In some cases you want to ensure that a specific element factory is used
while requiring some specific caps but this was not possible. You can
now do `qtmux:video/x-prores,variant=standard|factory-name=avenc_prores_ks`
to ensure that the `avenc_prores_ks` factory is used to produce the
'standard' variant of prores video stream.
This also enhances a bit the documentation
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6875>
The encoder can specify the a preferred_output_delay value to get better throughput
performance. The higher delay may get better HW performance, but it may increases
the encoder and pipeline latency.
When the output queue length is smaller than preferred_output_delay, the encoder
will not block to waiting for the encoding output. It will continue to prepare and
send more commands to GPU, which may improve the encoder throughput performance.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4359>
This counter is incremented once for every segment, meaning it would
e.g. overflow after 24 days when using 1ms segments. Once that happens,
completely wrong positions are reported and invalid memory is handed out
for writing/reading the next segments.
As the affected variables are unfortunately part of the public API of
the struct, a second set of variables is added together with accessor
functions and both variables are kept in sync for backwards
compatibility.
All existing users of the two variables are moved to the new ones but
external code might still run into the overflow.
This also slightly breaks API as external code updating the variables
will have no effect anymore but the only known user of this is
pulsesink.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6740>
If the muxer times out because of the latency deadline it can happen
that some pads have no caps yet. In that case skip creation of streams
for these pads and create updated section tables once the first buffer
arrives later.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6823>
This makes sure that for sparse streams (KLV, DVB subtitles, ...) the
muxer does not wait until the next buffer is available for them but
times out on the latency deadline and outputs data.
For non-live pipelines it will still be necessary for upstream to
correctly produce gap events for sparse streams.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6823>
* gst_ffmpegviddec_frame() is the only caller of gst_ffmpegviddec_video_frame()
and has the same signature. Just move the checks into a single function and
use that.
* Make it clear which frames are the input and output ones in
gst_ffmpegviddec_video_frame() to make issues like the one fixed in the previous
commit more obvious.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6851>
The FORCE_KEYFRAME frame which has GST_VIDEO_CODEC_FRAME_FLAG_FORCE_KEYFRAME
bit set should be the sync point. So we should let it be an IDR frame to begin
a new GOP, rather than just promote it to an I frame.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6619>
The FORCE_KEYFRAME frame which has GST_VIDEO_CODEC_FRAME_FLAG_FORCE_KEYFRAME
bit set should be the sync point. So we should let it be an IDR frame to begin
a new GOP, rather than just promote it to an I frame.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6619>
__STDC_NO_ATOMICS doesn't seem to exist. In fact the only compiler
I've found that sets any of those is msvc, but it sets
__STDC_NO_ATOMICS__, not __STDC_NO_ATOMICS.
__STDC_NO_ATOMICS__ is the one documented by C11 standard.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6848>
This fixes the code regarding dropping "ghost frames", that is to say input
frames which ended up not producing any decoded frame.
The iteration itself makes sense.. but it was stopping at the "input" frame and
not the decoded frame we just got back.
When dealing with I-frame codecs, ffmpeg will decode frames in separate frames,
so there is no guarantee that they are decoding in order.
Fixes playback issues with such codecs
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6842>
Set the P and B frame qp to I frame value to avoid generating delta
QP between different frame types. For ICQ and QVBR modes, we can
only set the qpi value, so the qpp and qpb values should be set to
the same value as the qpi.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6841>
Address below message reported by SDK debug layer.
ID3D12Device::CheckFeatureSupport: Unsupported Decode Profile Specified.
Use ID3D12VideoDevice::CheckFeatureSupport with D3D12_FEATURE_VIDEO_DECODE_PROFILES
to retrieve a list of supported profiles
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6839>
There's nothing requiring <= 64 channels except for getting the reorder
map and creating a channel mixing matrix, but those won't be possible to
call anyway as channel positions can only express up to 64 channels.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6819>
Certain V4L2 fourccs don't (yet) have DRM counter parts, in which case
we can't create DMA_DRM caps for them. This is usually the case for
specific tilings, which are represented as modifiers for DMA formats.
While using these tilings is generally preferable - because of e.g.
lower memory usage - it can result in additional conversion steps when
interacting with DMA based APIs such as GL, Vulkan or KMS. In such cases
using a DMA compatible format usually ends up being the better option.
Before the addition of DMA_DRM caps, this was what playbin3 ended up
requesting in various cases - e.g. prefering NV12 over NV12_4L4 - but
the addition of DMA_DRM caps seems to confuse the selection logic.
As a simple and quite robust solution, assume that peers supporting
DMA_DRM caps always prefer these and reorder the caps accordingly.
In the future we plan to have a translation layer for cases where
there is a matching fourcc+modifier pair for a V4L2 fourcc, ensuring
optimal results.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6645>
Doing it in gst_play_new() means that bindings that directly call
g_object_new() with the GType wouldn't end up initializing both.
This affects at least the Python and GJS bindings.
gst_init() is nonetheless only called from gst_play_new() once because
calling it from class_init would likely lead to problems as that's
called from somewhere in the middle of GObject.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6801>
Don't add an extra ref if non-floating as that ref will never be
unreffed.
gst_bin_add() is transfer floating (alias to transfer none).
Fixes a leak when a non-floating ref was provided as a return value in
the request-aux-sender signal.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6807>
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>
Modifying the `avoid-reencoding` property of `encodebin` could potentially cause
it to reconfigure itself, in which case the source pad will be removed and then
re-added.
Therefore set that property *before* attempting to link to that pad.
Fixes smart-render
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6757>
when pipline is
glvideomixerelement->glcolorconvertelement->gldownloadelement and
glcolorconvertelement is not passthrough, the gl bufferpool between
glvideomixerelement and glcolorconvertelement will not add gl sync meta
during allocating buffer. This will cause that glcolorconvert's inbuf
has no sync meta to wait for.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6756>
1. The PTS of all frames should not be changed.
2. Just update the DTS based on the PTS. For the frame which is not
reordered, the DTS is equal to PTS. For frame which is reordered,
the DTS is equal to previous DTS. For example:
Input: F0[D0, P0] -- F1[D1, P1] -- F2[D2, P2] -- F3[D3, P3]
Output: F0[I, D0, P0] -- F3[P, D0, P3] -- F1[B, D1, P1] -- F2[B, D2, P2]
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6703>
1. The PTS of all frames should not be changed.
2. Just update the DTS based on the PTS. For the frame which is not
reordered, the DTS is equal to PTS. For frame which is reordered,
the DTS is equal to previous DTS. For example:
Input: F0[D0, P0] -- F1[D1, P1] -- F2[D2, P2] -- F3[D3, P3]
Output: F0[I, D0, P0] -- F3[P, D0, P3] -- F1[B, D1, P1] -- F2[B, D2, P2]
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6703>
Fixing deadlock in below case
* GC lock is taken by background thread, and the background thread calls
gst_d3d12_ipc_client_release_imported_data() which takes ipc lock
* ipc lock is already taken in ipc thread and trying to pushing GC data
via gst_d3d12_command_queue_set_notify()
* gst_d3d12_command_queue_set_notify() is trying to take GC lock
but it's already taken by background thread
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6749>
Ideally, GPU waiting should be scheduled just before executing command list.
But handling the case outside of converter is a bit complicated.
Under an assumption that constructed command list will be executed
immediately, schedules GPU-side waiting inside of conversion method
to simplify the flow.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6749>
Neither LIBDIR nor LIBPL are set with the native windows Python
(unlike MSYS2), so we need to use `prefix` which takes us to the
rootdir of the Python installation.
The name is also different: it's python312.dll, not python3.12.dll.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6734>
`python.get_variable('FOO', [])` becomes `python.get_variable('FOO')`
due to how Meson treats empty arrays in arguments, which breaks the
fallback feature of get_variable().
So we need to actually check whether the variable exists before trying
to fetch it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6734>
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>
First it derived mapping was disabled for P010 formats, but also there's an
issue with interlaced frames.
It would be possible to disable derived mapping only for interlaced (H.264
decoder and vadeinterlace) but it would spread the hacks along the code. It's
simpler and contained to disable derived completely for Mesa <23.3
Fixes: #3450
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6729>
Instead of duplicating the GStreamer format to DRM fourcc mapping, this patch
uses the GstVideo library helpers. This duplicates the big O of looking for,
since the two lists are traversed, but it's less error prone.
Partially reverts commit 547f3e8622.
Fixes: #3354
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6731>
The assertion that was present before is a bit too harsh, since there is now
a (understandable) use-case where this could happen.
In gapless use-case, with two files containing the same type (ex:audio). The
first one *does* expose a collection with an audio stream, but decoding
fails (for whatever reason).
That would cause us to have configured a audio combiner, which was never
used (i.e. not active).
Then the second file plays and we (wrongly) assume it should be activated
... whereas the combiner was indeed present.
Demote the assertion to a warning and properly handle it
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3389
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6737>
If waylandsink received buffer rate is high which causes frame
drop, the cached staged buffer will be replaced when next buffer
needs to be rendered and be freed after redraw. But there is
chance to get memory leak if ended without redraw. So need to
free staged buffer when do gst_wl_window_finalize().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6670>
`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>
The gstreamer-rs repos use debian based images already,
which we can later base on this one. Additionally it's
good to have another distro target so we avoid weird
fedoraisms when possible.
It will also be simpler to keep it up to date, as we
don't need to run the test suite against this build as
well.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6656>
There was an issue with this equality check, which was to figure out what to do
with PCR pids (whether they were part of the streams present or not) and whether
we ignore PCR or not.
Turns out ... we already took care of that further up in the function.
The length check can be simplified by just checking whether the length of
the *original* PMT and the new PMT are identical. Since we don't store "magic"
PCR streams in those, we can just use them as-is.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6713>
Set as much information as possible on the slot (including the associated
track) *before* the associated source pad is added to the element.
We need this so that incoming event/queries can be replied to if they are
received when adding the pad
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6690>
An application that triggers a state transition from PLAYING to PAUSED
needs to acquire the LIVE_LOCK. Consequently the LIVE_LOCK must not be
taken while pushing anything on the pads because this operation might
get blocked by something that cannot be unblocked without the
application being able to proceed with the state transitions for other
elements in the pipeline. This commit extends the previous behaviour
where the live lock was released before pushing buffers (indirectly
through the unlock before subclass->create) to now also include
unlocking before pushing events.
The issue was discovered in a case for WebRTC where the application
tried to shut down a pipeline but an event originating from a video
source element (based on basesrc) was in the process of being pushed
down the pipeline when it got stuck on the STREAM_LOCK for the pad after
the rtpgccbwe element. This lock in turn was held by the rtcpgccbwe
element as it was in the process of pushing data down the pipeline but
was stuck on the blocking probes installed on dtlssrtpenc to prevent
data from flowing before dtls keys had been negotiated. What should have
happened here is that the blocking probes should be removed, but that
can only happen if the application may continue driving the state
transitions.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6671>
A DPB buffer held by codec picture object may not be writable
at the moment, then gst_buffer_make_writable() will unref passed buffer.
Specifically, the use after free or double free can happen if:
* Crop meta of buffer copy is required because of non-zero
top-left crop position
* zero-copy is possible with crop meta
* A picture was duplicated, interlaced h264 stream for example
Interlaced h264 stream with non-zero top-left crop position
is not very common but it's possible configuration in theory.
Thus gst_buffer_make_writable() should be called with
GstVideoCodecFrame.output_buffer directly.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6706>
A DPB buffer held by codec picture object may not be writable
at the moment, then gst_buffer_make_writable() will unref passed buffer.
Specifically, the use after free or double free can happen if:
* Crop meta of buffer copy is required because of non-zero
top-left crop position
* zero-copy is possible with crop meta
* A picture was duplicated, interlaced h264 stream for example
Interlaced h264 stream with non-zero top-left crop position
is not very common but it's possible configuration in theory.
Thus gst_buffer_make_writable() should be called with
GstVideoCodecFrame.output_buffer directly.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6706>
The goal of this code was, for programs which were updates (i.e. adding/removing
streams but not completely changing) to allow dynamic addition/removal of
streams without completely removing everything.
But this wasn't 100% tested and there are a bunch of issues which make it fail
in plenty of ways.
For now disable that feature and force the legacy "add all pads again and then
remove old ones" behaviour to make it switch.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6651>
In addition to device removed status monitoring in gst_d3d11_result()
method, if ID3D11Device4 interface is available,
an event handle will be used for device removed status update.
And "device-removed" signal is removed since applications can monitor
the device removed status via gobject notify
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6699>
Adding new property in order to notify users of device removed status.
Once device removed status is detected, application should release
all ID3D12Device objects corresponding to the adapter, including
GstD3D12Device object. Otherwise D3D12CreateDevice() call for the
adapter will fail.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6699>
It seems that when D3D11CreateDevice collides in time
with other D3D11 calls, in particular the proccess of
creating a shader, it can corrupt the memory in the driver.
D3D11 spec doesn't seem to require any thread safety from
D3D11CreateDevice. Following MSDN, it is supposed to be called
in the beginning of the proccess, while GStreamer calls it with each
new pipeline.
Such crashes in the driver were frequently reproducing on the
Intel UHD 630 machine.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6686>
We suspect that it's not thread safe to just create and
destroy the device from any thread, particularly because
of D3D11CreateDevice, that is not documented as thread-safe.
While D3D11CreateDevice is usually protected from outside
by the gst_d3d11_ensure_element_data, it still can cross
with the Release() method of another device.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6686>
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>
Otherwise, if we run in to the copy case, this can cause these
groups to stay around with queued flag set, but never actually
queued, until gst_v4l2_allocator_flush() is called, which then
erroneously frees the associated memories, causing the release
function to decrement the allocator refcount where it was never
incremented, resulting in early allocator disposal, and either
deadlock or use after free.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6552>
Clarify the fact that `encodebasebin->src_pad` is set when using a static source
pad (`encodebin`) and when not set it's dynamically added source
pads (`encodebin2`).
Fixes usage of encodebin2 when profiles are updated
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6667>
The caps obtained from parsing the allocation query is borrowed and
should not be unreffed. This fixes criticals assertion introduced in
1.24.1.
(gst-launch-1.0:242): GStreamer-CRITICAL **: 19:48:02.667:
gst_mini_object_unref: assertion 'GST_MINI_OBJECT_REFCOUNT_VALUE (mini_object) > 0' failed
Fixes: 5189e8b956 ("v4l2codecs: decoders: Add DMA_DRM caps support")
Closes#3462
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6679>
We used to only track the first one but this was wrong, so we start
tracking all the children properties here, adapting the test which
was already thought for this to be implemented.
At the same time add some flags to determine how children properties
need to be handled adding a mode that means that all duplicated
children props will be set together when the user sets that particular
child property. This is going to be tested in a following commit.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/918>
Some decoder drivers need to wait enough capture buffers before
starting to decode. But the dequeued buffer flag LAST but empty
has no chance to queue back to driver, which makes decode hang
after seek. So need to queue back such kind of buffer to driver.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6579>
Output buffers don't have to be writable. Accepting read-only buffers
from the V4L2 buffer pool allows upstream elements to write directly
into the V4L2 buffers without triggering a CPU copy into a new buffer
from the same V4L2 buffer pool every time.
Tested with the vivid output device:
GST_DEBUG=GST_PERFORMANCE:7 gst-launch-1.0 videotestsrc ! v4l2sink device=/dev/video5
With this change, gst_v4l2_buffer_pool_dqbuf() must be allowed to not
resize read-only memories of output buffers.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6572>
../subprojects/gst-plugins-bad/tests/check/libs/gstlibscpp.cc:41:
fatal error: gst/mpegts/gstmpegts-enumtypes.h: No such file or directory
Could only pass the needed deps to the libscpp test, but gets
messier to maintain, so let's at it for consistency.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6643>
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>
Since https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6153 ,
subtitle "decoders" (i.e. which decode to raw text) are no longer auto-plugged
by parsebin.
But if a given format does not have a parser at all, we would end up outputting
non-time/non-parsed outputs.
In order to mitigate the issue, until such parsers are available, we check if
the subtitle stream is in TIME format or not (i.e. whether it comes from a
parser or demuxer). If not, we attempt to plug in a subtitle "decoder".
Fixes#3463
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6592>
Fix
gstanalyticsmeta.c:134: Warning: GstAnalytics: "@instance"
parameter unexpected at this location
warning (caused by the extraneous empty line in the doc chunk)
and align function arguments with documentation and header file
(handle -> instance).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6585>
* RED_OR_ALPHA8 will map value to alpha for OpenGL, use R8 to avoid
2nd shader
* Determine texel size for proper texture memory preparation
* QByteArray::fromRawData() does shallow copy and thus leads to use of
corrupted memory
* Make sure RGBA dummy texture is fully opaque
* QRhiTexture::create() must be called to allocate texture resources
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6578>
During RTP-Info synchronization, clock_base was temporarily switched
from the actual clock-base to the base RTP time and then back some lines
later.
Instead directly work with the base RTP time. The comment about using a
signed variable for convenience doesn't make any sense because all
calculations done with the value are unsigned.
Similarly, rtp_clock_base was overridden with the rtp_delta when
calculating it, which was fine because it is not used anymore
afterwards. Instead, introduce a new variable `rtp_delta` to make this
calculation clearer.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6536>
It's not in the same period as the current RTP base time but always in
the very first period. This avoids using it again at a much later time.
The code in question is only triggered with rtcp-sync=rtp-info.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6536>
It is compared to other extended RTP timestamps all over rtpjitterbuffer
and since 4df3da3bab the initial extended RTP timestamp is not equal
anymore to the plain RTP time.
Continue passing a non-extended RTP timestamp via the `sync` signal for
backwards compatibility. It will always be a timestamp inside the first
extended timestamp period anyway.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6536>
When the buffer DTS is estimated based on arrival time at the
jitterbuffer (rather than provided on the incoming buffer itself),
it shouldn't be used for skew adjustment. The typical case is
packets being deinterleaved from a tunnelled TCP/HTTP RTSP stream,
and the arrival times at the jitter buffer are not well enough
correlated to usefully do skew adjustments.
This restores the original intended behaviour for the 'estimated dts'
path, that was broken years ago during other jitterbuffer refactoring.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6509>
For some cameras `gst_jpeg_parse_app0()` fails on a invalid segment.
While this is likely a driver or firmware bug that should be addressed
accordingly, it's not fatal and likely does not deserve a bus message on
every frame, flooding journals.
Turn down the volume of the warnings by turning them into object
warnings. If we conclude that in some cases we'd still want bus
warnings, they can be done more fine-grained in the
`gst_jpeg_parse_appX()` functions.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6490>
Reset the waiting thread counter in all places to be consistent
when sending signal for the audio ring buffer. This fix applies it to
pause, stop and release, which are states that will go into a callback
of the subclass. Having the waiting counter reset will avoid having
executing thread of the same subclass trying to take the mutex when
callong gst_audio_ring_buffer_advance.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6195>
GstD3D12Device objetct's internal resources are singletons per adapter
already though, the object itself is not a singleton.
Due to the singleton design (unlike other APIs such as d3d11),
d3d12 device context sharing is not a strict requirement
for zero-copy, but handles context ones to make things less noisy.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6513>
On practice a failure happened due to a race condition, the instance
already have been freed, but it could also happen if the instance
would be null.
Instead of crashing this sanity check is a more suitable option,
since with G_DEBUG=fatal-warnings it will crash too.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6475>
Adding RGBA, RGBx, BGRA, BGRx, VUYA and RGB10A2_LE format support for performance.
However, these formats are not still recommended if upstream can support
native YUV formats (e.g., NV12, P010) since NVENC does not expose
conversion related optiones. Note that VUYA format is 4:4:4 YUV format
already but NVENC runtime will convert it to 4:2:0 format internally
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6417>
If parameters remain similar enough to avoid either encoder reopening
or downstream renegotiation, avoid it.
This is going to be useful for dynamic parameters setting.
To check if the stream parameters changed, so the internal encoder has
to be closed and opened again, are required two steps:
1. If input caps, format, profile, chroma or rate control mode have changed.
2. If any of the calculated variables and element properties have changed.
Later on, only if the output caps also changed, the pipeline
is renegotiated.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6441>
If parameters remain similar enough to avoid either encoder reopening
or downstream renegotiation, avoid it.
This is going to be useful for dynamic parameters setting.
To check if the stream parameters changed, so the internal encoder has
to be closed and opened again, are required two steps:
1. If input caps, format, profile, chroma or rate control mode have changed.
2. If any of the calculated variables and element properties have changed.
Later on, only if the output caps also changed, the pipeline
is renegotiated.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6441>
The last frame which has the smallest diff should be consider as
the first choice rather than the golden frame. Especially when only
one reference available, this way can improve the BD rate about 5
percentage.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6379>
Some extreme case such as "videotestsrc pattern=1" can generate pure
white noise videoes, for which encoder may generate too big output
for current coded buffer size. We now consider the qindex and bitrate
to avoid that.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6483>
It might happen that the key event arrives when the d3d11videosink
is stopping. In case of GstD3D11WindowWin32 it can raise a
navigation event even when the sink is already freed, because the
window object's refcount may reach 0 in the window thread. In
other words sometimes the GstD3D11WindowWin32 lives few ms more
then the GstD3D11VideoSink, because it's freed asynchronously.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6476>
In the case of multi-channels transcoding, a context with child
sesseion can be parent for others, so we need to check if the
msdkcontext has any child session in the list to avoid session
leaks. Otherwise, we will see the failure of closing a parent
session because one of its child's child session not released.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6259>
And handle the case of a NULL buffer being returned cleanly, which is
valid as long as a buffer list is returned instead. Previously this
would cause an assertion because of calling gst_buffer_unref() with
NULL.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6460>
The attempt to free the domain data is happeing twice during the ptp deinit.
Once while iterating through the list domain_data and second while iterating
through the list domain_clocks, so this is crashing the application
trying to gst_ptp_deinit
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6443>
If we can calculate timestamps for buffers, then set the duration
on outgoing buffers based on the number of samples depayloaded.
This can fix the muxing to mp4, where otherwise the last packet
in a muxed file will have 0 duration in the mp4 file.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6447>
Calling gst_pad_peer_query_caps() without a filter can give us EMPTY caps, whereas all the code below
assumes that's not the case. Replacing query+intersect with a filtered query ensures we always get a subset
of the template caps back.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6429>
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>
Fixes this error:
dav1d| Subproject dav1d finished.
gst-plugins-rs| WARNING: Subproject 'dav1d' did not override 'dav1d' dependency and no variable name specified
gst-plugins-rs| Dependency dav1d from subproject subprojects/dav1d-1.4.1 found: NO
subprojects/gst-plugins-rs/meson.build:382:14: ERROR: Dependency 'dav1d' is required but not found.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6436>
Some driver doesn't implement enum_framesize. The maximum supported
size can be got by trying format with a very large size. Also need
to set max_width/max_height for this case, otherwise default encoded
buffer size 256kB is too small.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6416>
This reverts commit 8e923a8e2d.
This caused regressions, see #3303.
Without this commit, osxaudiosrc ! osxaudiosink won't work
right, but since that hasn't really been a huge problem
for years it's probably best to revert this until a proper
solution can be figured out.
(cherry picked from commit f04f86f3ee)
(cherry picked from commit 93255efece)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6405>
The --atleast-version implies --exists, but the implementation in
earlier commits had the version check applied any time the --exists was
checked, and the default value of the major and minor versions were set
to the GStreamer major and minor versions. The resulting behavior would
have gst-inspect return '1' if the plugin's version didn't match
gstreamer's even when --atleast-version was not specified in the command
line args. The change in this patch removes that behavior and adds
tests to verify that if --exists is specified WITHOUT --atleast-version
the version check will NOT be applied. If both arguments are specified
and the version does not match the arg-supplied version number, a new
return code of '2' is used to uniquely identify the failure.
Fixes#3246
Signed-off-by: Thomas Goodwin <thomas.goodwin@laerdal.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6191>
In an early non-linked scenario, this was causing a ton of criticals about the queue array,
because the output callback would still fire for leftover frames that were still being processed by VT
at the time the output loop stopped. This makes sure they're flushed correctly as well.
Also renames gst_vtdec_loop to gst_vtdec_output_loop for consistency with related functions.
wip
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6397>
Sometimes a call to negotiate (and thus drain) can happen from the output loop
(via finish_frame()), which will tell VT to output all internal frames, but that won't succeed
if we happen to decide to wait for the queue to empty (because the loop is waiting for draining to finish and
will not make space in the queue!). This commit adds an override for the queue size limit if we're draining/flushing.
This bug could happen for any formats, but was especially obvious for ProRes, which has dpb_size of 0.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6397>
Because ID3D12Device objects are singletons per adapter,
GstD3D12Device was following the API design, that is, keep track
of global GstD3D12Device objects and reuses it.
That means ID3D12Device object can be released at the time
when GstD3D12Device is destroyed.
But exetrnal APIs such as NVENC does not seem to be happy
with the released ID3D12Device, that could be a driver bug though.
Let's hold already opened ID3D12Device permanently without releasing
it for now.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6395>
`on_error()` can be called with a NULL details structure, so in that situation
the `gst_structure_copy()` would raise a critical warning. Create an empty
structure instead of attempting to copy a NULL one.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6385>
In order to simplify caps negotiations for clients and, notably, be more
compatible with va* decoders.
Crucially this allows clients to know ahead of time whether buffers will
actually be DMABufs.
Similar to GstVaBaseDec we only announce system memory caps if the peer
has ANY caps. Further more, and again like va decoders, we fail in
`decide_allocation()` if DMA_DRM caps are used without VideoMeta.
Apart from buggy peers this can happen e.g. when a peer with ANY caps
is used in combination with caps filters.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5890>
Most importantly rely on video info helpers instead of manual parsing
of caps, which will allow us to use additional helpers in the future.
While on it, tighen the check for supported formats - failing that
indicates a bug in caps negotiation - and make some style changes.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5890>
This ensures we don't create filter caps that are not supported by the
individual codec implementations, as well as that the resulting caps
have the required fields so they can be turned into a GstVideoFormat.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5890>
This is the maximum amount supported by aacenc. 8-channel output fully works.
16-channel also encodes fine, but codec-utils isn't able to parse its channel config,
so output level will not be shown in caps. For that to work, GASpecificConfig parsing
needs to be implemented. It's not a critical issue and can be worked on at a later date.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6359>
When this error gets caught the GstD3D11Device object raises the new
"device-removed" signal. This allows to handle the error from outside:
stop the playback, re-create the player, replace the catched GstContext by
the new one.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6193>
Adds the `atenc` element capable of encoding AAC-LC audio, using the AudioToolbox framework.
It's able to encode up to 7.1 channel configurations.
Comes with basic knobs for rate control (bitrate for CBR, quality for VBR).
Support for more profiles (LD, HE-AAC) should be simple, but is not included here because of bugs
with parsing of the AudioSpecificConfig.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6254>
None of the GL allocators actually offer a generic alloc() implementation. As a
side effect, they cannot be offered as they don't work with generic video
buffer pool.
Our specialized buffer pool can be dropped by tee or alphacombine as sharing the
same buffer pool over two branch is not supported by the pool API.
Fixes#3372
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6327>
Do not chain up to parent's GstBufferPool::start() which will do
preallocation. We don't want it to be preallocated
since there are various cases where negotiated downstream buffer pool is
not used at all (e.g., zero-copy decoding, IPC elements).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6326>
This fixes a crash in `gst_va_h264_enc_class_init` and `gst_va_h265_enc_class_init`
(and probably also in gst_va_av1_enc_class_init) when calling
`g_object_class_install_properties (object_class, n_props, properties);`
When rate_control_type is 0, the following code is executed in :
```
} else {
n_props--;
properties[PROP_RATE_CONTROL] = NULL;
}
```
n_props has initially a value of N_PROPERTIES but PROP_RATE_CONTROL
is not the last element in the array, so it's making
g_object_class_install_properties fail to iterate over the
properties array.
This applies the same fix to gstvah264enc.c, gstvah265enc.c and
gstvaav1enc.c.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6319>
On systems using UsrMerge (like openSUSE or Fedora), /lib64 is
a symlink to /usr/lib64. So dladdr is returning the path to
the gstreamer library in /lib64 in priv_gst_get_relocated_libgstreamer.
Later gst_plugin_loader_spawn tries to build the path to the
gst-plugin-scanner helper from /lib64 and ends up trying to use
/lib64/../libexec/gstreamer-1.0/gst-plugin-scanner which doesn't exist.
By canonicalizing the path with a call to realpath, gst-plugin-scanner
is found correctly under
/usr/lib64/../libexec/gstreamer-1.0/gst-plugin-scanner
Similar change applied to gstreamer/libs/gst/net/gstptpclock.c
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6322>
This matches autoplug in other places such as decodebin, otherwise we
will pick "randomly" based on the order in which plugins are
registered, which is mostly dependent on the order in which readdir()
returns items.
So let's make it predictable.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6227>
osxaudio has a few helper methods potentially useful in atdec (or future atenc), like GStreamer -> CoreAudio
channel mapping. Doesn't make sense to duplicate them in applemedia, and atdec is the only audio-oriented
element there anyway.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6223>
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>
Memory from gst_adapter_map() could live shorter than the GstMemory that the GstBuffer wraps around it, which in lucky
cases 'just' caused a re-use of the same memory for multiple (potentially still in use!) input buffers, but could easily
end up pointing to an already-freed memory.
Manifested when an AudioToolbox encoder kept getting silence inserted in seemingly random circumstances, turned out
to be the memory being re-used by GStreamer at the same time that the AT API was processing it...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6320>
This will mimic the playbin2 behaviour, which sets the "next" entry to be
NULL.
The biggest impact this has is that when going back to READY the current play
entry will be discarded (instead of being kept around for when you go back to
PAUSED/PLAYING).
Fixes#3371
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6324>
Since commit 4d86f994, when setting an RTSP media both shared and
reusable, streaming cannot be restarted after the first time all the
clients disconnect. That happens because the sockets (unlike
addresses) of GstRTSPStream are not cleared in
gst_rtsp_stream_leave_bin, and on restart sockets and addresses are
not allocated in gst_rtsp_stream_allocate_udp_sockets, and then the
check in create_sender_part fails. Fix this by clearing sockets in
gst_rtsp_stream_leave_bin.
Fixesgstreamer/gst-rtsp-server#113
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6325>
find_slot_for_stream_id() will return a slot which has the request stream-id as
active_stream *or* pending_stream (i.e. the slot on which that stream is
currently being outputted or will be outputted).
When figuring out which slot to use (if any) we want to consider stream-id
which *will* appear on a given slot which isn't outputting anything yet the same
way as if we didn't find a slot yet.
Fixes races when doing intensive state changes
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6270>
With the new copy_opaque system, the corresponding frame is stored in the
picture opaque ref.
This also handles the case where the "regular" opaque might be empty in the
case of "DECODE_ONLY" frames, since it that field is set in `get_buffer2()`
which might not be called for those frames
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6301>
When we're doing a state change from PLAYING to NULL, first we invoke
gst_rtspsrc_loop_send_cmd_and_wait (..., CMD_CLOSE, ...) during
PAUSED_TO_READY which will schedule a TEARDOWN to happen async on the
task thread.
The task thread will call gst_rtspsrc_close(), which will send the
TEARDOWN and once it's complete, it will call gst_rtspsrc_cleanup()
without taking any locks, which frees src->streams.
At the same time however, the state change in the app thread will
progress further and in READY_TO_NULL it will call gst_rtspsrc_stop()
which calls gst_rtspsrc_close() a second time, which accesses
src->streams (without a lock again), which leads to simultaneous
access of src->streams, and a segfault.
So the state change and the cleanup are racing, but they almost always
complete sequentially. Either the cleanup sets src->streams to NULL or
_stop() completes first. Very rarely, _stop() can start while
src->streams is being freed in a for loop. That causes the segfault.
This is unlocked access is unfixable with more locking, it just leads
to deadlocks. This pattern has been observed in rtspsrc a lot: state
changes and cleanup in the element are unfixably racy, and that
foundational issue is being addressed separately via a rewrite.
The bandage fix here is to prevent gst_rtspsrc_stop() from accessing
src->streams after it has already been freed by setting src->state to
INVALID.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6302>
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>
And also re-timestamp them with the current buffer's PTS.
Not doing so keeps the timestamps of event packets as
GST_CLOCK_TIME_NONE or the timestamp of the previous buffer, both of
which are bogus.
Making sure that (especially) the first packet has a valid timestamp
allows putting e.g. the NTP timestamp RTP header extension on it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5173>
Parse the speed and scale in the server's response
*before* the range, so that the range start/stop
are swapped (or not swapped) correctly based
on the server's actual chosen values. Otherwise,
the old rate from the segment is used - what the
last seek asked for, but not necessarily what
the server chooses.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6248>
In the situation where playback starts from a keyframe before
the target playback segment, then the first buffers will be
outside the configured segment and gst_segment_to_stream_time()
will return GST_CLOCK_TIME_NONE unconditionally.
If drop-out-of-segment is false, the RTP buffers will not be
dropped, but will be sent witout ONVIF extension timestamps
and given GST_CLOCK_TIME_NONE timestamps on the receiver.
Instead, use gst_segment_to_stream_time_full() to extrapolate
stream time outside the segment so that such buffers still
get assigned their correct timestamps on the receiver.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6248>
Don't accidentally include the stuffing byte (if present)
into the bottom field size. It should only be included in the
total segment length.
Fixes problems with FFmpeg not rendering the subtitles
with a stuffing byte, giving a "Invalid object location!" error.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6250>
Cocoa version of glwindow only checks the preferred size upon window creation. glimagesink sets the size right before
calling gst_gl_window_show(), which might be way after the window is created in some cases. If the size was set too
late, glimagesink on macOS would remain 320x240 unless manually resized.
This change makes sure to resize the existing window when _show() is called.
Curiously, this has always been an issue, but went from manifesting every once in a while to being almost completely
broken once old event loop workarounds were removed and gst_macos_main() was introduced.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6185>
Don't use g_return_val_if_fail() to catch the
open-ended segment or empty segment cases in
gst_segment_to_running_time_full()
g_return_val_if_fail() is for programmer errors,
and can be compiled out with a flag.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6219>
Provide a clock from the source that is a monotonic system clock with
the rate corrected based on the measured and ideal capture rate of the
frames.
If this clock is selected as pipeline clock, then provide perfect
timestamps to downstream.
Otherwise, if the pipeline clock is the monotonic system clock, use the
internal clock for converting back to the monotonic system clock.
Otherwise, use the monotonic system clock time calculated in the above
case and convert that to the pipeline clock.
In all cases this will give a smoother time than the previous code,
which simply took the difference between the driver provided capture
time and the current real-time clock time, and applied that to the
current pipeline clock time.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6208>
Otherwise there's a small window between querying the state and doing
the transfer in which a frame could be dropped, and we would then output
the frame right after the dropped one as if it was the dropped frame.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6208>
In low delay B mode, the P frame is converted as B frame with forward
references. For example, One P frame may refers to P-1, P-2 and P-3 in
list0 and refers to P-3, P-2 and P-1 in list1.
So the num in list0 and list1 does not reflect the forward_num and
backward_num. The vaapi does not provide ref num for forward or backward
so far. In this case, we just consider the backward_num to be 1 conservatively.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6249>
In b_pyramid mode, B frames can be ref and prevPicOrderCntLsb can
be the B frame POC which is smaller than the P frame. This can cause
POC diff bigger than MaxPicOrderCntLsb/2 and generate wrong POC value.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6249>
<doc xml:space="preserve" filename="../subprojects/gstreamer/gst/gstvecdeque.c">#GstVecDeque is an object that provides standard double-ended queue (deque)
functionality based on an array instead of linked lists. This reduces the
overhead caused by memory management by a large factor.</doc>
<doc xml:space="preserve" filename="../subprojects/gstreamer/gst/gstvecdeque.c">Drops the queue element at position @idx from queue @array and copies the
data of the element or structure that was removed into @p_struct if
<doc xml:space="preserve" filename="../subprojects/gstreamer/gst/gstvecdeque.c">address into which to store the data of the dropped structure, or NULL</doc>
<doc xml:space="preserve" filename="../subprojects/gstreamer/gst/gstvecdeque.c">Finds an element in the queue @array, either by comparing every element
with @func or by looking up @data if no compare function @func is provided,
and returning the index of the found element.</doc>
<doc xml:space="preserve" filename="../subprojects/gstreamer/gst/gstvecdeque.c">Returns the head of the queue @array without removing it from the queue.</doc>
<doc xml:space="preserve" filename="../subprojects/gstreamer/gst/gstvecdeque.c">Returns the item at @idx in @array, but does not remove it from the queue.</doc>
<doc xml:space="preserve" filename="../subprojects/gstreamer/gst/gstvecdeque.c">Returns the item at @idx in @array, but does not remove it from the queue.</doc>
<doc xml:space="preserve" filename="../subprojects/gstreamer/gst/gstvecdeque.c">Returns the tail of the queue @array, but does not remove it from the queue.</doc>
<doc xml:space="preserve" filename="../subprojects/gstreamer/gst/gstvecdeque.c">Returns the tail of the queue @array, but does not remove it from the queue.</doc>
<doc xml:space="preserve" filename="../subprojects/gstreamer/gst/gstvecdeque.c">Returns the head of the queue @array and removes it from the queue.</doc>
<doc xml:space="preserve" filename="../subprojects/gstreamer/gst/gstutils.h">Store a 16 bit unsigned integer value in big endian format into the memory buffer.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">Number of audio formats in #GstAudioFormat.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.c">Returns a string containing a descriptive name for the #GstAudioFormat.
Since 1.26 this can also be used with %GST_AUDIO_FORMAT_UNKNOWN, previous
versions were printing a critical warning and returned %NULL.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">Gets the current segment base number of the ringbuffer.
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">Current segment base number of the ringbuffer.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">Gets the current segment number of the ringbuffer.
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">Current segment number of the ringbuffer.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">Check if the ringbuffer is acquired and ready to use.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">Sets the current segment number of the ringbuffer.
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h">The structure containing the format specification of the ringbuffer.
@ -10919,12 +10987,18 @@ string is not a known format.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.c">Returns a string containing a descriptive name for the #GstAudioFormat.
Since 1.26 this can also be used with %GST_AUDIO_FORMAT_UNKNOWN, previous
versions were printing a critical warning and returned %NULL.</doc>
<doc xml:space="preserve" filename="../subprojects/gstreamer/libs/gst/base/gstqueuearray.c">#GstQueueArray is an object that provides standard queue functionality
based on an array instead of linked lists. This reduces the overhead
caused by memory management by a large factor.</doc>
<doc xml:space="preserve" filename="../subprojects/gstreamer/libs/gst/base/gstqueuearray.c">Clears queue @array and frees all memory associated to it.</doc>
<doc xml:space="preserve" filename="../subprojects/gstreamer/libs/gst/base/gstqueuearray.c">Drops the queue element at position @idx from queue @array.</doc>
<doc xml:space="preserve" filename="../subprojects/gstreamer/libs/gst/base/gstqueuearray.c">Drops the queue element at position @idx from queue @array and copies the
data of the element or structure that was removed into @p_struct if
<doc xml:space="preserve" filename="../subprojects/gstreamer/libs/gst/base/gstqueuearray.c">Finds an element in the queue @array, either by comparing every element
with @func or by looking up @data if no compare function @func is provided,
and returning the index of the found element.</doc>
<doc xml:space="preserve" filename="../subprojects/gstreamer/libs/gst/base/gstqueuearray.c">Index of the found element or -1 if nothing was found.</doc>
@ -13136,8 +13155,9 @@ and returning the index of the found element.</doc>
<doc xml:space="preserve" filename="../subprojects/gstreamer/libs/gst/base/gstqueuearray.c">Returns the head of the queue @array without removing it from the queue.</doc>
<doc xml:space="preserve" filename="../subprojects/gstreamer/libs/gst/base/gstqueuearray.c">pointer to element or struct, or NULL if @array was empty. The
@ -13208,8 +13232,9 @@ remove it from the queue.</doc>
<doc xml:space="preserve" filename="../subprojects/gstreamer/libs/gst/base/gstqueuearray.c">Returns the item at @idx in @array, but does not remove it from the queue.</doc>
<doc xml:space="preserve" filename="../subprojects/gstreamer/libs/gst/base/gstqueuearray.c">Returns the item at @idx in @array, but does not remove it from the queue.</doc>
<doc xml:space="preserve" filename="../subprojects/gstreamer/libs/gst/base/gstqueuearray.c">Returns the tail of the queue @array, but does not remove it from the queue.</doc>
<doc xml:space="preserve" filename="../subprojects/gstreamer/libs/gst/base/gstqueuearray.c">Returns the tail of the queue @array, but does not remove it from the queue.</doc>
<doc xml:space="preserve" filename="../subprojects/gstreamer/libs/gst/base/gstqueuearray.c">Returns the head of the queue @array and removes it from the queue.</doc>
<doc xml:space="preserve" filename="../subprojects/gstreamer/libs/gst/base/gstqueuearray.c">pointer to element or struct, or NULL if @array was empty. The
<doc xml:space="preserve" filename="../subprojects/gstreamer/libs/gst/base/gstqueuearray.c">Pushes @data to the queue @array, finding the correct position
by comparing @data with each array element using @func.
@ -13338,6 +13370,7 @@ and expected access patterns, a different data structure might be better.
Assumes that the array is already sorted. If it is not, make sure
<doc xml:space="preserve" filename="../subprojects/gstreamer/libs/gst/base/gstqueuearray.c">Pushes the element at address @p_struct into the queue @array
(copying the contents of a structure of the struct_size specified
when creating the queue into the array), finding the correct position
@ -13372,6 +13405,7 @@ and expected access patterns, a different data structure might be better.
Assumes that the array is already sorted. If it is not, make sure
<doc xml:space="preserve" filename="../subprojects/gstreamer/libs/gst/base/gstqueuearray.c">Sorts the queue @array by comparing elements against each other using
<doc xml:space="preserve" filename="../subprojects/gstreamer/libs/gst/base/gstqueuearray.c">Allocates a new #GstQueueArray object for elements (e.g. structures)
of size @struct_size, with an initial queue size of @initial_size.</doc>
<doc xml:space="preserve" filename="../subprojects/gstreamer/libs/gst/base/gstqueuearray.c">Allocates a new #GstQueueArray object for elements (e.g. structures)
of size @struct_size, with an initial queue size of @initial_size.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/pbutils/encoding-profile.c">Converts a string in the "encoding profile serialization format" into a
GstEncodingProfile. Refer to the encoding-profile documentation for details
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/pbutils/encoding-profile.c">A newly created GstEncodingProfile or NULL if the
input string is not a valid encoding profile serialization format.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/pbutils/encoding-profile.c">The string to convert into a GstEncodingProfile.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/pbutils/encoding-profile.c">Sets the name of the #GstElement that implements the #GstPreset interface
to use for the profile.
This is the name that has been set when saving the preset.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/pbutils/encoding-profile.c">Sets the name of the preset to be used in the profile.
This is the name that has been set when saving the preset.
You can list the available presets for a specific element factory
using `$ gst-inspect-1.0 element-factory-name`, for example for
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/pbutils/encoding-profile.c">Sets the name of the #GstPreset's factory to be used in the profile.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/pbutils/encoding-profile.c">Sets the name of the #GstPreset's factory to be used in the profile. This
is the name of the **element factory** that implements the #GstPreset interface not
the name of the preset itself (see #gst_encoding_profile_set_preset).</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/pbutils/encoding-profile.c">The name of the preset to use in this @profile.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/pbutils/encoding-profile.c">The name of the element factory to use in this @profile.</doc>
<type name="utf8" c:type="const gchar*"/>
</parameter>
</parameters>
@ -2175,6 +2211,21 @@ single segment before the encoder, #FALSE otherwise.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/pbutils/encoding-profile.c">Converts a GstEncodingProfile to a string in the "Encoding Profile
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/pbutils/encoding-profile.c">A string representation of the GstEncodingProfile.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/pbutils/encoding-profile.c">A #GstStructure defining the properties to be set to the element
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/pbutils/gstpluginsbaseversion.h">The minor version of GStreamer's gst-plugins-base libraries at compile time.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/pbutils/gstpluginsbaseversion.h">The nano version of GStreamer's gst-plugins-base libraries at compile time.
Actual releases have 0, GIT versions have 1, prerelease versions have 2-...</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/pbutils/codec-utils.c">Transform a seq_level_idx into the level string</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/pbutils/codec-utils.c">the level string or %NULL if the seq_level_idx is unknown</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/pbutils/codec-utils.c">Transform a level string from the caps into the seq_level_idx</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/pbutils/codec-utils.c">the seq_level_idx or 31 (max-level) if the level is unknown</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/pbutils/codec-utils.c">Converts a RFC 6381 compatible codec string to #GstCaps. More than one codec
string can be present (separated by `,`).
@ -3790,6 +3869,7 @@ Moreover, you can set the extra properties:
* `|single-segment=true` (See #gst_encoding_profile_set_single_segment)
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/rtp/gstrtcpbuffer.h">Valid value for the first two bytes of an RTCP packet after applying
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/video/navigation.c">a bit-mask representing the state of the modifier keys (e.g. Control,
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/video/navigation.c">Create a new navigation event for the given key mouse double click.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/video/navigation.c">a bit-mask representing the state of the modifier keys (e.g. Control,
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/video/navigation.c">The type of mouse event, as a text string. Recognised values are
"mouse-button-press", "mouse-button-release" and "mouse-move".</doc>
"mouse-button-press", "mouse-button-release", "mouse-move" and "mouse-double-click".</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/video/navigation.h">An event cancelling all currently active touch points.</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.c">Converts a FOURCC value into the corresponding #GstVideoFormat.
If the FOURCC cannot be represented by #GstVideoFormat,
@ -10414,8 +10463,10 @@ no corresponding FOURCC value, 0 is returned.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/video/video-format.c">Returns a string containing a descriptive name for
the #GstVideoFormat if there is one, or NULL otherwise.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/video/video-format.c">Returns a string containing a descriptive name for the #GstVideoFormat.
Since 1.26 this can also be used with %GST_VIDEO_FORMAT_UNKNOWN, previous
versions were printing a critical warning and returned %NULL.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/video/navigation.c">a bit-mask representing the state of the modifier keys (e.g. Control,
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/video/navigation.c">Create a new navigation event for the given key mouse double click.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/video/navigation.c">a bit-mask representing the state of the modifier keys (e.g. Control,
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/video/video-format.c">Returns a string containing a descriptive name for
the #GstVideoFormat if there is one, or NULL otherwise.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/video/video-format.c">Returns a string containing a descriptive name for the #GstVideoFormat.
Since 1.26 this can also be used with %GST_VIDEO_FORMAT_UNKNOWN, previous
versions were printing a critical warning and returned %NULL.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/vulkan/gstvkoperation.h">Encoder query result. Expected to be used in gst_vulkan_operation_get_query()</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/vulkan/gstvkimagebufferpool.c">Encode @caps are used when the buffers are going to be used either as encoded
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/vulkan/gstvkimagebufferpool.c">the #GstStructure with the pool's configuration.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-bad/gst-libs/gst/vulkan/gstvkqueue.c">Creates a #GstVulkanDecoder object if @codec decoding is supported by @queue</doc>
Rafael Caricio
