cleanups in future since we can
extract variables from pkg-config dependencies as well as
internal dependencies using this mechanism.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1183>
When ass hinting value is set to anything other than NONE,
subtitles cannot use smooth scaling, thus all animations will jitter.
The libass author warns about possibility of breaking some scripts when it is enabled,
so lets do what is recommended and disable it to get the smooth scaling working.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1201>
WebVTT in ISO MP4 is specified in ISO 14496-30,
and needed for DASH support. It's stored in an
mp4 specific format. To handle it compatibly,
the wvtt boxes are converted back into WebVTT text
and pushed as application/x-subtitle-vtt
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1182>
Swap the `need_process` boolean check on qtdemux streams
for a direct function pointer to the splitting function,
so we can stop adding extra cases to the single growing
`gst_qtdemux_process_buffer()` function.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1182>
Currently the extension data length specified in the RTP header would
say it was shorter then the data serialised to a packet. When
combining the resulting buffer, the underlying memory would still
contain the extra (now 0-filled) padding data.
This would mean that parsing the resulting RTP packet would potentially
start with a number of 0-filled bytes which many RTP formats are not
expecting.
Such usage is found by e.g. RTP header extension when allocating the
maximum buffer (which may be larger than the written size) and shrinking
to the required size the data once all the rtp header extension data has
been written.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1146>
The code in the aes elements assumes OpenSSL >= 1.1.0:
- implicit library initialization;
- version retrieved with OpenSSL_version(OPENSSL_VERSION);
and it fails to build with older versions.
Specify the required OpenSSL version explicitly in meson.build so that
the elements are excluded on older systems (e.g. Ubuntu 16.04) and the
rest of GStreamer can still build.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1067>
An inactive pad is a pad which, in live mode, hasn't yet received
a first buffer, but has been waited on at least once.
Exposing API to support this behaviour allows users of aggregator
subclasses to request pads, and not start pushing data on those
immediately, while avoiding systematic timeouts.
Subclasses must check in explicitly to this behavior, most likely
by exposing a user-facing property, and must check whether a pad
needs ignoring when aggregating. That is because by design,
aggregator subclasses don't get a list of "ready" pads, but instead
directly iterate element->sinkpads.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/867>
Since commit a55dafe341, stream-scoped tags no
longer appeared as top-level tags, introducing a behaviour regression, specially
for MP3 files.
The `gst_discoverer_info_get_tags()` API now returns all tags detected for the
given media, as documented.
A new API is introduced to get container-specific tags,
`gst_discoverer_container_info_get_tags()`. The discoverer tool was adapted to
use it. `gst_discoverer_info_get_tags()` is now deprecated in favor of
`gst_discoverer_container_info_get_tags()` and
`gst_discoverer_stream_info_get_tags()`.
Fixes#759
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1107>
gst_va_vpp_complete_caps_features() now receives the @feature_name to
add and return if @caps doesn't provide it.
So, instead of two nested loops, now the function is a single loop,
traversing @caps to find if each structure already contains the requested
@features_name.
It's important to add missing caps features with @caps, in order to
not lost information.
The function caller does the external loop by calling per each
available caps feature.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1024>
In order to make more readable the caps transformation, the operation
was split in two phases:
1. Rangify the supported caps structures.
2. Add the missing (and supported) caps features.
Step 1 modified its logic, by copying any unrecognized structure.
It's a previous step required for allowing ANY caps feature as
passthrough.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1024>
For monorepo build and ugly/bad, for advanced feature
option API like get_option('xyz').required(..) which
we use in combination with the 'gpl' option.
For rest of modules for consistency (people will likely
use newer features based on the top-level requirement).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1084>
There are streams in the wild that have to add a SCTE-35 trigger in
another e.g. GA94 stream. Most encoders would replace the GA94
descriptor ID with the CUEI one temporarily, but there are some that
will add two registration ID descriptors, one with GA94 and one with
CUEI.
Failing to parse the CUEI registration ID in that case would return
FALSE in _stream_is_private_section , therefore setting it as known PES
and pushing packets downstream instead of calling handle_psi.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/979>
In the encoded streams we might not have all the information about the
raw video stream, but when reencoding they end up being specified, even
if those are default values.
As vp8 decoders always output frames in some YUV color space we can
ensure that when upstream doesn't specify any value in its caps we
use the default one which is what we end up doing when decoding/reencoding
anyway, so this way downstream (matroskamux in that case) doesn't need
to be able to renegotiate (which it doesn't).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1062>
We should also take into account whether data is currently pending when checking
for gap on streams. It could very well be that some streams have very low
bitrate (and spread out) data. For those we don't want to push out a gap event.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1179>
This is only enabled in push time mode. Furthermore it's only enabled for now if
PCR is to be ignored.
The problem is dealing with streams where the initial PTS/DTS observation might
be greater than following ones (from other PID for example). Before this patch,
this would result in sending buffers without any timestamp which would cause a
wide variety of issues.
Instead, pad segment and buffer timestamps with an extra
value (packetizer->extra_shift, default to 2s), to ensure that we can get valid
timestamps on outgoing buffers (even if that means they are before the segment
start).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1179>
Introduces a `libraries` variable that contains all libraries in a
list with the following format:
``` meson
libraries = [
[pkg_name, {
'lib': library_object
'gir': [ {full gir definition in a dict } ]
],
....
]
```
It therefore refactors the way we build the gir so that we can reuse the
same information to build them against 'gstreamer-full' in gst-build
when linking statically
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1093>
Since 547570cd79 we do not always build
PyGObject and our development environment is broken when trying to use
GStreamer python when built against system PyGObject with the following
error importing Gst in there:
```
12345678** (gst-plugin-scanner:710617): CRITICAL **: 11:45:02.343: can't find gi.repository.Gst
Traceback (most recent call last):
File "/usr/lib/python3.9/site-packages/gi/repository/__init__.py", line 23, in <module>
from ..importer import DynamicImporter
File "/usr/lib64/python3.9/site-packages/gi/importer.py", line 33, in <module>
from .overrides import load_overrides
ImportError: cannot import name 'load_overrides' from 'gi.overrides' (/var/home/thiblahute/devel/gstreamer/gstreamer/subprojects/gst-editing-services/bindings/python/gi/overrides/__init__.py)
Factory Details:
```
The approach to fixing it is to implement override `gi` in
`gst-python/gi/` which we add to `PYTHONPATH`) and in there reset the
`gi` module to the right place and we get overrides from paths from
`_GI_OVERRIDES_PATH` we set in `gst-env.py` which points to all the
overrides that will be installed.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1155>
Making a null check in gst_va_decode_picture_free () indicates pic->buffers or pic->slices
can be null, then in _destroy_buffers () the pointers are dereferenced, which is detected
as dereference after null check by Coverity. Thus, modify the code to do null check in
_detroy_buffers ().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1143>
Retrieving the pad template caps from a ghost pad returns ANY which when
merged with any other caps will return ANY. ANY is not very specific
and may cause suboptimal code paths in e.g. decoders that assume the
lowest common denominator when presented with ANY caps.
Fixes negotiating dma-buf with vaapidecodebin between glupload in the
video sink element.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1144>
And use a simple GStreamer pipeline as testsrcbin with GstTranscoder
doesn't let us easily set the framerate of the source and we end up
having videorate dropping frames leading to the rendered file having
an unprecise duration.
This should fix races with `check.gst-editing-services.pythontests.pyunittest.python.test_assets.TestTimeline.test_reload_asset`
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1130>
The first approach to fixate was simply a copy&paste of both
videoconvert and videoscale, trying to keep their logic as isolated
as possible. But that brought duplicated and sparse logic.
This patch merge both approaches simplifying the fixate operation.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1109>
Add a helper function to get, from GstVideoInfo and GstBuffers flags,
the VA interlace surface flags. This is used currently by vainterlace
element, but it will be used in vapostproc too if it can process
interlaced frames.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1109>
Guard against the orientation not coming from an inexistant tag, nor
from the application (rotation set to "auto") which caused an assertion.
When the application requests the auto rotation method, make sure it is
resolved to a rotation that's applicable.
ERROR:gstreamer/subprojects/gst-plugins-good/ext/gtk/gtkgstglwidget.c:745:gtk_gst_gl_widget_set_rotate_method: code should not be reached
Fixes: 103ceb853a
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1134>
The src and sink elements no longer link against libsoup. It is now loaded at
runtime. If any version is resident already, it is used. Otherwise we first try
to load libsoup3 and if it's not found we fallback to libsoup2.
For the unit-tests, we now build one version of the test unit file per libsoup
version found. So if both libsoup2 and libsoup3 are available on the host, the
CI will cover them both.
Based on initial patch by Daniel Kolesa <dkolesa@igalia.com> and
Patrick Griffis <pgriffis@igalia.com>.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1044>
Actually extract the .o objects from the convience libraries and put
them into the main one. Without this, they will just be referenced by
the .pc file, but it will be unusable because they are not installed.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1122>
This new class is a helper for fast/tricky copy of surfaces. First it
tries to copy using the function vaCopy in libva 1.12. If it fails, or
it's not available, a GstVaFilter is tried to be instantiated with the
allocator's parameters, and if succeed, it's used for copying the
source surface.
This is required for dmabuf surfaces with drm modifier.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1023>
Initially GstVaSample processed its GstBuffer member to get the
VASurfaceID. But it might cases where we already have the VASurfaceID
to process by the filter.
This patch enables the possibility to pass the surfaces rather than
the buffers. In order to validate the surfaces a function to check
surfaces were added.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1023>
If the USE_PLAYBIN3=1 env var is set, we want to replace
playbin with playbin3, but separate to that, we always
want to register the 'playbin3' element so that applications
which explicitly use playbin3 work regardless of the env var.
This fixes `USE_PLAYBIN3=1 gst-validate-launcher`, for example.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1102>
... instead of index of DXGI adapter.
The order of IDXGIAdapter1 enumerated via IDXGIFactory1::EnumAdapters1
can be varying even there's no rebooting in case that GPU preference order
is updated by user (for example, it can be done by using NVIDIA Control Panel
in case of multi-GPU laptop system) and eGPU is another possible case.
So, for an element which requires fixed target GPU requirement,
index based device enumeration is unreliable.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1098>
* gst_d3d11_device_new_for_adapter_luid()
Used for creating D3D11 device for a DXGI adapter (i.e., GPU)
corresponding to a LUID (Locally Unique Identifier).
This method can be useful for interop with other APIs such as
Direct3D12, MediaFoundation, CUDA, etc.
* gst_d3d11_device_new_wrapped()
Allows creating a new GstD3D11Device object by using already
configured ID3D11Device. This is conceptually equivalent to
gst_gl_context_new_wrapped()
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1098>
Change locking around group deactivation to avoid deadlocks
when shutting down exactly as a buffering message arrives.
The PLAYBIN3_LOCK now protects the active field of the
source group. Everything else is still protected by the
source-group-lock.
Also properly protect group switching operations with
the PLAYBIN3_LOCK everywhere.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1049>
Previously suspended medias immediately reached the UNPREPARED state
without going through the media's unprepare() vfunc. This didn't allow
the media subclass to do any additional cleanup, and for example the
shutdown-eos property of GstRTSPMedia was ignored.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1090>
Windows doesn't support fork so every test will be performed in
one process. So the test_meta_custom_transform() is being
failed because "test-custom" custom meta is being used/defined in
another test test_meta_custom() as well.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1086>
The GST_VIDEO_DECODER_ERROR() should be used only for robust/error-resilient
decoding purpose. Any other error codes such as not-negotiated or flushing
should be returned without modified for upstream to be able to handle
it immediately. (for example, application might want to try other
decoder element on not-negotiated)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1070>
tests/check/meson.build uses the openjpeg_dep variable
unconditionally, and the subdir_done() is useless anyway, since the
plugin is only built if openjpeg_dep.found() is true. Fixes:
..\tests\check\meson.build:23:0: ERROR: Unknown variable "openjpeg_dep".
In particular, this fixes the build on UWP since we disable openjpeg
explicitly in Cerbero when building for UWP.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1069>
handle_*_request() functions were all retrieving the session media from
the session by calling gst_rtsp_session_get_media () which is a transfer-none
call. If a session timeout happens at that time, the session media may get freed
making the pointer invalid..
Fixes#757
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1053>
Instead of assuming that the PTS of a keyframe is the lowest PTS of a
GOP, wait until the DTS has passed this PTS and take the minimum PTS up
to that point. That way the minimum PTS of a GOP can be determined, at
least for closed GOP streams. Open GOP streams still can't be handled
properly.
By knowing the minimum PTS of each GOP, keyframes can be requested at
the correct time relative to the GOP (and thus fragment) start and
fragment overflow calculations can calculate the correct durations of
the GOPs.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1005>
In an embedded system where all services run as seperate users it is
useful to have the gstreamer registry readable by all so it can be
re-used, in similar manner as a host system where one user have seperate
applications running but all share same registry.
To make this possible introducing GST_REGISTRY_MODE for adjusting the
changing mode of the registry binary when finishing up with the
temporary file (which has restricted access).
Fixes: #692
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/825>
If the query has already been destroyed at this point, GST_IS_QUERY will
read garbage, can return false and we will try to unref it again.
Instead, make note of whether the item is a query when we dequeue it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1029>
boolean return value is not sufficient for representing the reason
of error in most cases. For instance, any errors around new_sequence()
would mean negotiation error, not just *ERROR*.
And some subclasses will allocate buffer/memory/surface on new_picture()
but it could be failed because of expected error, likely flushing
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1019>
boolean return value is not sufficient for representing the reason
of error in most cases. For instance, any errors around new_sequence()
would mean negotiation error, not just *ERROR*.
And some subclasses will allocate buffer/memory/surface on new_picture()
but it could be failed because of expected error, likely flushing
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1019>
boolean return value is not sufficient for representing the reason
of error in most cases. For instance, any errors around new_sequence()
would mean negotiation error, not just *ERROR*.
And some subclasses will allocate buffer/memory/surface on new_picture()
but it could be failed because of expected error, likely flushing
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1019>
warning C4003: not enough arguments for function-like macro invocation 'warning'
G_STMT_END macro is extended to the below form with MSVC
__pragma(warning(push)) \
__pragma(warning(disable:4127)) \
while(0) \
__pragma(warning(pop))
So MSVC preprocessor will extend it further to
__pragma(VBI_CAT_LEVEL_LOG(push)) ...
Should rename warning() debug macro function therefore.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1018>
libgudev is a problematic dependency, particularly in sandboxed
environments, such as flatpak.
This patch implements a way to get the available VA devices using
brute-forced traverse of /dev/drm/renderD* directory. Thus usable in
those sandboxed environments.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1027>
When move the libgstva, libgudev dependency was moved as part of the
library, though it's not use by the library but the plugin. This patch
moves back libgudev dependency to the plugin.
Also HAVE_LIBDRM is move to the library which is the one who use it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1027>
Some decoding APIs support delayed output for performance reasons.
One example would be to request decoding for multiple frames and
then query for the oldest frame in the output queue.
This also increases throughput for transcoding and improves seek
performance when supported by the underlying backend.
Introduce support in the vp9 base class, so that backends that
support render delays can actually implement it.
Co-authored by Seungha Yang <seungha@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/987>
Since 0d95d9258b we respect the asset stream-id in `GESUriSource` so
we can not work with unknown or broken stream ID in the assets.
We just ignore them, warning about it and we should fix that in
demuxer so they don't expose pad without providing a stream id for them.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1001>
This patch contains two updates:
1. Instead of checking for dependency already checked just to verify a
version, we use the dependency version API.
2. Update the deprecated function get_pkgconfig_variable.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/997>
It's possible to have installed MediaSDK environment
package (libmfx-dev in Debian) without libva environment package. This
setup will lead to a breakage of meson configuration.
The fix is to get the libva's driver directory variable after the
dependency is validated as found.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/998>
When using the following setup (the error can be reproduced using
simpler sender pipelines), the receiver resynchronises the clock on RTCP
packets. The effect was that a couple seconds were cut out of the
playback because an initial RTCP packet was dropped.
When sending out all RTCP packets (setting sync=FALSE on the RTCP
updsink), the playback is fine.
This syncs rtpsink with rtpsrc (where this property was already set).
gst-launch-1.0 filesrc location=899-en.mp3 \
! mpegaudioparse \
! mpg123audiodec \
! audioconvert \
! audioresample \
! avenc_g722 \
! rtpg722pay
! rtpsink uri=rtp://239.1.2.3:1234
gst-launch-1.0 uridecodebin rtp://239.1.2.3:1234?encoding-name=G722 \
! autoaudiosink
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/993>
Since the base class now does the parsing, there is no need
to reproduce that code in all the subclasses, just pass the attributes
which are the only relevant bit anyway.
Also, only store the direction if the subclass accepted the caps
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/906>
If we are not receiving a sync-point for a very long time, we need to
keep asking for them. The request-sync-point logic keeps track of how
many keyunitrequests we are allowed to send, but that would not matter
if we don't keep asking.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/930>
When there are elements between the demuxer and the muxer that
introduce an offset to the running time, or when offsets are
set on pads by the application, this shift must be taken into
account when calculating the final pts_adjustement.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/913>
mpegtsmux can receive SCTE sections from two origins: events
created by the application, and events forwarded downstream by
mpegtsdemux, containing sections that may not have been fully
parsed, and additional data to help tsmux translate times to
the correct domain, both for requesting keyframes and calculating
an accurate pts_adjustment.
The complete approach is documented further in a comment above
the relevant function.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/913>
Instead of modifying the splice times in the incoming sections
to running time and expecting eg mpegtsmux to convert those back
to its local PES time domain, which might be impossible when
those splice times are encrypted or the specification is extended,
transmit the needed information to the muxer as separate fields in
the event:
* A pts offset field can be used by the muxer in order to calculate
a final pts_adjustment
* A rtime_map can be used by the muxer to determine the correct
running times at which it should request keyframes
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/913>
Makes it possible to support passing SCTE 35 cue points from
demuxer to muxer, while preserving correct timing.
This will also improve ex nihilo cue points injection, as splice
times and durations are now interpreted as running time values,
and may trigger key unit requests.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/913>
