Existing tensor decoder has been bifurcated into two seperate gst
element as:
`yoloodv5tensordecoder`: decodes tensors output(masks) from detection-only
models e.g yolov8s.onnx
`yolsegv8tensordecoder`: decoder tensors output(masks and logits) from
segementation models e.g FastSAM or yolov8s-seg
YOLOv8 model have same tensor output format as FastSAM, so for better
generalization rename fastsamtensordecoder to yolotensordecoder. This
also requires code adaptation to support Yolo based model.
In RTSP 2.0, they started recommending that clients use an
empty SET_PARAMETER request as the preferred keep-alive mechanism
as all cameras must support that method. It also works
for RTSP 1.0, so use it as the preferred method unconditionally.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9577>
When rtspsrc is using an interleaved connection to the camera,
it will continously receive a stream of tunnelled RTP packets
from the server, so the rtspconnection doesn't return ETIMEOUT
and trigger the keepalive. Add a check on the RTSP timer
and send the keepalive whenever we haven't sent a message to
the server in some time.
Fixes problems with some cameras that don't see the RTCP traffic
as sufficient proof of liveness, when using TCP/HTTP tunnelled
modes
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9577>
Now that VideoPool accept DMA DRM, it is quite likely that the size does not get
provided in the configuration even if the format have a matching GstVideoFormat.
Fix this by reporting back the minimum expected size in the configuration. This
allows the configuration handshake to succeed.
Fixes regression caused by !9345
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9600>
x265 encoder_headers return headers with SEI after encoding the frame,
while the output frame also contains SEI so two identical header
blocks appeared.
Cache the headers at init, leaving only a single copy in the stream.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9604>
Fix race condition where wrong caps were sometimes received before
the last gap internal clip by setting explicit framerate in video
caps and adjusting test timing accordingly.
The test now uses framerate=1/1 with single buffer expectations
instead of relying on default 30fps timing that created timing
races with the internal gap clip processing.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9608>
Always add GstTensorMeta to buffer instead of re-using existing GstTensorMeta
when it's already present. The reason for the change is when we have cascaded
inference elements overwriting (tflite) or leaking (ONNX) GstTensorMeta when we
have cascaded inferences. We didn't create and API to append to GstTensorMeta
and I think it's more convenient to have a new GstTensorMeta for each
inferences. It's also more clear that a group of tensor was produced by one
inference but doesn't limit tensordecoders from using tensors produced by
multiple inference. I don't see much value in having all tensors data always
inside one GstTensorMeta since appending would mean re-allocation of the tensors
array anyway.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9564>
LCEVCdec supports different pixel aspect ratios other than 1/1. This change
forwards the pixel aspect ratio of the base picture to the LCEVC decoder,
and also updates the output pixel aspect ratio caps base on the one from the
enhanced frame.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9463>
The output resolution is not always twice as big as the input resultion divided
by the pixel aspect ratio. This is the case for LCEVC '0D' mode, where the
output resolution is the same as the input resolution, and the only enhancement
is the picture being clearer.
This patch uses LCEVC_PeekDecoder() after sending the LCEVC enhancement data to
know what the output resolution will be before allocating the output picture.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9463>
Even though the LCEVC decoder works fine without this, it is recommended to
set read access to base pictures that are sent to the decoder, and write access
to enhanced pictures that are received from the decoder.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9463>
.. in decoders.
Inline session parameters allows to not create session parameters handlers for
every new stream parameters (such as SPS and PPS for H.264, for example), but
instead to pass them as a chained structure in the decoding main structure. This
is completely align with GStreamer decoder base classes.
Even that the previous approach is kept, if the devices doesn't support video
maintenance2, it shows a lot of validation errors.
Also it was required to add another parameter when enabling extension to verify
if the extension is linked with a device feature and if it is enabled.
Bump Vulkan API (and driver version for both decoders and encoders) to 1.4.306
Also bumped the ABI_CHECK_TAG because the CI finally catches up with the vulkan
video symbols that are not exposed by a public header (tough they are binary
public).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9585>
Some extensions need to have enabled certain feature in the device. This patch
does that check by adding a new field in the extension list which is a function
that can be mapped to gst_vulkan_physical_device_has_feature_*() functions.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9585>
See #2611
When transitioning to PLAYING we call IDeckLinkInput::StartStreams() (via GstDecklinkInput::start_streams). This can
fail for various reasons, including if the hardware device is already in use.
Previously, we logged StartStreams() failure as an element error but otherwise continued to successfully transition to
PLAYING. Now, if StartStreams() returns an error we fail the state change.
If StartStreams() fails then a subsequent call to StopStreams(), when transitioning PAUSED to READY, will also fail.
Previously, if StopStreams() failed then the state change also failed. Unfortunately this prevented the element from
later being disposed, which in turn meant the associated hardware resources was never freed. Consequently, if a
decklinkvideosrc failed to transition PAUSED to READY, then the associated hardware device could not subsequently be
used by any other decklinkvideosrc.
Now, we log an element error if StopStreams() fails but otherwise consider the state change to have succeeded. This
means that the element can be disposed and the associated hardware resource released.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9578>
The responsibility to change the framerate to the one
requested on the VideoTrack should always be the last time effect in it.
This introduces a new `capsfilter` at the end of video effect that
will be in passthrough most of the time.
t show
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9013>
There was a case where we could end up having more buffers on the sinkpads
than what was expected, depending on timings. This ensure that we have a fixed
number of buffers produced by the testsrc, and we play until EOS.
Replace theora encoder/decoder chain with fakevideodec for testing
QoS handling in videorate rate property tests. This simplifies the
test pipeline and removes dependency on theora codecs.
Also adjust test expectations to match RGBA format output from
fakevideodec instead of I420 from theoradec, and limit buffer
counts to make tests more deterministic.
This also fixes the rate we set for 2.0 test as we were actually testing
0.5 there.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9013>
When executing a stop action, ensure that any pending wait_message_action
is cleared and unreffed to prevent it from being executed after the
scenario has been stopped. This prevents potential crashes or unexpected
behavior when stopping a scenario that has a pending wait action.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9013>
Use the local context variable that was retrieved earlier instead of
action->priv->context which is always NULL at this point since we
reset it right before. This ensures we invoke the callback on the
correct context.
And make sure that the wait message still has a reference when setting done
otherwise it is unref before calling the context invoke function
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9013>
Add support for an 'overrides' field in validatetest meta structures that
allows changing issue severity levels during test execution. This enables
tests to pass when encountering known issues by downgrading their severity.
The overrides field accepts an array of change-severity structures with:
- issue-id: The issue ID to override
- new-severity: New severity level (critical, warning, issue, ignore)
Currently only change-severity overrides are supported. The feature follows
the same pattern as expected-issues and is only available in .validatetest
files, not .scenario files.
Includes comprehensive documentation and a test case demonstrating the
functionality.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9013>
Refactor the videorate element to properly separate upstream and downstream
time domains when the `rate` property is set.
The previous implementation had issues with time domain conversions when
changing rates, especially during seeks after which we were basically confusing
input and output timestamps.
This also fixes rate changes as we are now tracking the wanted input timestamps
and not confusing them with the output, so this way we do not drop buffers when
the rate is changed while playing, meaning that the related tests have been
fixed
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9013>
When the dots tracer is detected among active tracers, switch from
SHOW_ALL to SHOW_FULL_PARAMS to prevent parameter ellipsization.
The dots tracer is designed to work with gst-dots-viewer which can
handle long parameter values, so we preserve the full text in dot files.
This improves the viewing experience when using the dots tracer with
the gst-dots-viewer tool.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9547>
This checks if a tracer has registered to any hook after
initialization and automatically registers it to the "none" hook if not
so the lifetime of those tracer is the same as tracer who registers to hooks.
For example the `dots` tracer was being freed right after initialization
which was unexpected.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9547>
When GStreamer pipeline graphs contain references to other pipeline dot files,
users should be able to navigate between related pipeline views by clicking
on these references. This enables exploring complex pipeline hierarchies
where one pipeline references sub-pipelines.
The implementation detects text elements containing pipeline-dot references,
styles them as web links, and handles click events to navigate to the
referenced pipeline while preserving existing text selection and drag
functionality for other elements.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9547>
Enable text selection on SVG text elements by changing CSS user-select
properties from 'none' to 'text' and cursor from 'default' to 'text'.
Add JavaScript event handling to intercept mousedown events on text
elements, preventing dragscroll interference while allowing normal text
selection. This preserves the existing drag-to-pan functionality for
non-text areas while making text elements selectable and copyable.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9547>
With video_maintenance1 extension is possible to create images independent of a
the video profile list, under that image will be processed.
With that extension is possible to share the same image for dynamic transcoding.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9550>
Change the logic to skip only devices which have "Input" as their IOID. The ALSA
Input/Output identifier (IOID) it may be either "Input", "Output", or NULL; with
the latter meaning that the device supports both input and output.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9534>
This was broken in b29262e662b86cd76225894f7504ee505125fa49.
The `gst_util_uint64_scale` emitted a critical warning and returned
`GST_CLOCK_TIME_NONE`, so beyond removing the warning this fix does not
change behavior.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9415>
The validation complained with
VUID-VkImageCreateInfo-usage-requiredbitmask]:
vkCreateImage(): pCreateInfo->usage is zero.
This patch force to use the internal defaults in vkimagebufferpool if no usage
is defined.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9544>
With merge request !9438 the validation layer started to complain with
VUID-VkDeviceCreateInfo-pNext-02829
This patch fixes this ill-usage of Vulkan API, by removing the feature enabling
of sampler ycbcr conversion, since it was promoted since Vulkan 1.1, and in
GStreamer is only used in Vulkan Video operations, which are only enabled in
Vulkan 1.3+.
Also, these features detection and enabling were moved to a function called when
filling the physical device data, in order to check the API version of the
device driver before adding the enabling of the feature.
Finally, the getters were adapted to use the version feature structure if the
device driver version matches.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9544>
Pushing the buffer via gst_pad_push () in transform_ip () function
causes downstream elements to process the buffer with a reference
count > 1. This leads to performance issue if there are downstream
elements which modify the buffer memory.
However, in drop-only mode this reference is not required.
So, let GstBaseTransform push the buffer in drop-only mode.
Fixes#4258
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9532>
Don't reuse the same offset and size variables when reading
the byterange out of a MAP directive, as it can overwrite
values from a pending BYTERANGE directive for the next
fragment URI.
Fixes problems where the EXT-X-MAP directive has been written
into the playlist between an EXT-X-BYTERANGE and the fragment
URI it applies to.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9533>
Don't reuse the same offset and size variables when reading
the byterange out of a MAP directive, as it can overwrite
values from a pending BYTERANGE directive for the next
fragment URI.
Fixes problems where the EXT-X-MAP directive has been written
into the playlist between an EXT-X-BYTERANGE and the fragment
URI it applies to.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9533>
If mpegtsmux receives the same caps again, it wrongly claims the stream
type changed:
error: Stream type change from 06 to 8f not supported
This adds a unit test that demonstrate the issue in the very hacky way.
I have seen this happening with the below pipeline when upstream caps
changes. Since the caps filter fixates the caps received by opusenc and
mpegtsmux, the stream type cannot change.
...
! audioconvert
! audio/x-raw,format=S16LE,channels=2,rate=48000
! opusenc bitrate=128000
! mpegtsmux
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9430>
The TsMuxStream internal_stream_type field stores the original
'full' stream type (such as Opus), while the stream_type field
stores the value that will actually be written into the MPEG-TS
packets according to the codec mappings. When checking if
input caps are changing stream type, check the original type.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9430>
In order to link videomaintenance1, and others to come, without knowing if
Vulkan 1.4 features are chained in the device properties structure, a static and
inlined function was added in gstvkphysicaldevice-private.h. It was added in a
header file to avoid compiler warnings if it's not used because of old Vulkan
headers.
Also the value dump videomaintenance1 was moved to another function to pack
there all the future features to query which aren't part of a Vulkan release
yet.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9492>
This patch is the payment to my technical debt.
The symbol GST_VULKAN_HAVE_VIDEO_EXTENSIONS is defined at compilation-time if
the user requests for the usage of the Vulkan Video extensions. And we used this
symbol for anything related with Vulkan Video. But this is not the smartest
approach.
The rule should be:
- If the code allocates Vulkan Video resources, use
GST_VULKAN_HAVE_VIDEO_EXTENSIONS
- Otherwise, use the Vulkan's guard for the used API
In this way, API version bumps will be easier.
Also, this commit marks the end of GST_VULKAN_HAVE_VIDEO_EXTENSIONS guarded code
for readability.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9492>
Both gst_vulkan_image_buffer_pool_set_config() and
gst_vulkan_image_buffer_pool_alloc() functions share the same code to create
Vulkan images for different purposes.
This patch refactor them in a new helper function that creates the images and
stores them in a buffer if it's passed as output parameters, such as the
offsets.
This patch also adds specifics guards for Vulkan's symbols for better grained
API usage, but also for prepare_buffer() the guard is set where the symbol is
used.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9492>
The Vulkan Video extensions can be available, according to the specification,
since Vulkan 1.1, but with other extensions dependencies. That's why this patch
adds a field in the extension structure, which represents the extension
dependency that the specified extension requires. And they are specified by
Vulkan Video extensions.
This allow to have a single function to check if the extension can be enabled
both by optional extensions and video extensions.
Regardless that video extension can be loaded since Vulkan 1.1, they are rather
loaded since Vulkan 1.3, when synchronization2 was promoted, so it isn't
checked as video_queue dependency.
Finally, this patch checks for each guard symbol.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9492>
This is a continuation of !9483, but without back-porting.
Instead of checking the driver's API version to figure out if a physical device
feature is available and enabled, or even more, instead of checking for enabled
extensions in the driver, this patch adds private functions in the physical
driver to get the availability and enabling of features such as sampler ycbrc
conversion, synchronization2, timeline semaphore and video maintenance1.
And these new functions are used internally in the GstVulkanOperation object,
and the private object GstVulkanDecoder.
This approach is computationally cheaper, simpler and precise.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9492>
plugin_init() will not get called if element/feature registration
happens manually, such as when using linking only specific plugin
features with gstreamer-full. That is possible when plugins contain
static features.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9496>
Since the sequence of IMMDeviceEnumerator::EnumAudioEndpoints()
followed by IMMDevice::Activate() is not atomic, Activate() may fail
if the enumerated device becomes invalidated before probing.
In such cases, retry device probing
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9468>
Add a pre-commit hook that automatically checks if GES children properties
documentation is up to date when GES source files are modified. The hook
runs the document-children-props.py script and warns if the generated
documentation files have changes that need to be committed.
This ensures that children properties documentation stays synchronized
with code changes and prevents outdated documentation from being merged.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5987>
The script was failing with newer pygobject versions because the
__enum_values__ attribute has been removed. Updated to use proper
introspection by:
- Getting enum type from property default value
- Using enum_class.values for the first enum value (index 0)
- Creating other enum values with enum_type(i) constructor
- Extracting name and nick from enum values properly
Also fixed ruff linting issues:
- Removed unused imports and variables
- Used specific exception types instead of bare except
- Added noqa comment for gi import ordering
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5987>
For the features VkPhysicalDeviceSamplerYcbcrConversionFeaturesKHR,
VkPhysicalDeviceSynchronization2Features and
VkPhisicalDeviceTimelineSemaphoreFeatures
The Vulkan specification states:
If the `VkPhysicalDeviceXXXFeatures` structure is included in the `pNext`
chain of the `VkPhysicalDeviceFeatures2` structure passed to
`vkGetPhysicalDeviceFeatures2`, it is filled in to indicate whether each
corresponding feature is supported. If the application wishes to use a
`VkDevice` with any features described by `VkPhysicalDeviceXXXFeatures`,
it **must** add an instance of the structure, with the desired feature members
set to `VK_TRUE`, to the `pNext` chain of `VkDeviceCreateInfo` when creating
the `VkDevice`.
And that was missing in the code.
Strangely, that functionality doesn't have a Valid-Usage ID assigned, so it
isn't caught by the validation layer.
This patch adds the structures in the physical devices to get them and later set
them in the device creation.
In order to link these features, videomaintenance1, and others to come, without
knowing if Vulkan 1.3 features are chained in the device properties structure, a
static and inlined function was added in gstvkphysicaldevice-private.h. It was
added in a header file to avoid compiler warnings if it's not used because of
old Vulkan headers.
Also the value dump videomaintenance1 was moved to another function to pack
there all these queried features.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9483>
If there is only unknown stream-type streams in the current collection
don't post an error straight away. This fixes a problem with RTSP
cameras and legacy upstream collection building, if the first
stream that rtspsrc outputs is the ONVIF metadata track. That
happens often on bandwidth-constrained camera inputs, as the
video and audio will naturally take longer to arrive.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9433>
Launch-lines will be pasted into a shell, and `gst_value_serialize()`
yields strings that will likely be interpreted by the shell. For
example:
`gst-launch-1.0 ... ! osxaudiosink unique-id="AppleUSBAudioEngine:BEHRINGER:UMC202HD\ 192k:12345678:1\,2"`
The shell will remove the double-quotes `"` but keep the `\ ` which
means the output of `gst_value_deserialize()` will not be the original
string, and the launch line will not work.
So let's use `gst_value_serialize()` only if the string is non-ASCII,
and if it's ASCII and needs quoting, we do some shell quoting.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9466>
VA encoders, at reconfiguration, have to check if the rate-control was changed
by the user, but since user parameters setting are in another thread, the
comparison was racy.
This patch locks the object to compare the current rate-control with the one set
by the user.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9480>
Make it work a little more like RTP. Have the source interact with the
clock and set the capture time on each packet. Then the other elements
can use that to do adjustments. Since AVTP is always very low latency,
it can be assumed that the gPTP clock at the packet reception is very
close to the sending time, never more than 2 seconds off, so the
timestamps can be compared directly.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9412>
Caps may change the framerate used and the frame counting approach for
timestamps, needs to account for this by taking a snapshot of the current frames
and running time to add to all subsequent produced frames. Code is mostly taken
from videotestsrc.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9455>
When the base videodecoder class re-attempts a negotiation after flush, the
vabasedec `need_negotiation` flag isn't necessarily set to TRUE, because in that
situation the input state hasn't changed.
By always chaining up we are sure that buffer pool negotiation will always be
attempted.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9457>
When the base videodecoder class re-attempts a negotiation after flush, the
vabasedec `need_negotiation` flag isn't necessarily set to TRUE, because in that
situation the input state hasn't changed.
By always chaining up we are sure that buffer pool negotiation will always be
attempted.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9457>
When the base videodecoder class re-attempts a negotiation after flush, the
vabasedec `need_negotiation` flag isn't necessarily set to TRUE, because in that
situation the input state hasn't changed.
By always chaining up we are sure that buffer pool negotiation will always be
attempted.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9457>
When the base videodecoder class re-attempts a negotiation after flush, the
vabasedec `need_negotiation` flag isn't necessarily set to TRUE, because in that
situation the input state hasn't changed.
By always chaining up we are sure that buffer pool negotiation will always be
attempted.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9457>
This is needed if the LCEVC enhancement data is part of the video stream as SEI
and the demuxer outputs 'lcevc=false' src caps because LCEVC enhancement data is
not stored as a separate stream in the container.
To clarify, 'lcevc=true' just means that the video buffers have LCEVC metadata
attached. Therefore, it is valid to have a stream with LCEVC enhancement data as
SEI with 'lcevc=false' as long as it is not attached as metadata.
This will be needed once we add support for the demuxer to attach LCEVC metadata
to video buffers if it is stored in a separate track.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9427>
This broke in 81bdef0150df3306136137a5187dc25a7c5f4aee. Simplify
removal of the plugin from the plugin list if features are listed, and
fix invocation of the script -- it was adding an extra space before
the first plugin filename.
Also ensure that gst-full CI is run when the script changes.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9452>
Just because a plugin name is a substring of another plugin, doesn't
mean it is the same plugin. If the list of plugins can contain
a filename, that will be a static lib of the form libgstfoo.a
For example, if you had `-p applemedia -e app:appsrc,appsink` then
`applemedia` would be removed from the plugins list.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9442>
Add d3d12fisheyedewarp element that performs fisheye image dewarping
using D3D12. A UV remap LUT texture is generated via a compute shader,
and the actual remapping is performed in a pixel shader using this LUT
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9441>
If enabled, wasapi2src/sink will post a warning message instead of an error,
when device failures occur, such as open failure, I/O error,
or device removal.
The element will continue to produce/consume audio buffers and behave as if
a capture/render device were active, allowing pipeline to keep running even when
no audio endpoint is available
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9326>
By default, `gst-env.py` spawns a new shell with an updated environment
suitable for working with current branch.
The option `only-environment` dumps the resulting environment on `stdout`
without spawing a new shell. This can be used to save it to a file and `source`
it later, possibly from another shell.
However the resulting environment contains all the env vars from the initial
environment, some of which are session, display or DE specific (e.g.
`XAUTHORITY`, `DESKTOP_SESSION`, `WAYLAND_DISPLAY`, ...). This can cause
problems when sourced later.
Reproduction:
1. `gst-env.py --only-environment > env-full`
2. `source env-full`
3. `gst-launch-1.0 videotestsrc ! ximagesink` <= this works
4. Reboot
5. `source env-full`
6. `gst-launch-1.0 videotestsrc ! ximagesink` <= this doesn't work because
`XAUTHORITY` was overriden with previous value in step 5.
This was initially observed with a Qt application running with the X11 backend.
This commit changes the behaviour of `only-environment` to only dump env vars
that are actually set or modified by `gst-env.py`.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8846>
Upon RTCP timeout, rtpsession emits a signal that we catch to set a
timed_out private field, this also exposes it as a property of the
transport in order for users (such as rtspclientsink) to get notified
about it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9124>
... even if hardware does not support interlaced encoding at bitstream level.
Although interlacing information is not written in the bitstream,
that information can be signalled via container, thus allow interlaced
stream.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9328>
Add a new action signal to allow applications to capture
the most recently rendered frame directly from the swapchain
back buffer.
Unlike the existing "last-sample" property, which exposes
the raw input sample before any sink-side processing, this
signal captures the final displayed image after any internal
image processing (e.g., UV remap, color balance, overlay) has been
applied.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9423>
- Add a videoconvert element before the videosink so that the output
works no matter what format gets negotiated (A444_16LE for me)
- Specify a reasonable video format and size with a capsfilter, so
we don't default to something silly like A444_16LE @ 240p.
- Add a timeoverlay element, so it's obvious when stoppping/restarting
the pipeline that the input stream is just picked up again from the
moment the consumer pipeline is restarted.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9422>
Modify the API to retrieve the tensor meta to check for the dimensions
as well.
Also fix an API mistake, the buffer whose dimensions should be checheck
is the one inside the GstTensor, not another buffer some outside.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9411>
Similar to h264parse and h265parse, this patch enhances the element to parse
LCEVC enhancement data from SEI, and attach it to output buffers as GstLcevcMeta.
The 'lcevc' field in the output caps is also set to TRUE or FALSE depending on
whether LCEVC data is present or not.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9365>
Similar to h264parse and h265parse, this patch improves the element to parse
the SEI registered user data from NAL units. The core structure of H266 SEI for
ITU-T T.35 is the same as the other parsers, so we can re-use the same logic.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9365>
The older 0.28 version has a buggy --msvc-syntax option:
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8717#note_3001648
This 32-bit version was compiled from our meson port of pkg-config:
https://gitlab.freedesktop.org/gstreamer/meson-ports/pkg-config
The toolchain used was MSYS2's mingw32-gcc toolchain
`mingw-w64-i686-gcc` so that it uses 32-bit MSVCRT, which gives it
maximum compatibility with all Windows versions. UCRT DLLs are
specific to the VS version and Windows version.
This version will also not insert any system library or include paths,
like the previous 0.28 version. The build commands were:
```
meson setup --optimization=s --strip \
--wrap-mode=forcefallback \
-Dpc_path='' \
-Dsystem_include_path='' \
-Dsystem_library_path='' \
_mingw32
meson install --destdir $PWD/install -C _mingw32
cp install/bin/pkg-config.exe .
zip pkg-config-$VERSION.zip pkg-config.exe
```
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9376>
This causes servers such as AWS MediaLive to drop the connection, and
the message flow chart as documented in the "spec" always has the
server sending it first, and the client replying to it on reception of
the Set Peer Bandwidth, which we do since 286a3829b637.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9360>
Previous version passed just enough data to the parser to detect
main/main-10, but main-422-10 is one of the range extensions profiles.
Those need a few more bits to be accurately detected, and since those
were just uinitialized memory previously, we'd incorrectly end up with
main or main-10 when the encoder was in fact giving us 4:2:2 10bit
output.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9348>
Similar to h264parse, this patch enhances the element to parse LCEVC enhancement
data from SEI, and attach it to output buffers as GstLcevcMeta. The 'lcevc'
field in the output caps is also set to TRUE or FALSE depending on whether LCEVC
data is present or not.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9335>
The download element relied on a fuzzy translation from GStreamer format to a
DRM fourcc, and then all supported modifiers for that fourcc. Since !9306 this
was fixed to only enumerate that way when direct import is used.
Flag direct upload to the transform caps helper, so that we now enumerate all
non-external formats again.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9339>
When using direct DMABuf upload, supported DRM formats and modifiers
pairs should be translated to RGBA. Instead of overwriting the translation
to RGBA, which may endup having nothing to override, we introduce a new
flag for the transform helper, so it can do direct translation.
This fixes a regression introduced by !9306, and fixes more negotiations
issues.
Fixes#4525
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9339>
self->window is created with the first frame, so it is not available when
properties are set during construction of the element.
Skip calling gst_wl_window_ensure_fullscreen() in this case.
The window is already constructed with the current configured fullscreen state,
nothing else in needed here.
Without this, running e.g. 'gst-launch-1.0 -v videotestsrc ! waylandsink
fullscreen=true' will result in:
GStreamer-Wayland-CRITICAL **: 14:11:19.921: gst_wl_window_ensure_fullscreen: assertion 'self' failed
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9283>
Some WebRTC implementations such as Pion are unhappy if the
profile-level-id isn't returned with a compatible profile as the
RFC requires. Let's try to reform it
In practice, the correct way to do this would be to not use caps
intersection, but to instead implement the correct RFC compliant
SDP O/A negotiation of formats.
Include a unit test written by Philippe Normand
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9031>
This reverts commit fd1428d3ebf937d7656c3788df3ef3ff85fecc31.
As reported in #4524, this changes cause regressions. The problem is due to a
bug in how vp9parse interact with parsebin, presenting downstream negotiation of
alignment to work. This revert to being stuck using frame alignment always,
which fortunately works with libvpx, though less efficient.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9329>
The virtual driver VISL does not support dynamic resolution changes. Before the
vp9parser fix, the first frame would be decoded and display and the rest
dropped. Though, since we fixed the handling of super frame, that frame is now
dropped, since it is marked DECODE_ONLY. All other frames in that stream can't
be decoded. So no frame gets decoded which leads to an error.
Simply skip this test until visl can handle this type of stream.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8944>
Only the last frame of a super frame should be displayed by default. When
converting from super frame to frame, mark all frames as decode only except the
last one. This fixes vp90-2-22-svc_1280x720_3.ivf conformance test with
stateless decoders such as VA.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8944>
H.264 base class oriented for hardware accelerated encoders, such as Vulkan, VA
and others.
1. It can be parametrized for hardware limits, such as lists size, b-frames
supports, etc.
2. It produces a GOP structure map [IDR, R/I/B, ...)
3. It proposes parameters set and other strucures such as bitrate limites.
Subclases can modify those structures.
4. It calls the subclass encode virtual method implementation.
It doesn't handle rate control algorithms or other encoding quality mechanisms.
For a deeper introduction to the class there was a lighting talk in the GstConf
2024: <https://www.youtube.com/watch?v=-fQY54KHH38>
Co-authored-by: He Junyan <junyan.he@intel.com>
Co-authored-by: Michael Grzeschik <m.grzeschik@pengutronix.de>
Co-authored-by: Stéphane Cerveau <scerveau@igalia.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7197>
We were building a different image cause we needed to do
a couple things that are not possible yet through
cbuild/ci-templates.
Now that we have FDO_DISTRIBUTION_POST_EXEC we can
do it on the same job and use the same image tag to
extend the image.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9277>
... and merge wasapi2{capture,render}deviceprovider into single
wasapi2deviceprovider since we can enumerate input/output audio
devices at once using IMMDeviceEnumerator
This is a preparation for complete porting to Win32 API
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9307>
Install properties at the given offset as intended instead of at 0.
Currently there are no elements with any properties, so this has no
effect. This change is needed if any element adds properties in the
future.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9179>
This is for cases where:
* We *do* want to refer to the PCR stream to figure out global positioning, gap
detection, wrapover correction.
* But we do not want to apply any skew correction to the output
This is useful for cases where:
* the input stream has already been clock-corrected (for example with
mpegtslivesrc)
* or where the output doesn't require synchronization against a clock (ex: for
storage)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9301>
Docker/oci and thus gitlab-runner, default to a root
user inside the namespace, even if its an unprivileged
one.
This can cause issues and let permission bugs sneak in,
as we are functionally root when running the build.
Switch the build jobs to run with our new "containeruser"
so we avoid much of it.
Our user is still in the wheel/sudo group but that's fine
as long we don't elevate the privileges unintentionally.
Noticeably for the time being, we will need to chown the
CI_PROJECT_DIR checkout as the gitlab runner might try
to reuse pre-existing and cached volumes of the project
checkout.
Additionally we need to change the ccache path, so we
will avoid the existing cache owned by "root".
Close https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2433
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8763>
We are currently running pip as root which will cause it
to install into /root/.local cause it wants to do a user install
Set the user-base in the gloabl config, /etc/pip.conf, so pip will
both install there and subsequently look there if we invoke it
from any other user.
This makes pip install ofc require elevated permissions, as it
will be writting into /usr/local from now on
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8763>
Create a file we can check at runtime, and identify if the
environment we run against is one of our CI build images.
Useful mostly for our internal scritps so we can match against
metadata rather than heuristics, ex. if /subprojects exists
Conceptually similar to /.flatpak-info
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8763>
- Add case where we have element in subset that are equal to element in
superset. In this case subset is still a subset if at least one element is a
subset of the corresponding element in superset.
- clarify gst_value_is_subset inline documentation
- Add helper VALUE_TYPE_SINGLETON (type) to identify GType that represent a
value that is not a collection.
- Fixed issue in gst_value_is_subset_array_array (<1, 2, 3>, <0, 1, 4, 2, 3>)
would have returned TRUE because because the alignment of the subset could be
done multiple times. Now once alignment of set as been done once, everything
else in the superset need to be a subset of corresponding element without
interuption. is_subset (<1, 2, 3>, <0, 1, 2, 3>) => TRUE, but is_subset (<1,
2, 3>, <0, 1, 4, 2, 3>) => FALSE.
- If both array are fixed value and we perform a is_subset on them, both value
can't be equal to be considered subset. (For consistency with other fixed
values)
- Define gst_value_subtract (singleton, array) and
gst_value_subtract (array, singleton), to allow a gst_value_is_subset
(singleton, array) and gst_value_is_subset (array, singleton). General
case of gst_value_is_subset (v1, v2) use gst_value_subtract (v1, v2) and
gst_value_subtract (v2, v1) to evaluate _is_subset(), therefore we need these
for is_subset() operation that involve singleton and array.
Note gst_value_subtract (array, array) is not implemented but this case is not
require by gst_value_is_subset () and as it has a direct handling. Warned if
the case is used.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9220>
When doing direct dmabuf upload, we rely on the GL stack for doing the color
transformation. The caps we transform from GL to DMABuf are always with a format
of RGBA. Instead of listing all GstVideoFormat and translating them back into
DRM formats, simply list all supported DRM format for the context.
This enable rendering DRM formats that don't have an shader based emulation
implemented such as NV15.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9306>
The transformation was fuzzy, adding random modifiers to the list. Use the newly
introduce helpers from 1.26 to precisely convert GStreamer formats to a DRM
fourcc and modifier pair and vice-versa.
This fixes support for formats that have a GstVideoFormat value and requires a
modifier.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9306>
It is not possible to do frame cropping when DMABuf caps feature is negotiated.
The VideoInfo size is zero, resulting in empty destination buffers, and video
convert library may not understand what the format actually is.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9305>
If the conformance window does not requires cropping the top or left of the
window, we can use GstVideoMeta to crop in a zero-copy fashion. If a copy
is needed, the frame copy can also handle it, and is a lot faster.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9305>
Similar to what qtdemux does in `gst_qtdemux_stream_update_segment()`,
in which case we want to resend tags according to our position.
By setting the changed flag for all stream tags we ensure they will all
be (re-)send as we progrees.
This notably ensures that rotation tags are not lost when video players
finish playback and jump back to the beginning of a video.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8429>
Similar to and inspired by glimagesink, xvimagesink and others.
The waylandsink never transform the buffer in any way but delegates this to the
Wayland compositor with the Wayland buffer transform API.
Rotation and window size are already supported, so this just changes the video
surface geometry that is communicated to the Wayland compositor.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9210>
Cleans up the code and fixes two issues:
- If there are no streamheaders in the caps but we have `HEADER`
buffers, it would run `gst_buffer_list_foreach` with `self->headers`
being `NULL`.
- The code forgot to unmap the buffer if it decided to ignore it.
Fixes: 0a562a92d7ee38d8919d1b802add84d3c93b59eb
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9284>
Adding "hip-amd-precompile" build option. If enabled, AMD kernels
will be precompiled at build time. Also "hip-hipcc-arch" build option
(corresponding to --offload-arch hipcc option) is added
so that user can specify target GPU arch instead of auto-detection by hipcc
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8923>
In a previous version of the clock time conversion code, scheduled playback used
to be started (from 0) when transitioning to PLAYING and stopped when
transitioning PLAYING->PAUSED. This worked fine when converting running times
using the internal clock. However, now the decklink clock will produce values
that are monotonically increasing and do not reset to 0 at the same moments as
running time anymore. This means that the clock adjustments could attempt to
convert a small running time based on a large clock time e.g. after pausing
for many hours. As the adjustment code is a simple linear interpolation based on
the current clock times (large) using the provided value (small), the small
differences in the rate could result in very large differences in the
output time.
Fix by instead using both internal and external clock times based on the values
that gst_clock_get_calibration() will return. By doing so, small changes in the
rate calculations between the internal and external clock times will not result
in potentially large differences in the output internal time from
gst_clock_unadjust_with_calibration().
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/4197
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9267>
Prevent race condition where gst_play_sink_do_reconfigure() could be called
from a pad probe while stream synchronizer pads are being released during
GST_STATE_CHANGE_PAUSED_TO_READY transition.
The race occurred when:
1. State change starts releasing stream synchronizer pads
2. Pads are unblocked earlier in the state change, allowing events to flow
3. A streaming thread triggers sinkpad_blocked_cb -> gst_play_sink_do_reconfigure
4. Reconfiguration tries to use already-released pad pointers
5. New pad creation fails with assertion in gst_pad_iterate_internal_links
The fix adds GST_PLAY_SINK_LOCK around the pad cleanup to ensure atomic
cleanup and prevent concurrent access during state transitions.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9233>
After gst-validate-1.0 notices the first expected error on the bus, it will
queue the completion of that action while the pipeline is still processing the
error and getting to the basesrc for the second error.
Mark the second error as 'sometimes=true', as it's not critical to the test.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9118>
Actions are no longer always called directly from the main loop
so we can't depend on G_SOURCE_REMOVE to remove it. It's also
possible while returning up the call chain after running stop
to try to re-add it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9118>
Colorimetry bt2100-pq is a variant of colorspace V4L2_COLORSPACE_BT2020
where the transfer function is SMPTE 2084 which is used by HDR content.
Colorimetry 1:4:5:3 is a full-range variant of colorspace
V4L2_COLORSPACE_470_SYSTEM_BG
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9166>
Android 10 (API 29) added support for isHardwareAccelerated() to
MediaCodecInfo to detect whether a particular MediaCodec is backed by
hardware or not. We can now use that to ensure that the video hw-codec
is PRIMARY+1 on Android, since using a software codec for video is
simply not feasible most of the time.
If we're not able to detect isHardwareAccelerated(), perhaps because
the Android API version is too old, we try to use the codec name as
a fallback.
Also rank PRIMARY+1 the c2.android c2.exynos and c2.amlogic audio
codecs alongside OMX.google, because they are known-good.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9225>
Most of the messages can be printed with INFO threshold since they are
only printed on plugin registration.
Fix printing of codec caps, since GST_PTR_FORMAT truncates the output
in almost every case that I saw.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9225>
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/4317
This patch fixes a regression in
a35bf1e3847351703542869755b6b9e536b9d2fd.
The new calls to tracers in the affected change were being done while
holding the pad object lock, which wasn't the case in the old ones,
leading to the latency tracer deadlocking in gst_object_get_parent().
The dependency on the pad lock for those calls was accidental. This
patch removes it by temporarily unlocking during the affected calls,
not unlike how it's done when calling a probe callback or the
send_event() function of the downstream element.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8925>
When there are gaps between segments in the DASH manifest (e.g. it's a
recording of a live stream during which the camera was turned on and off
a couple times), seeking to a timestamp that belongs into a gap leads to
repeat_index calculation yielding a nonsensical negative number (because
ts < segment->start).
In such event set the playback to continue at the first repetition of
the segment that follows after the gap.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8620>
A user-supplied window handle (external_view) becomes the superView
of internal_view, which is closed with [view removeFromSuperview].
This fails silently if external_view = NULL (no handle supplied).
Call [win_internal_id close] in this case. Fixes#4432.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9049>
The copy of the exact same stream-start event prevents the multiqueue's sink
event function from being called because it is already stored on both pads at
link time
The text streams are no longer considered sparse by the multiqueue, so
interleave calculation is broken and makes us consume a lot of ram and we can
end up killed by the kernel because of this
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8412>
We never implemented correctly the selection of memories in a buffer given the
plane.
This patch uses gst_buffer_find_memory() for that. The offset is checked via the
video meta either in the input and output buffers, or the default offset given
the format and size.
This patch also requests the video meta option for the output buffers.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9219>
This patch adds the possibility to attach video metas to allocated buffers if
API users request that option.
The main reason to add this meta is for the future usage of
gst_buffer_find_memory(), where the requested offset determines the memory
index.
In the case of multi-memory Vulkan images the offset of every component is
larger than the default offset determined by the format and frame size, because
of the buffer minimum size and memory alignment determined by the used driver.
Then, the offset is set in the buffer video meta.
In the case of single memory Vulkan images, the default offset is set in the
video meta.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9219>
It is entirely possible that the in progress may still provide some state
updates until the ICE object is destroyed, these state updates should
not really be done when webrtcbin is in the process of destroying itself
and access freed data.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9216>
The fact that static-files followed the (nondeterministic) file system
order is likely what caused the dots-viewer executable not to be binary
reproducible. Enabling this feature that was added upstream should fix
it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9208>
When producing the profile_teir_level() struct, account for emulation
prevention bytes (0x0, 0x0, 0x3) in the source SPS. Also copy from the correct
starting point in the source SPS and don't put the NAL header into codec_data.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9205>
This proved to be helpful for previous protocol experiments, so let's
upstream it. Inspired by the corresponding code in Weston.
Protocols need to be placed in a `protocols` subdirectory and can be
declared in the following way in `meson.build`:
```
['color-management-v1', 'internal' ],
```
Note the `v1` being part of the name.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9186>
When creating a faststart file, qtmux has to take into account that
any 32 bit stco atom may need to be converted into its 64 bit version
due to the offset that is added before the mdat data by the moov atom,
the extra atoms and the mdat header. If a stco atom is converted into
co64, the chunk offset of all stco and co64 atoms has to be increased,
which in turn may also cause that other stco atoms need to be converted
into co64, modifying the chunk offsets again.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9102>
Using the Wayland color-management and color-representation protocols.
The implementation queries supported values from the compositors and tries
to convert them into GstVideoColorimetry values. It currently *does not*
pass these upstream to decoders etc. as GstCaps for negotiation.
On the Wayland side it uses named transfer functions, named primaries,
matrices and ranges. The straight alpha mode is also set if supported
by the compositor.
On setting caps it translates the GstVideoColorimetry from the GstVideoInfo
back to into a Wayland parametric image description and color representation
for the video surface if possible. If a colorimetry is not fully
support, we bail out and if wayland objects already exist they get reset or
deleted.
Note that not all GstVideoColorimetry values are implemented yet.
Useful debug options: GST_DEBUG=wlwindow:4,wldisplay:4
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6830>
Merge request !5651 introduced a regression for uploading VulkanBuffer. It can
be verified by
gst-launch-1.0 videotestsrc ! video/x-raw, format=NV12 ! vulkanupload ! \
video/x-raw\(memory:VulkanBuffer\) ! fakesink
Nobody has complained about this regression, so we assume it's a very niche
use-case.
This patch solve the regression by instantiating a different buffer pool
depending on the output cap features, and configuring it accordingly.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9218>
Change so that the handling of NAL unit that can't be parsed when using
AU alignment is the same as when using NAL alignment, ie drop the data
if it can't be parsed.
If the AU contains more than one NAL unit any correctly parsed NAL unit
in the AU is kept.
Fixes#4436
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8976>
If using NAL aligment and only dropping part of the AU, the size
argument given to gst_base_parse_finish_frame was wrong and this assert
in gst_base_parse_finish_frame hit
'gst_adapter_available (parse->priv->adapter) >= size' failed
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8976>
Certain encoder user controls as well as codec controls can be
changed while the pipeline is in playing state.
e.g. Codec controls like video_bitrate, frame_ltr_index,
use_ltr_frames, etc. and user controls like horizontal_flip,
vertical_flip can be changed while the pipeline is running.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9168>
caps.get_structure(0).get_name() used to work, but the usage of
StructureWrapper broke it but forcing to wrap it in a "with" statement.
We already have a Structure subclass, simply make it reference its
parent Gst.Caps object.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9148>
The LCEVC_ReceiveDecoderBase() and LCEVC_ReceiveDecoderPicture() APIs from the
LCEVCdec SDK can return LCEVC_Error if the enhancement data is wrong. This
change improves the lcevcdec element to check for those errors and stop the
pipeline when that happens.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9057>
The profile argument passed to gst_encode_base_bin_set_profile is now
transfer-full. This issue was noticed after commit
6beb709d43d2023e7e5dc8f1ee1323bc28c9d1d8 which fixed profile refcount handling
in transcodebin.
Driving-by, an encoding profile leak was also fixed in _set_profile, in case
it's called for an already active element.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9167>
When GST_DISABLE_GST_DEBUG is defined, the log context functions were
missing their stub implementations, causing link errors. Add the missing
stub functions to the disabled debug section.
Also add (nullable) annotation to gst_log_context_get_category since
it can return NULL when debugging is disabled.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9122>
Use gst_pad_push_event() instead of manually getting peer pad and calling
gst_pad_send_event() otherwise the pad probe on the sinkpad wouldn't get the
events.
Also ensure navigation events are posted to the bus even when gst_pad_push_event()
fails in the render function, so navigation remains functional regardless
of rendering issues.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9130>
compositor will record rendering commands using multiple threads
(i.e., blending commands are recoded using thread pool, and
background one is recorded on aggregate thread).
And there can be temporary refcount increase (so not writable).
Updates fence once all rendering commands have been submitted.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9157>
Adding an option to keep them no matter what.
Log files are often pretty large and keeping them around can be annoying,
usually people won't look at logs files for passing tests, and we do not
even print them out.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9127>
If the driver has size alignment requirement, suggest the
difference between aligned size required by diver and the actual
size of the frame as padding requirement to the upstream element.
This will ensure that the buffer size allocation is as per the
driver requirement.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8971>
When using the custom WebKitMediaSrc element (used by WebKit and able to
perform an initial seek in playbin), a stall caused by streamsynchronizer
was detected during an initial seek. The flow of events revealed that the
intertwining of the initial configuration of the streams with the reset
caused by the flush events from the seek left streamsynchronizer in an
inconsistent state:
streamsynchronizer0:sink_0 (video) events, starting before the seek:
stream-start --> Sets the stream to wait
flush-stop --> Clears the stream wait flag
caps
tag
segment
stream-collection
(buffers start to come and flow properly)
streamsynchronizer0:sink_1 (audio) events, happening after seek:
(no flush events, because the stream hadn't been initialized when the seek happened)
stream-start --> Sets the stream to wait
caps
segment
(stalled because the stream is in wait mode!)
The code in streamsynchronizer expects that all the streams are in wait
state before releasing all of them at once. The flush on the video stream
broke that assumption and that's why the audio stream is never released in
that scenario.
Avoiding the clearing of the wait flag on flush-stop isn't an actual solution
to the problem, as it creates other side effects and at least makes the
gst-editing-services/seek_with_stop test to timeout. The alternate solution
implemented in this patch consists on analyzing if the other streams different
from the one newly added (after the flush) aren't waiting (which would mean
that they've all been unlocked after all of them were waiting before), and,
in that case, mark the new stream as also not waiting.
A new test_stream_start_wait test case has been added to demonstrate this
problem. The test case creates a video stream, pushes a buffer, then
simulates a seek by pushing flush-start, flush-stop, stream-start and segment
events. Note that the flush-stop clears the video stream waiting flag.
After that, a new audio stream is created and stream-start and new segment
events are sent. Note that stream-start will set the audio stream to wait.
Then a buffer is pushed on each stream. In the failing case, the test hangs.
In the working case (after this fix), the test runs properly because the
fact of having seen a stream-start also helps to clear the wait flag.
A second new test_stream_start_wait_sparse test has also been added to prove
that this mechanism can also work with sparse streams (a special case of the
current stream-start handling code). This test behaves like the previous one,
but there's no video buffer after the seek (it'll come in the future, as the
stream is sparse, but actually never comes). The buffer after the seek in the
audio stream starts at its due time. Streamsynchronizer is able to ignore
the wait for the video stream and produce audio buffers on time.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4544>
Nowadays VK_EXT_debug_report is considered deprecated and it's recommended to
replace it it VK_EXT_debug_utils, which offer a way to ignore messages
considered false positives.
The approach is to try the extension first, if available at compilation time, if
not or if it fails to load, VK_EXT_debug_report fallbacks.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9119>
Convert buffer metadata getter/setter methods to Python properties for
more intuitive access. This makes the API more Pythonic by allowing
direct property assignment (buf.pts = 42) instead of method calls
(buf.set_pts(42)).
Updated properties: flags, pts, dts, duration, offset, offset_end
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9134>
Align the calculations for the number of samples per block with the
calculations in adpcmdec.
For MS ADPCM we have in adpcmdec:
samples = (blocksize - 7 * dec->channels) * 2 + 2 * dec->channels;
outsize = 2 * samples;
outbuf = gst_buffer_new_and_alloc (outsize);
This gives us the total output byte size in 16 bits samples. To get back
to the samples, dividing by the channels and 2, we get the right samples per
block as:
int spb = ((strf->blockalign / strf->channels) - 7) * 2 + 2;
Which we can then use to calculate the bitrate in riff-media.
A similar calculation for DVI ADPCM is needed to get the right bitrate
in all cases.
Tested with the sample in https://bugzilla.gnome.org/show_bug.cgi?id=636245
and another (failing before this patch) sample.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9082>
At the moment we are always accepting the input events to forward into
GStreamer infrastructure. This works but we might have other uses for
such events elsewhere in the QtQuick scene so allow opting out to this
behaviour.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9085>
After gst-validate-1.0 notices the first expected error on the bus, it will
queue the completion of that action while the pipeline is still processing the
error and getting to the basesrc for the second error.
Mark the second error as 'sometimes=true', as it's not critical to the test.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9060>
There is a small window during stopping when tasks have been add
to the transfer_requests queue, but never transfered to the
active_transfers array and causes the fallowing error:
GTask 0x7f2c4400e930 (source object: (nil), source tag: (nil)) finalized
without ever returning (using g_task_return_*()).
This potentially indicates a bug in the program.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9060>
- Unset stream ids if a collection does not contain them
- Automatically select a default stream of a type if the stream type is
enabled but no stream is selected yet when receiving the stream collection
- Warn if there's a collection update via streams-selected and if there are
unexpected streams being selected, or actually selected streams not being
found
- Improve debug output a bit
Among other things this also makes sure that we don't forget a selected stream
id when disabling a track so that when enabling it again later the same one can
be enabled again.
See https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/4344
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9081>
The part of the code that is commented with:
```
/* gst_validate_action_set_done() does not finish the action
* immediately. Instead, it posts a task to the main thread to do most
* of the work in _action_set_done().
*
* While the EOS handling lock guarantees that if an action had to call
* gst_validate_action_set_done() it has done so, it does not guarantee
* that _action_set_done() has been called.
*
* Is it possible that this handler is run before _action_set_done(), so
* we check at this point for actions that have a pending_set_done and
* call it before continuing. */
```
was not being executed in the case where the scenario was 'ignoring EOS'
while it was also required.
Also fix potential use after free in that specific code path.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9014>
Now that we have the monorepo, we always have a checkout
of gstreamer and we don't need to explicitly clone it again.
cbuild in ci-templates will always clone the repository in
/tmp/clone and we can use that to initialize the cache.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8994>
In the rare case when we get EOS on all pads before we get any buffer we
would finish the file without writing headers, i.e. the file would be
corrupt. This patch makes sure that a header is always written.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9061>
The build files had quite a few things wrong:
* Not using the method: kwarg, which can cause the wrong Qt to be
used for building
* There was no way to enable the build for them
* Qt was being detected multiple times, differently
* Unnecessary check for libGL
* have_cxx was being used incorrectly
* Qt tool detection was outdated
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9046>
Add a new API to control logging behavior, particularly for implementing
"log once" functionality and periodic logging. This helps avoid spamming
logs with repetitive messages.
The API provides:
- Static and dynamic context creation
- Configurable message identity calculation
- Periodic reset capability
- Context-aware logging macros
- Element message variants with context support
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6855>
As of Gst >= 1.24 we can just use GstVideoInfoDmaDrm APIs. Note
that SHM formats match DRM ones with only two exceptions.
No functional changes intended (for backporting) apart from
supporting a few more formats - those present in video-info-dma.c
but missing in the removed mapping.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8983>
A new element wrapping the LiteRT (aka TensorFlow Lite) inference engine.
It currently supports only CPU.
Co-authored-by: Daniel Morin <daniel.morin@collabora.com>
Co-authored-by: Denis Shimizu <denis.shimizu@collabora.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8523>
Add support to the ALSA device provider to enumerate PCM outputs that do
not correspond to a physical sound device i.e. they are "virtual" sinks,
like the plug, dmix, or softvol PCM outputs that can be setup in the ALSA
configuration files.
The main use-case for this is allowing usage of GstDeviceMonitor in setups
where there is no audio server and have custom ALSA audio configurations.
As those are likely to be uncommon, the feature is opt-in: a list of device
names and wildcard patterns separated by semicolons must be assigned to the
GST_ALSA_PCM_ALLOW environment variable before such PCM outputs will be
enumerated by the ALSA device provider. This allows either scanning all
PCM outputs, listing individual outputs, providing simple patterns with
'*' wildcards (which match only at the start or end of the name), or
a combination of them:
GST_ALSA_PCM_ALLOW=1 # Enable listing PCM outputs.
GST_ALSA_PCM_ALLOW='*' # Same, using a wildcard.
GST_ALSA_PCM_ALLOW='out_1;out_1' # Exact listing.
GST_ALSA_PCM_ALLOW='out_*' # Using a wildcard.
GST_ALSA_PCM_ALLOW='out_*;other_*;line_out' # Multiple items.
The main motivation for this patch is supporting enumeration of PCM outputs
in the WebKit GTK and WPE ports, which use GstDeviceMonitor to determine
which devices may be chosen for sound output. While on desktops typically
PulseAudio or PipeWire are used nowadays, on embedded devices it is often
desirable to avoid them and use custom configurations that perform audio
routing and processing using only ALSA.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8831>
Add handling for the selectable query as an element query,
on top of the existing pad query handling. This is useful
for uridecodebin when handling stream collection messages
before any adaptivedemux source pads have been exposed.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9002>
It was already parsed above in general for all video codecs. Just put the number
of fields into the JPEG-2000 in the specific field.
As a side effect this also actually checks if enough data is available.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8929>
templ is the new one that is being stored and that needs to be ref-sinked,
padtempl is the old one that just needs to be unreffed.
Fixes leaking the old template, and also makes sure that the new template is not
floating which can cause use-after-frees with bindings as they might wrongly
take ownership of a still floating template.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8988>
Stop testing DSD rates in gst_alsa_detect_dsd_rates() if the rate becomes zero
or negative. This avoids an infinite loop if gst_alsa_probe_supported_formats()
is used on a PCM sink defined like the following in the ALSA configuration file:
pcm.buggy {
type plug
slave.pcm "buggy_volume"
hint.description "Causes an infinite loop in GStreamer"
}
pcm.buggy_volume {
type softvol
slave.pcm "buggy_dmix"
control.name "buggy_volume"
}
pcm.buggy_dmix {
type dmix
ipc_key 12345
slave {
pcm "hw:0,0"
period_size 1024
buffer_size 4096
}
}
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8985>
So far we simply ignored it for MEMORY_DMABUF passthrough caps
without known negative cosequences, but with upcoming more complicated
caps negotiations it's becoming an issue, thus fix it.
Fixes: 7e71d4f753 ("gl: upload: Add DMA_DRM passthrough upload")
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8964>
It's not really possible to safely interpret the content afterwards if
it's missing.
Even for AVC3, the codec_data doesn't need to contain a SPS/PPS, but
it still needs to be present to tell downstream elements about the size
of the nal unit length field.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8955>
When the float python type is used inside a GstValueArray, it is
converted to a G_TYPE_DOUBLE GValue. Sometimes it is important to be
able to force G_TYPE_FLOAT GValue, for instance to set the the
"mix-matrix" property on audioconvert.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8957>
Without the codec_data, it's impossible to know the size of the field
for the number of NALu in a buffer. And since nal_length_size is unkown
the stream can't be parsed and payloaded and we risk an infinite loop.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8954>
This pool isn't reusing its buffers, which makes it pointless to enable
the cache
Holding an extra buffer in free queue can also lead to a deadlock when
the pool's max buffer count is configured low (commonly 2).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8939>
Gitlint returns error code 253 for a "wrong invocation" i.e. missing filename
for the commit message. That one signals a successful existing install.
Return a warning when either Git is not found (so no hooks will run) or Gitlint
failed its hook installation.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8764>
By design, unixfd is meant to be used for zero-copy and failing when the data is
not FD based memory is wanted to help debug pipelines. Though, there exists
cases, notably with RTP payloader and demuxers, where its not possible
to get all the data into FD memory through allocation queries.
To allow using unixfd for these cases, introduce a property on the unixfdsink
that enable copying the non FD data into freshly allocated memfd.
Co-authored-by: Nicolas Dufresne <nicolas.dufresne@collabora.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8861>
Make sure the pool recognizes DMA_DRM caps and parses them into
GstVideoInfo that has GstVideoFormat corresponding to "drm-format"
in the caps.
Fixes
gst_video_frame_map_id: assertion 'info->finfo->format ==
meta->format' failed
Where the left side of the assertion was GST_VIDEO_FORMAT_DMA_DRM and
the right side was some GstVideoFormat converted from DRM fourcc.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8913>
../meson.build:231: WARNING: DEPRECATED use of the `plugins` variable in gst-python.
../meson.build:232: WARNING: The variable should now be called `gst_plugins` and use:
../meson.build:233: WARNING: `declare_dependency( link_with: <plugin_target>,
variable: {'full_path': <plugin_target>.full_path()})` instead
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8753>
Wrap dependencies add a ton of warnings with the latest GCC in Fedora
42. Squelch them by specifying that these dependencies are not
a part of the gstreamer project, and should be treated as system deps.
libsoup needs some porting work for the bump, and vorbis/lame are
already at their latest releases.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8753>
Currently only a wpevideosrc2 element is exposed. GL and SHM buffer rendering
are supported, navigation events too (touch is un-tested). Audio pads handling
is not supported yet (that requires new WPE API).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8789>
Especially in force-live=true mode it was possible to produce buffers before the
element was set to PLAYING as long as a clock was available already.
This could easily lead to outputting buffers too early, and e.g. before the
correct base time is set and available.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8897>
In #8693 the issue was fixed for par > 1 without noticing that
it was also broken for par > 1.
Given that the natural width and height only changes when par != 1,
the logic is simplified to do:
* par_n < par_d -> the height is corrected
* par_n > par_d -> the width is corrected
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8828>
After READY -> NULL -> READY state change, the configured video
orientation didn't get applied on the new GstVaFilter instance.
Resettig prev_direction to default value in update_properties ensures
gst_va_filter_set_orientation() isn't inadvertently skipped.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8871>
Some frames are dequeued with error flag, which may cause AV unsync if decoder
does not drop them as soon as possible. So add "output-error-dequeued" and
"capture-error-dequeued" signal for v4l2 to drop such frames.
Fixes#3031
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5479>
Similar to NV12_10LE40, this is a 422 variant. This format is also named
NV20 (20bit per pixels) in other stack and is produced by rkvdec
decoder.
Co-authored-by: Sebastian Fricke <sebastian.fricke@collabora.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5612>
Make sure to always update next_cluster_offset, if next cluster offset
isn't known set it to zero. If next_cluster_offfset isn't updated it will
be the same as current and if the cluster parsing fails the same cluster
will be parsed again leading to duplication of the data in the cluster.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8855>
Work around Meson issues if this env var contains characters that
cannot be represented in the active code page (e.g. CP1252).
UnicodeDecodeError: 'charmap' codec can't decode byte 0x90 in position 865: character maps to <undefined>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8747>
When plugin_loader_load_and_sync returns false in plugin_loader_replay_pending,
the cur Glist l->pending_plugins will be added to the blacklist.
However, the l->pending_plugins might have already been loaded and freed in handle_rx_packet,
so causing a use-after-free issue.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8747>
The only thing that can be used in the chain function is the input segment. The
output segment might not be available at all yet or out of sync with the current
input segment.
Also because of that, the unadjusted timestamp has to be used for the
calculations as the adjustment is only part of the output segment.
This fixes the deadline calculation and the handling of force-keyunit events for
encoders using frame reordering (i.e. setting a minimum PTS).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8842>
Calculating intersection-of-union (IoU) is a very common operation used by
tensor-decoder handling tensors from vision models. Having this in a library
will improve maintainability and ease of writing tensor-decoder.
- Post-fix _uint: We might eventually want to handle different datatype that we
woule post-fix with _type
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8805>
Due to race condition it was previously possible that
gst_element_add_pad was not completed for each RTSP stream before
signal 'no-more-pads' was emitted.
Race condition explained:
Lets say two RTSP streams are created: Video and Audio.
1. Callback new_manager_pad is called for the Video stream =>
stream->added=TRUE.
all_added=FALSE because both streams are not yet added.
Call gst_element_add_pad and emit signal 'pad-added' for Video stream.
2. Callback new_manager_pad is called for Audio stream =>
stream->added=TRUE.
all_added=TRUE because both streams are added.
Call gst_element_add_pad and emit signal 'pad-added' for Audio stream.
3. Lets say gst_element_add_pad for the audio stream completes before
the video stream. Since the audio stream already has all_added==TRUE
this will result in the signal 'no-more-pads' to be emitted before
gst_element_add_pad for the video stream is completed.
Solution is to move the logic that sets added=True and checks if all
streams are added to after gst_element_add_pad. This will make sure
signal 'no-more-pads' is not emitted until all code in
gst_element_add_pad is completed for all streams.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8832>
There was a race condition where the demuxer would seek back to beginning after
determining the duration and while that seek was in progress one pad would
attempt to push a new buffer downstream, leading to a critical warning in
gst_pad_push().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8785>
Devcontainer [1] is a popular container schema that makes
IDEs and tooling aware about the container image you want
to use and develop against.
The most popular implementation of devcontainer are Visual
Studio and VSCode and this allows us to have a out of the
box pre-defined and working build environment that is very
close to the environment CI runs against.
This also allows VSCode on Windows and Mac to setup a
Linux docker container (vm) to develop against if so
desired.
[1] https://containers.dev/
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7816>
We install the rust toolchain, but then we have to manually
source it in CI since the headless containers skip over
the shell profiles that rustup typically uses.
Ideally we'd set these as variables in the main image, but
we don't have access to the buildah instance used in
ci-templates/cbuild.
However adding them to the toolbox image is good enough to
have the toolbox setup work ootb even if it doesn't call the
ci/scripts/source_image_env.sh script like the gitlab-ci jobs
will do.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7816>
Previously the script will give you a command like:
```
toolbox create gst-toolbox ..
```
Which is a bit redundant since we are already creating
toolbox, we don't need to label it as such.
Now instead use gst-$GST_UPSTREAM_BRANCH for the name,
so we will suggest gst-main and gst-1.24 instead
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7816>
driftsamples currently uses the requested skew directly, even if it
exceeds cexternal.
Use the approach that skew_slaving uses to fix this. As a side benefit,
this makes the custom_slaving and skew_slaving code easier to compare.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8605>
This reverts commit 8c017c79c5736c9e45e635df210e08550287646d.
1.22 was the correct pkg-config version. It's only the subproject
version that was wrong. Since we bumped libva.wrap to 2.22 version, h266
is now always available when using the subproject.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8800>
The element should allocate GstFdMemory large enough to fit incoming
memory's size plus its potential offset.
Fixes "gst_memory_resize: assertion 'size + mem->offset + offset <=
mem->maxsize' failed".
Fixes an issue reproducible on Raspberry Pi 4 that results in a garbled
image on the receiver's end:
gst-launch-1.0 libcamerasrc ! unixfdsink socket-path=/tmp/socket
gst-launch-1.0 unixfdsrc socket-path=/tmp/socket ! autovideosink
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8025>
This helps debug cases when the remote is offerer and m-line does not match with already existing transceivers.
In this case, it will create new ones with sendrecv direction without any warning.
Similar with code from _create_answer_task
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8735>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/allocators/gstfdmemory.c">Return a %GstMemory that wraps a generic file descriptor.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/allocators/gstfdmemory.c">the total size of the memory represented by @fd</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/allocators/gstfdmemory.c">the offset of valid data in the memory</doc>
<type name="gsize" c:type="gsize"/>
</parameter>
<parameter name="size" transfer-ownership="none">
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/allocators/gstfdmemory.c">the size of valid data in the memory</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/analytics/gstanalyticsbatchmeta.h">The intended use of this struct is that analytics elements read the buffer or
buffer list and potentially attach new #GstMeta with additional information.
Only one of @buffer or @buffer_list is going to be set, or both are %NULL.
Elements that need to modify @buffer (e.g. to attach a new meta) must first
call gst_buffer_make_writable() and store the writable buffer in this struct.
Similarly, a writable @buffer_list must be ensured with
gst_buffer_list_make_writable().
Modifying any other fields must be done with special care to ensure that the
data flow of the stream is not broken.
It is possible for @buffer and @buffer_list to be %NULL, e.g. if there was no
buffer for this batch but there were serialized events like a gap event.
Note that @serialized_events always contains all currently active serialized
events and not only the changed events compared to the previous buffer.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/analytics/gstanalyticsbatchmeta.h">All sticky #GstEvent that relate to this buffer</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/analytics/gstanalyticsbatchmeta.c">Gets the #GstCaps from a buffer</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/analytics/gstanalyticsbatchmeta.c">Gets the #GstSegment from a buffer</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/analytics/gstanalyticsbatchmeta.c">The #GstSegment if there is one</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/analytics/gstanalyticsbatchmeta.c">Gets the current stream id from a buffer</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/analytics/gstanalyticsbatchmeta.h">This meta represents a batch of buffers from one or more streams together
with the relevant events to be able to interpret the buffers and to be able
to reconstruct the original streams.
When used for multiple streams and batching them temporarily, caps of type
`multistream/x-analytics-batch(meta:GstAnalyticsBatchMeta)` should be used,
with the original caps of each stream in an array-typed `streams` field. The
original caps of each stream might be extended by additional fields and the
order of the streams in the array corresponds to the order of the @streams
array of the meta. In this case, empty buffers would be used without any
#GstMemory and
When used for a single stream, the original caps might be used together with
the `meta:GstAnalyticsBatchMeta` caps feature and potentially extended by
additional fields to describe the kind of batching and its configuration,
e.g. that each batch is made of 25% overlapping 320x320 slices of the
original video frame.
The timestamp, duration and other metadata of each batch can be retrieved
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/analytics/gstanalyticsbatchmeta.h">#GstAnalyticsBatchStream for this batch</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/analytics/gstanalyticsbatchmeta.h">Index of the stream in the meta's stream array</doc>
<type name="guint" c:type="guint"/>
</field>
<field name="buffers" writable="1">
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/analytics/gstanalyticsbatchmeta.h">#GstAnalyticsBatchBuffer in this batch for this stream</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/analytics/gstanalyticsbatchmeta.h">The caps feature to be used on streams that make use of this meta.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/analytics/gstanalyticsclassificationmtd.h">Handle containing data required to use gst_analytics_cls_mtd APIs. This type
is generally expected to be allocated on the stack.</doc>
@ -1302,7 +1468,10 @@ pixel to instance of an object is identified.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/analytics/gsttensor.c">Allocates a new #GstTensor of @dims_order ROW_MAJOR or COLUMN_MAJOR and
with an interleaved layout</doc>
with an interleaved layout.
For example, a two-dimensional tensor with 32 rows and 4 columns, @dims would
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/analytics/gsttensor.c">Validate the tensor whether it mathces the reading order, dimensions and the data type.
Validate whether the #GstBuffer has enough size to hold the tensor data.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/analytics/gsttensor.c">TRUE if the #GstTensor has the reading order from the memory matching @order,
dimensions matching @num_dims, data type matching @data_type
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/analytics/gsttensor.c">The order of the tensor to read from the memory</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/analytics/gsttensor.c">The number of dimensions that the tensor can have</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/analytics/gsttensor.c">An optional array of dimensions, where G_MAXSIZE means ANY.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/analytics/gsttensor.c">a constant string with the name of the data type</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/analytics/gsttensor.h">Describe the type of data contain in the tensor.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/analytics/gsttensor.h">A non-IEEE 8-bit floating point format with 4 exponent bits and 3 mantissa bits, with NaN and no infinite values (FN).
See [this paper for more details](https://onnx.ai/onnx/technical/float8.html)</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/analytics/gsttensor.h">A non-IEEE 8-bit floating point format with 4 exponent bits and 3 mantissa bits, with NaN, no infinite values (FN) and no negative zero (UZ).
See [this paper for more details](https://onnx.ai/onnx/technical/float8.html)</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/analytics/gsttensor.h">A non-IEEE 8-bit floating point format with 5 exponent bits and 2 mantissa bits.
See [this paper for more details](https://onnx.ai/onnx/technical/float8.html)</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/analytics/gsttensor.h">A non-IEEE 8-bit floating point format with 5 exponent bits and 2 mantissa bits, with NaN, no infinite values (FN) and no negative zero (UZ).
See [this paper for more details](https://onnx.ai/onnx/technical/float8.html)</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/analytics/gsttensor.h">Indicate to read tensor from memory in row-major or column-major order.</doc>
@ -1481,6 +1727,25 @@ smaller than #GstTensorMeta.num_tensors</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/analytics/gsttensormeta.c">Get the first tensor from the #GstTensorMeta identified by @id.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/analytics/gsttensormeta.c">Finds the first tensor with the requsted ID in the meta</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/analytics/gsttensormeta.c">Get the first tensor from the #GstTensorMeta identified by
@tensor_id, matching the reading order, dimensions and the data
type and optionally the dimensions. Validate whether the
#GstBuffer has enough size to hold the tensor data.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/analytics/gsttensormeta.c">The order of the tensor to read from the memory</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/analytics/gsttensormeta.c">The number of dimensions that the tensor can have</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/analytics/gsttensormeta.c">An optional array of dimensions, where G_MAXSIZE means ANY.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/analytics/gstanalyticsbatchmeta.c">Adds a #GstAnalyticsBatchMeta to a buffer or returns the existing one</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/analytics/gstanalyticsbatchmeta.c">The new #GstAnalyticsBatchMeta</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/analytics/gstanalyticsmeta.c">Attach a analysis-results-meta-relation meta (#GstAnalyticsRelationMeta)to @buffer.
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/analytics/gsttensormeta.c">Adds a #GstTensorMeta to a buffer or returns the existing one</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/analytics/gsttensormeta.c">Adds a #GstTensorMeta to a buffer</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/analytics/gstanalyticsbatchmeta.c">Gets the #GstAnalyticsBatchMeta from a buffer</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/analytics/gstanalyticsbatchmeta.c">The #GstAnalyticsBatchMeta if there is one</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/analytics/gstanalyticsmeta.c">Retrives the meta or %NULL if it doesn't exist</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/analytics/gstanalytics_image_util.c">Calculate the intersection over the union (IoU) of the two areas defined by
the bounding box 1 and bounding box 2. IoU is a measure of how much two
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/analytics/gstanalytics_image_util.c">Calculate the intersection over the union (IoU) of the two areas defined by
the bounding box 1 and bounding box 2. IoU is a measure of how much two
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/analytics/gstanalyticsmeta.c">Gets the string version of the name of this type of analytics data</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/analytics/gsttensor.c">a constant string with the name of the data type</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">This function blocks until a sample or an event or EOS becomes available or the appsink
element is set to the READY/NULL state or the timeout expires.
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">Get the number of currently queued buffers inside @appsink.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">Get the number of currently queued bytes inside @appsink.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">Get the amount of currently queued time inside @appsink.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">Check if @appsink will drop old buffers when the maximum amount of queued
data is reached (meaning max buffers, time or bytes limit, whichever is hit first).</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">%TRUE if @appsink is dropping old buffers when the queue is
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">Returns the currently set #GstAppLeakyType. See gst_app_sink_set_leaky_type()
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">Get the maximum amount of buffers that can be queued in @appsink.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">This function blocks until a sample or an event becomes available or the appsink
element is set to the READY/NULL state.
@ -593,9 +665,10 @@ you must check the caps on the samples to get the actual used caps.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">Instruct @appsink to drop old buffers when the maximum amount of queued
data is reached, that is, when any configured limit is hit (max-buffers, max-time or max-bytes).</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">When set to any other value than GST_APP_LEAKY_TYPE_NONE then the appsink
will drop any buffers that are pushed into it once its internal queue is
full. The selected type defines whether to drop the oldest or new
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">Set the maximum amount of buffers that can be queued in @appsink. After this
amount of buffers are queued in appsink, any more buffers will block upstream
@ -707,7 +800,7 @@ case new buffers will be discarded.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">This function blocks until a sample or an event or EOS becomes available or the appsink
element is set to the READY/NULL state or the timeout expires.
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">The number of currently queued buffers inside appsink.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">The number of currently queued bytes inside appsink.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">The amount of currently queued time inside appsink.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">Drop old buffers when the buffer queue is filled.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">When set to any other value than GST_APP_LEAKY_TYPE_NONE then the appsink
will drop any buffers that are pushed into it once its internal queue is
full. The selected type defines whether to drop the oldest or new
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">Maximum amount of buffers in the queue (0 = unlimited).</doc>
<type name="guint" c:type="guint"/>
@ -838,6 +960,14 @@ condition.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">Maximum total duration of data in the queue (0 = unlimited)</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">Wait for all buffers to be processed after receiving an EOS.
@ -967,7 +1097,7 @@ can pull samples at its own rate.
Note that when the application does not pull samples fast enough, the
queued samples could consume a lot of memory, especially when dealing with
raw video frames. It's possible to control the behaviour of the queue with
the "drop" and "max-buffers" / "max-bytes" / "max-time" set of properties.
the "leaky-type" and "max-buffers" / "max-bytes" / "max-time" set of properties.
If an EOS event was received before any buffers, this function returns
%NULL. Use gst_app_sink_is_eos () to check for the EOS condition.</doc>
@ -988,7 +1118,7 @@ Events can be pulled when the appsink is in the READY, PAUSED or PLAYING state.
Note that when the application does not pull samples fast enough, the
queued samples could consume a lot of memory, especially when dealing with
raw video frames. It's possible to control the behaviour of the queue with
the "drop" and "max-buffers" / "max-bytes" / "max-time" set of properties.
the "leaky-type" and "max-buffers" / "max-bytes" / "max-time" set of properties.
This function will only pull serialized events, excluding
the EOS event for which this functions returns
@ -1053,7 +1183,7 @@ can pull samples at its own rate.
Note that when the application does not pull samples fast enough, the
queued samples could consume a lot of memory, especially when dealing with
raw video frames. It's possible to control the behaviour of the queue with
the "drop" and "max-buffers" / "max-bytes" / "max-time" set of properties.
the "leaky-type" and "max-buffers" / "max-bytes" / "max-time" set of properties.
If an EOS event was received before any buffers or the timeout expires,
this function returns %NULL. Use gst_app_sink_is_eos () to check
@ -2076,6 +2206,10 @@ on outgoing buffers.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">The amount of currently queued time inside appsrc.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">The total duration in nanoseconds of the data stream. If the total duration is known, it
is recommended to configure it with this property.</doc>
@ -2103,6 +2237,10 @@ GstAppSrc::format should be time. However, possibly #GstAppSrc can support
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">Instruct the source to behave like a live source. This includes that it
will only push out buffers in the PLAYING state.</doc>
@ -2146,6 +2284,14 @@ latency calculations of #GstBaseSrc.</doc>
queue drops below this percentage of max-bytes.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">The total size in bytes of the data stream. If the total size is known, it
is recommended to configure it with this property.</doc>
<doc xml:space="preserve" filename="../subprojects/gstreamer/libs/gst/base/gstaggregator.c">The number of currently queued buffers inside this pad</doc>
<doc xml:space="preserve" filename="../subprojects/gstreamer/libs/gst/base/gstaggregator.c">Enables the emission of signals such as #GstAggregatorPad::buffer-consumed</doc>
<type name="gboolean" c:type="gboolean"/>
@ -2604,6 +2616,9 @@ the first buffer that is received.</doc>
<doc xml:space="preserve" filename="../subprojects/gstreamer/libs/gst/base/gstaggregator.h">Start at the running time
<doc xml:space="preserve" filename="../subprojects/gstreamer/libs/gst/base/gstaggregator.h">Start at the current running time when reaching %GST_STATE_PLAYING.</doc>
<doc xml:space="preserve" filename="../subprojects/gstreamer/libs/gst/base/gstbaseparse.c">If set to %TRUE, baseparse will unconditionally force parsing of the
incoming data. This can be required in the rare cases where the incoming
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/gl/gstglutils.h">EGL is responsible for the colorspace conversion. In this case, all
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/gl/gstglfilter.c">#GstGLFilter helps to implement simple OpenGL filter elements taking a
@ -8608,6 +8625,19 @@ require this to be called with a valid handle before drawing can commence.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/gl/gstglwindow.c">whether an visible output surface has been requested</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/gl/gstglwindow.c">Configure whether a visible output surface is requested.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/gl/gstglwindow.c">Sets the resize callback called every time a resize of the window occurs.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/gl/gstglcolorconvert.h">The currently supported formats that can be converted</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/gl/gstglmemory.h">List of video formats that are supported by #GstGLMemory</doc>
@ -807,7 +807,7 @@ the tags after the life-time of @info, you will need to copy them.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/pbutils/gstdiscoverer.c">Parses a #GVariant as produced by gst_discoverer_info_to_variant()
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/pbutils/gstpluginsbaseversion.h">The micro version of GStreamer's gst-plugins-base libraries at compile time.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/sdp/gstsdpmessage.c">a #GstStructure belonging to a #GstCaps SDP mapping, see also
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/sdp/gstsdpmessage.c">Initialize @media so that its contents are as if it was freshly allocated
with gst_sdp_media_new(). This function is mostly used to initialize a media
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/sdp/gstsdpmessage.c">a #GstStructure belonging to a #GstCaps SDP mapping, see also
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/sdp/gstsdpmessage.c">Initialize @media so that its contents are as if it was freshly allocated
with gst_sdp_media_new(). This function is mostly used to initialize a media
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/transcoder/gsttranscoder.c">(transfer full): A GstStructure containing extra details about the error</doc>
@ -494,9 +494,9 @@ See also #gst_transcoder_get_message_bus()</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/transcoder/gsttranscoder.c">(transfer full): A GstStructure containing extra details about the warning</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/transcoder/gsttranscoder.c">(transfer full): A GstStructure containing extra details about the error</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/transcoder/gsttranscoder.c">(transfer full): A GstStructure containing extra details about the warning</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/video/video-format.h">Number of video formats in #GstVideoFormat.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/video/video-format.h">10-bit grayscale, packed into 16bit words (6 bits left padding)</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/video/video-format.c">Converts a FOURCC value into the corresponding #GstVideoFormat.
If the FOURCC cannot be represented by #GstVideoFormat,
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/vulkan/gstvkfullscreenquad.c">@cmd must be locked with gst_vulkan_command_buffer_lock().</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/vulkan/gstvkfullscreenquad.c">whether @cmd could be filled with the necessary commands</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/vulkan/gstvkvideoutils.c">whether @a and @b contains the same information.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/vulkan/gstvkvideoutils.c">%TRUE if @profile is correct and matches with @codec</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/vulkan/gstvkvideoutils.c">VkVideoCodecOperationFlagBitsKHR described by @profile</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/vulkan/gstvkvideoutils.c">#GstVulkanVideoProfile to convert into a #GstCaps</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/vulkan/gstvkvideoutils.c">%TRUE if @caps was parsed correctly, otherwise %FALSE</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/vulkan/gstvkwindow.h">#GstVulkanWindow is an opaque struct and should only be accessed through the
provided api.</doc>
@ -7916,27 +7754,6 @@ associated #GstVulkanFence is signalled</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/vulkan/gstvkvideoutils.c">%TRUE if @caps was parsed correctly, otherwise %FALSE</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/webrtc/datachannel.c">Send @data as a data message over @channel.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/webrtc/datachannel.c">Send @str as a string message over @channel.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/webrtc/ice.h">A string containing the address of the candidate. This value may be
an IPv4 address, an IPv6 address, or a fully-qualified domain name</doc>
<type name="utf8" c:type="gchar*"/>
</field>
<field name="port" writable="1">
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/webrtc/ice.h">The network port number used by the candidate</doc>
<type name="guint" c:type="guint"/>
</field>
<field name="stream_id" writable="1">
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/webrtc/ice.h">A string that uniquely identifies the object that is being
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/webrtc/ice.h">A string specifying the protocol (tcp or udp) used to transmit data
on the @port</doc>
<type name="utf8" c:type="const gchar*"/>
</field>
<field name="relay_proto" writable="1">
<type name="utf8" c:type="const gchar*"/>
</field>
<field name="prio" writable="1">
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/webrtc/ice.h">The candidate's priority, corresponding to RTCIceCandidate.priority</doc>
<type name="guint" c:type="guint"/>
</field>
<field name="url" writable="1">
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/webrtc/ice.h">For local candidates, the url property is the URL of the ICE server
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/webrtc/webrtc_fwd.h">An "active" TCP candidate is one for which the transport
will attempt to open an outbound connection but will not
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/webrtc/webrtc_fwd.h">A "passive" TCP candidate is one for which the transport
will receive incoming connection attempts but not attempt
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/webrtc/webrtc_fwd.h">An "so" candidate is one for which the transport will attempt
to open a connection simultaneously with its peer.</doc>
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