Since the AV1 specification is not explicitly mentioning about
the array size bounds, array sizes in scalability structure
should be defined as possible maximum sizes that can have.
Also, this commit removes GST_AV1_MAX_SPATIAL_LAYERS define from
public header which is API break but the define is misleading
and this patch is introducing ABI break already
ZDI-CAN-22300
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5824>
This commit corrects the mapping relationship between RGB and BGR in GST and DRM.
The previous mapping was incorrect, causing potential color mismatches in the output.
The changes are as follows:
{WL_SHM_FORMAT_RGB888, DRM_FORMAT_RGB888, GST_VIDEO_FORMAT_BGR},
{WL_SHM_FORMAT_BGR888, DRM_FORMAT_BGR888, GST_VIDEO_FORMAT_RGB},
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5792>
If the ladspa plugin is enabled explicitly or via auto-features, the
liblrdf dependency can not be disabled.
As the RDF parsing currently provides hardly any features, the possibility
to disable it fairly useful.
Fixes: #3168
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5802>
When doing a segment seek, the base offset in the new segment
would be increased by segment.position which is basically the
timestamp of the last packet. This does not include the duration
of the last packet though, so might be slightly shorter than the
actual duration of the clip or the requested segment.
Increase the base offset by the segment duration instead when
accumulating segments, which is more correct as it doesn't cut
off the last frame and makes the effective loop segment duration
consistent with the actual duration returned from a duration
query.
In case a segment stop was specified it's also possible that
some data was sent beyond the stop that's necessary for decoding
so the base offset increment should be based on that then and
not on the timestamp of the last buffer pushed out.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5787>
This avoid a build failure when compiling against OpenSSL 3.2.0. The
problem is when windows.h is included before WinSock2.h. Because
windows.h includes winsock.h[1]. Defining _WINSOCKAPI_ stops windows.h
including winsock.h.
Error:
```
[748/1041] Compiling C object ext/dtls/gstdtls.dll.p/gstdtlscertificate.c.obj
FAILED: ext/dtls/gstdtls.dll.p/gstdtlscertificate.c.obj
[...]
Windows Kits\10\include\10.0.17763.0\shared\ws2def.h(235): error C2011: 'sockaddr': 'struct' type redefinition
Windows Kits\10\include\10.0.17763.0\um\winsock.h(482): note: see declaration of 'sockaddr'
```
[1] https://stackoverflow.com/a/1372836
Closes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3167
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5783>
Clip tile rows and cols to 64 as describe in AV1 specification
to avoid writing outside array range but preserve sb_cols
and sb_rows value which are used to futher computation.
Fixes ZDI-CAN-22226 / CVE-2023-44429
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5734>
When the subclass attempts to finish without an explicit `out_buffer`,
we take a buffer from our adapter. We need to make this buffer writable
before copying the metadata.
This led to data races such as in the following pipeline, which randomly
messed up the buffer PTS:
gst-launch-1.0 -e audiotestsrc timestamp-offset=5555 num-buffers=100 \
! opusenc ! tee name=t ! queue ! opusparse ! fakesink silent=0 \
t. ! queue ! opusparse ! fakesink silent=0 -v | grep '0000, dur'
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5720>
Whenever that caps changes does not imply that a new segment will start.
Don't reset the last_ts if only the caps have changed. This fixes issues
if you have a stream without only first frame with TS=0, and have resolution
change happening. This was a regression introduced by !3059, which issue was
described in #1352. The reported issue is still fix after this change.
Fixes#1034
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5712>
When seek flush, gst v4l2 buffer pool flush is not atomic which will
lead double enqueue buffer (qbuf) issue, and v4l2 buffer pool qbuf is
also not atomic which will lead no free buffer found in the pool.
1. add lock for calculate enqueue number in streamon function
2. add lock for v4l2 capture end streamoff in pool flush function
3. lock the whole funciton of v4l2 buffer pool qbuf, then the buffer
pool index and qbuf operation are atomic
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5695>
When copying a buffer, for example with gst_buffer_make_writable(), the
new buffer might reference the same GstMemory as the src buffer,
making those memories not writable. If the src buffer gets disposed
first it should return to its buffer pool, but since some of its
memories are not writable it gets discarded and new buffer/memory gets
allocated.
Solves this by making the new buffer keep a reference to the src buffer,
that ensures that by the time the src buffer gets disposed no other
buffer are referencing its memories and it can thus return safely to its
pool.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5696>
gst_buffer_add_parent_buffer_meta() is used when a GstBuffer uses
GstMemory from another buffer that was allocated from a pool. In that
case we want to make sure the buffer returns to the pool when the memory
is writable again, otherwise a copy of the memory is created. That means
the child buffer must drop its ref to the memory first, then drop the
ref to parent buffer so it can return to the pool when it is the only
owner of the memory.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5696>
Take the case into account when user filters have been set before the
source gets updated.
Note that the further linking of the filters, if present, happens below
in the `gst_camera_bin_check_and_replace_filter()` calls.
The audio filter is still affected by the same issue but left out for
now.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5682>
Even if IDXGIOutput6 says current display colorspace is HDR,
captured texture via IDXGIOutputDuplication::AcquireNextFrame()
is converted frame by OS unless we use IDXGIOutput5::DuplicateOutput1()
with DXGI_FORMAT_R16G16B16A16_FLOAT format, in order for captured
frame to be scRGB color space. Then application should perform
tonemap operation based on reported display white level, color primaries, etc.
Since we don't have any tonemapping implementation, ignores colorimetry
reported by IDXGIOutput6.
Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3128
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5679>
This commit makes sure that pads are valid for linking
after the pads has been temporarily unlocked in the linking process.
Not doing this opens up for a race condition where
pads potentially can be linked twice.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5678>
Because we treat raw audio chunks/samples as keyframes, they were interfering
with seek time adjustment.
Became apparent when the accompanying video stream was I-frame only,
for example ProRes.
Since raw audio streams can be seeked freely, it's fine to just ignore them here,
giving priority to the real keyframes in the video stream.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5674>
This is how it was documented and how it worked before the port to GstPlay.
Without this, applications expecting signals to be emitted directly
without anything running the main context will simply not receive any
signals.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5673>
The code seems to validate that the media-level fingerprint matches
the fingerprint of the previous media or of the whole session. There
is no such requirement in any RFC I found. The session-session one
is just meant to act as a fallback when there is no media-level
fingerprint.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5663>
If text width ever reached 1px, for example after resizing the output window, the overlay would stop rendering
and never return again. The 1px condition itself does not seem to make much sense here anyway.
This was a chain of events: width reached 1, so the composition was set to NULL. Then, after resizing the output window,
push_frame() was called but would not attempt to renegotiate because composition is NULL. This caused the width/height
to never be updated again, as that only happens during negotiation, so the overlay was gone for good.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5623>