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>
