Fix-up after commit e56b78417. Unclear how that could
ever have worked. Code only built because basically
everything was disabled due to the missing config.h
include.
- Include config.h so that HAVE_* defines are available.
- Fix missing variable declarations that somehow disappeared
when moving the stats gathering code block into a separate
function in a new file.
- Fix structure variable modified to match name in new function.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8211>
The matrix multiplication makes some assumption about the element
values to simplify the math with fixpoint values. If this is allowed
for the given matrices is now checked first.
Then the debug output for matrix and a comment have been fixed.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8127>
Adds a new signal to webrtcbin, to allow for placement
of an object after rtp, before sendbin. This is usable for
objects such as congestion control elements, that don't want
to be burdened by the synchronization requirements of rtpsession.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7940>
.. and other framerate values with a 1.001 denominator.
The coded framerate denominator value is a code that maps to
either 1 (for 1) or 1.001 (for 2) not a direct value.
Before, 29.97fps would be announced as 15fps because it
would calculate 30/2 instead of 30/1.001.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8226>
As part of this
- Add accessors for the stream ID and selection API based on the
stream ID
- Deprecate the old index-based APIs
- Remove playbin support
- Implement the track enable API based on stream selection
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7648>
The gst_srtp_dec_decode_buffer() function modifies the input buffer after making
it writable, so the pointer might change as well, depending on the refcount of
the buffer.
This issue was detected using a netsim element upstream of the decoder in a
WebRTC pipeline.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8198>
We can use gst_uri_from_string_with_base () to join base url
and the fragment url path.
The previous method of forming base url in update_base_url(),
by looking for the string 'manifest' or 'Manifest' is insufficient.
A query may include these string in their paths and thus an invalid
base url string will be kept.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8193>
The main difference with the WIP av1-in-mpegts mapping is that the payload data
is not startcode-escaped. Most of the rest is sensible usage of it:
* Custom AV1G (AV1 Gstreamer) registration descriptor instead of AV01
* AV1CodecConfigurationRecord is stored in the same 0x80 custom descriptor and
conforms fully to the isobmff spec (i.e. does not the HDR fields from the
provisional mpegts specification which conflict with that one).
* Data is stored as OBU
* Access Unit is the frame level (same as provisional mpegts mapping)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4442>
The reason why the STATE lock was taken was to avoid issues where we would be
adding (and activating) elements at the same time as urisourcebin would be
brought down to READY. That would cause those new elements to potentially return
ERRORS because of not-negotiated/flushing-pads
But that creates a really bad deadlock (state lock is taken to deactivate the
streaming thread which .. is currently grabbing the state lock).
Instead, we can just ignore the warning/error messages that might occur when
shutting down.
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/4075
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8162>
When a TD is being processed, it is not always pushed immediatly. Resetting
the time information lead to lost of timestamp in TU to Frame conversion. The
TU would be formed by buffer of [TD][Frame], and the timestamp taken from
the TU buffer was lost then the TD was handled.
The handling of TS should be entirely done by the 3 functions:
- gst_av1_parse_handle_obu_to_obu() (direct input to output)
- gst_av1_parse_handle_to_big_align() Reset DTS on detected TU or TD
- gst_av1_parse_handle_to_small_and_equal_align() PTS on show frame, flat DTS
Fixes: 79312357a6
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8026>
V4L2 and DRM choose different, incompatible ways to represent
tiled/compressed etc. formats. While the later uses combinations of
format fourccs and opaque, vendor/hardware specific modifiers, for the
later every such combination is a distinct new format.
Traditionally Gst implemented each of the V4L2 formats if needed.
Given the large number of tiling and compression modes, this is
quite work intensive - and often actually not needed.
In many situations Gst just needs to pass buffers from V4L2 to DRM in
the form of EGL, VK, Wayland or KMS.
Thus implement a direct translation for some V4L2 formats to DRM ones,
limited to the DMA_DRM API, allowing much quicker enablement of formats
while requiring peers to use external implementations (usually Mesa or
KMS) for tiling etc.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7355>
Use GST_VIDEO_DECODER_ERROR instead of just erroring out
unconditionally, so that the error handling behaviour is
determined by the "max-errors" property and we'll just
continue after decoding errors now instead of erroring out.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8163>
Allows reducing the initial stack size of GPU threads. Cuda should
automatically increase this value if a kernel requires a larger stack.
Can save roughly 40MB of GPU memory for a single nvh264enc instance.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8158>
The goal is equivalent to adding DMA_DRM caps to the existing raw
uploader, but creating a new dedicated uploader allows separating the
logic more easily.
Test:
- use an AMD system
- modprobe visl
- gst-launch-1.0 filesrc location=some-1080p-content.mkv ! parsebin ! v4l2slh265dec ! glimagesink
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8126>
Extended is identical to main but allows for FMO/ASO features to be
used, and prevent using CABAC. Using similar logic to "baseline",
assume that if we support main, we can also do extended.
This fixes the following fluster vectors, which otherwise would fail
when trying to link the parsebin pad.
- BA3_SVA_C
- MR6_BT_B
- MR7_BT_B
- MR8_BT_B
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8164>
Contains the following updates:
* New properties on avfvideosrc: screen-crop-*
* H265 and H265 Alpha support in vtdec and vtenc (VideoToolbox)
* ProRes support in vtenc
* New properties on vtenc elements: rate-control, data-rate-limits,
max-frame-delay
* New plugin atenc (AudioToolbox) with support for encoding AAC
* Plugin move: atdec moved from -bad to -good
* New property on osxaudio elements: unique-id
* OS X -> macOS
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8120>
1. Add some comments explaining what headers and libs are expected on
what systems
2. Only look in default incdirs if no incdir is specified
3. Require libnvbufsurface.so on Jetson when cuda-nvmm=enabled
4. Require libatomic on Jetson when cuda-nvmm=enabled
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8021>
Previously the code tried to be thread-safe by only locking when modifying the
list and leaking the old list, but this was not sufficient. When removing a log
function, its user_data would be freed but this log function and its user_data
might afterwards still be used during logging which then could lead to memory
corruption.
To avoid that, use an RW lock: get a write lock whenever modifying the list and
get a read lock whenever only using the list of log functions for logging.
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3660
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7151>
