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>
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>
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>
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>
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>
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>
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>
../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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
