The pool currently defaults to performing a layout transition to
VK_IMAGE_LAYOUT_TRANSFER_DST_OPTIMAL, with some special exceptions for
video usages. This may not be a legal transition depending on the usage.
Provide an API to explicitly control the initial image layout.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5881>
When we finish a frame, we pass a size which semantic can easily be confused.
Improve the documentation to clarify that the parameter size is the amount of
input data being consumed and, if set, the output_buffer size can differ.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5754>
VA drivers allocate surfaces given their properties, so there's no need to
provide a buffer size to the VA pool.
Though, the buffer size is provided by the driver, or the canonical size
is used for single planed surfaces.
This patch removes the need to provide a size for the function
gst_va_pool_new_with_config() and adds a helper method to retrieve the surface
size, gst_va_pool_get_buffer_size(). Also change the callers accordingly.
Changes for custom VA pool creation will be addressed in the following commits.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5805>
- GstAnalyticRelationMeta is a base class for analytics
meta. It's able to store analytics results (GstAnalyticRelatableMtd)
and describe the relation between each analysis results.
- GstAnalysisRelationMeta also contain an algorithm able to explore
analysis results relation using a bfs.
- Relation(edge) between analysis results (vertice) are stored in an adjacency-matrix
that allow to quickly identify if two analysis results are related and by
which relation they related. It also work for indirect relation
and can provide the path of analysis results by which two
analysis results are related.
- One allocation per buffer to store analysis results. Here we rely on
the application to guess how much space will be required to store all
analysis results. This is something that could be improved
significantly but it's a starting point.
- Define common analysis results, classification, object-detection,
tracking that are subclass of GstAnalyticRelatableMtd. The also
provide exemple of how to extend GstAnalyticRelatableMtd to have them
benefit for the mechanim to express relation with other analysis
results.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4962>
Instead of guessing the DRM format and modifier, pass a DRM video info to
gst_va_dmabuf_memories_setup().
Still, it checks for the DRM parameters in DRM info, if they are not available,
as in the case of V4L2 buffers, the part of the video info is used.
This is an API breakage, but since the plugin is still in stage, it's still
allowed.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5264>
Formatters might call "loaded" from the `gessrc` streaming thread
meaning that the `->formatters` field need to be protected.
Several other APIs are called from gesbasedemux, in some radom
thread, so we should ensure that this is all MT. safe, and the API
makes it simple.
Co-authored-by: Philippe Normand <philn@igalia.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5431>
Adds list of formats that should be used by element in needs to passthrough
video. It contains the full list of video format plus DMA_DRM format
and will be extended in the future as needed. This patches includes 3 new
symbols:
- GST_VIDEO_FORMATS_ANY_STR
- GST_VIDEO_FORMATS_ANY
- gst_video_formats_any()
The last one can be used by bindings or for code that prefers having
GstVideoFormat values instead of strings.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5386>
While the suspend modes NONE and PAUSED provided a low startup latency
for connecting clients they did not ensure that streams started on
fresh data.
With this property we can maintain the low startup latency of those
suspend modes while also ensuring that a stream starts on a key unit.
Furthermore, by modifying the value of a new property,
ensure-keyunit-on-start-timeout, it is possible to accept a keyunit of
a certain age but discard it if too much time has passed and instead
force a new keyunit.
Fixes#2443
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4334>
Move the GstStructure field into public struct for direct access, that's
easier than having to call a function to get it. It is not an API/ABI
breakage to extend the public structure of a GstMeta because they are
always allocated by inside GStreamer. The structure is exposed already
by gst_custom_meta_get_structure() which does not return a copy/ref, so
it is locked into holding a GstStructure forever anyway.
Also add gst_meta_register_custom_simple() because most of the time only
a name is required, tags and transform functions are more niche
use-case.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5385>
Fixes a potential GPU stall if an immediately freed texture/buffer is
attempted to be reused immediately by the CPU, e.g. when uploading.
Problematic scenario is this:
1. element does GPU processing reading from texture
2. frees the buffer back to the pool
3. pool acquire returns the just released buffer
4. GPU processing then has to wait for the previous GPU operation to
complete causing a stall
If there was a reliable way to know whether a buffer had been finished
with across all GPU drivers, we would use it. However as that does not
exist, this workaround is to keep the released buffer unusable until the
next released buffer.
This is the same approach as is used in the qml (Qt5) elements.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5144>
Add gst_audio_ring_buffer_set_errored() that will mark the
ringbuffer as errored only if it is currently started or paused,
so gst_audio_ringbuffer_stop() can be sure that the error
state means that the ringbuffer was started and needs stop called.
Fixes a crash with osxaudiosrc if the source element posts
an error, because the ringbuffer would not get stopped and CoreAudio
would continue trying to do callbacks.
Also, anywhere that modifies the ringbuffer state, make sure to
use atomic operations, to guarantee their visibility
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5205>
Section 3.4 in RFC8835 states that if a WebRTC endpoint uses an HTTP
proxy to access the Internet it MUST include the "ALPN" header. This
commit adds this header.
By default the ALPN used when connecting to the TURN/TCP server via a
proxy is set to "webrtc". It can be changed by adding an alpn url
option for the http-proxy. For example:
http://user:pass@my.http.proxy.com:8080?alpn=c-webrtc
This will add the header "ALPN: c-webrtc" to the HTTP proxy CONNECT
request.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4212>
Adds gst_queue_array_sort for sorting and gst_queue_array_push_sorted{,struct} for pushing in a sorted order.
All three functions accept a comparison GCompareDataFunc along with optional user_data to pass to it.
In gst_queue_array_sort a small workaround was needed to correctly sort non-struct arrays. Like what _find() already
does, we need to dereference our pointers first, to make sure we can use the same comparison functions everywhere.
This is done via a small wrapper around the provided comparison function.
The array can also wrap around (tail ends up 'before' the head), in which case we have to reorder the array (similar to
what do_expand() does) to then be able to use an existing sorting function, like g_qsort_with_data().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5112>
If a depayloader aggregates multiple RTP buffers into one buffer only
the last RTP buffer was checked for header extensions. Now the
depayloader remembers all RTP packets pushed before a output buffer is
pushed and checks all RTP buffers for header extensions.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4979>
These 10bit formats are identical to NV12_16L32S, but 64bytes of data is being
prefixed with 16bytes data with four pixels of lower 2bits per byte. For
MT2110T, the lower two bits set so each bytes contains a column of 4 pixels,
also describe as tiled lower 2 bits. MT2110T has been chosen as a name to match
the vendor chosen name. This format is unlikely to exist for other vendors.
For MT2110R, the 2 low bits are in raster order.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3444>
The current way of dma caps uses the drm-format to replace the orginal
format field. The absence of format field means it can accept all formats.
It causes problems when clipping with other old DMA or video/x-raw(ANY)
caps, the result will contain both format field and drm-format field,
which is not valid DMA caps.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4981>
This GST_VIDEO_FORMAT_DMA_DRM is introduced for DMABuf kind feature
usage. It represent the DMA DRM kind memory. And like the ENCODED
format, it should not be interpreted and mapped as normal video format.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4981>
Adding GST_CUDA_CRITICAL_ERRORS env variable so that program can be
terminated on unrecoverable error.
Example)
GST_CUDA_CRITICAL_ERRORS=2,700 gst-launch-1.0 ...
In this example, CUDA_ERROR_OUT_OF_MEMORY(2) and
CUDA_ERROR_ILLEGAL_ADDRESS(700) are registered as critical error
and program will be aborted on those errors
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4729>
Previously it was possible that a shared media was just in the process
of being unprepared because the last client disappeared, while another
client retrieved it from the cache and then tried to use it. Unless the
media was reusable this would've then failed unnecessarily.
To avoid this it is necessary to lock the media directly in
gst_rtsp_media_factory_construct() and return a locked media. After
locking the cached media it is necessary to check if the media was ever
unprepared or is actually reusable and based on that either reuse it or
create a new media.
This minimally changes the gst_rtsp_media_factory_construct() API to
always return a locked media, and adds a new
gst_rtsp_media_can_be_shared() function to check if a media can actually
be shared in practice.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4606>
check_version(1.23.1) would return TRUE for a git development version
like 1.23.0.1, which is quite confusing and somewhat unexpected.
We fixed this up in the version check macros already in !2501, so this
updates the run-time check accordingly as well.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4513>
Allowing better control over the way discovery happens and allowing
us to expose a proper API.
This also adds the potential of implementing more multi-threaded
discovery in a clean way in the future.
This allows us to cleanly expose the new
GstDiscoverer::load-serialize-info signal.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3911>
