Use of VK_PIPELINE_STAGE_2_ALL_COMMANDS_BIT instead of
specific VK_PIPELINE_STAGE_2_VIDEO_DECODE_BIT_KHR
Fix for VUID-vkCmdPipelineBarrier2-srcStageMask-03849
pDependencyInfo->pImageMemoryBarriers[0].srcStageMask
(VK_PIPELINE_STAGE_2_VIDEO_DECODE_BIT_KHR) is not compatible with
the queue family properties
(VK_QUEUE_GRAPHICS_BIT|VK_QUEUE_COMPUTE_BIT|VK_QUEUE_TRANSFER_BIT|
VK_QUEUE_SPARSE_BINDING_BIT|VK_QUEUE_PROTECTED_BIT) of this
command buffer. The Vulkan spec states: The srcStageMask member
of any element of the pMemoryBarriers, pBufferMemoryBarriers, or
pImageMemoryBarriers members of pDependencyInfo must only
include pipeline stages valid for the queue family that was
used to create the command pool that commandBuffer was allocated
from (
https://www.khronos.org/registry/vulkan/specs/1.3-extensions/
html/vkspec.html#VUID-vkCmdPipelineBarrier2-srcStageMask-03849)
The frame barrier should use a compatible srcStageMask for all
the queues.
Remove reset_pipeline_stage_mask as it is redundant
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6780>
Add pitch tests with different forward and backward playback rates.
Those tests depend on the libSoundTouch version to validate the buffers
checksums. The actual version uses libSoundTouch 2.3.2, use the
`--force-fallback-for=soundtouch` meson option to build using the same
version.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6247>
- fully protect accesses to the libsoundtouch API that is not
thread-safe.
- fully protect accesses to GstPitch members that could be read by a
downstream query thread while written by an upstream streaming thread
or a user thread.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6247>
- use the `GST_PITCH_GET_PRIVATE` accessor when needed
- rename `out_seg_rate` to `output_rate` to use the same name as the parameter
- rename `seg_arate` to `segment_applied_rate` to improve readability
- apply gst-indent to gstpitch.hh/cc
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6247>
When changing playing rate, the output segment was not correctly
calculated because the stream time ratio was computed using the previous
input segment rate instead of using the actual rate. This was producing
wrong results for the output segment start and end values.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6247>
The h26xdecoder 'stop' method was not called
as the vulkan h26x class rewires the video decoder
'stop' base method to its own one.
It was causing some memory leaks such as dangling parser
and dpb in h26xdecoder base class.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6782>
Since we don't want to expose video decoding API outside of GStreamer, the
header is removed from installation and both source files are renamed as
-private.
The header must remain in gst-libs because is referred by GstVulkanQueue,
which's the decoder factory.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6723>
`rsvg_handle_get_dimensions()` and `rsvg_handle_render_cairo()` are
deprecated, and the replacement librsvg functions as specified in the
migration guide are `rsvg_handle_get_intrinsic_size_in_pixels()` and
`rsvg_handle_render_document()`.
However, those are not drop-in replacements, and actually have
breaking semantics for our use-case:
1. `intrinsic_size_in_pixels()` requires SVGs to have width+height or
the viewBox attribute, but `get_dimensions()` does not. It will
calculate the geometry based on element extents recursively.
2. `render_cairo()` simply renders the SVG at its intrinsic size on
the specified surface starting at the top-left, maintaining
whatever transformations have been applied to the cairo surface,
including distorted aspect ratio.
However, `render_document()` does not do that, it is specifically
for rendering at the specified aspect ratio inside the specified
viewport, and if you specify a viewPort that does not match the
aspect ratio of the SVG, librsvg will center it.
Matching the old behaviour with the new APIs is a lot of work for no
benefit. We'd be duplicating code that is already there in librsvg in
one case and undoing work that librsvg is doing in the other case.
The aspect ratio handling in this element is also kinda atrocious.
There is no option to scale the SVG while maintaining the aspect
ratio. Overall, element needs a rewrite.
Let's just disable deprecations. The API is not going anywhere.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6726>
If propose_allocation comes before set_caps, self->video_info
has not been extracted from caps and self->video_info.size is 0.
It causes buffer pool fail to set config . So need to use info
size got from query instead when propose_allocation.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6666>
When using v2.0.2 of the subproject, it triggers werror for
unused functions that come from the fdkaac headers.
This avoids errors like the following when werror is set.
```
subprojects/fdk-aac-2.0.2/fdk-aac/FDK_audio.h:757:29: error: ‘FDKlibInfo_lookup’
defined but not used [-Werror=unused-function]
757 | static FDK_AUDIO_INLINE INT FDKlibInfo_lookup(const LIB_INFO* info,
```
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6611>
There was a potential busy loop occuring because when we were taking
data from the internal ccbuffer, we were not resetting which field had
written data. This would mean that the next time data was retrieved
from ccbuffer, it was always from field 0 and never from field 1.
This only affects usage of cc_buffer_take_separated() which is only used
by cdp->raw cea608.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6423>
Cea608 (valid) padding removal is available on the input side of ccconverter
or configurable on cccombiner. cccombiner can now configure whether
valid or invalid cea608 padding is used and for valid padding, how long
after valid non-padding to keep sending valid padding.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6300>
The transport stream only returned the CAPS for the first matching PT entry
from the `ptmap`. Other SSRC with the same PT where not included. For a stream
which bundled multiple audio streams for instance, only the first SSRC was
knowed to the SSRC demux and downstream elements.
This commit adds all the `ssrc-` attributes from the matching PT entries.
The RTP jitter buffer can now find the CNAME corresponding its SSRC even if it
was not the first to be registered for a particular PT.
The RTP PT demux removes `ssrc-*` attributes cooresponding to other SSRCs
before pushing SSRC specific CAPS to downstream elements.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6119>
Syncrhonizing buffer commits to the streaming thread can lead to
dropped frames when frame callbacks are not processed before the
next frame is ready for rendering. Depending on the drift between
the wayland compositor and buffer source timings, this can lead to
periods of significant frame drop, especially when the media frame
rate is close to the display frame rate.
Cache buffers in the streaming thread and peform commits on the
display thread to eliminate the buffer commit racing.
The implementation is the same for both waylandsink and gtkwaylandsink,
so move it to the common wayland library under gst-lib.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6133>
Some subtitle "decoders" had a wrong category of "Parser", which `parsebin`
relies on to identify elements which do not *decode* streams but *parse* them.
This would cause such subtitle decoders to be plugged in within parsebin,
preventing the original stream to be properly used by (more efficient)
downstream decoders or subtitle renderers.
Fixes#1757
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6153>
In rtpbin we already systematically check for all property names
except latency, correct that.
In webrtcbin we need to check before trying to use the do-retransmission
property.
This is useful for the case where an element like identity gets passed
to rtpbin's request-jitterbuffer property, when the application wants
to use webrtcbin in an SFU situation, with no reordering and no added
latency
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6112>
* Bump the rank of the musepack v7/v8 FFmpeg demuxers to SECONDARY
* Bump the rank of the musepack v7/v8 FFmpeg audio decoders to SECONDARY
* Demote the rank of the musepackdec element to MARGINAL
This is for two reasons:
* The musepack library is no longer maintained, whereas the FFmpeg
implementation can/will receive fixes
* The `musepackdec` implementation was a all-in-one "parsing and decoding" blob
which doesn't play nicely with decodebin3 and others
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3033
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6074>
Sends a gap event if nothing to output for a given input buffer.
For example, an input buffer might not contain any caption data
for downstream requested field, then we need to inform downstream
of the case.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6073>
WebKit commit b12e7ed2ad3a ("[WPE] Upstream the new WPE platform API
https://bugs.webkit.org/show_bug.cgi?id=265286")[1] added a `WPEView` typedef
which clashes with our `WPEView` class.
Rename the `WPEView` class to `GstWPEThreadedView` to avoid the collision.
Also prefix the `WPEContextThread` class with `Gst` and rename the
source files to reflect the new class name and use lowercase while at it
for consistency
[1] b12e7ed2ad
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6065>
