Use VIDIOC_CREATE_BUFS ioctl to create buffers instead of VIDIOC_REQBUFS
because it allows to create buffers also while streaming.
To prepare the introduction of VIDIOC_REMOVE_BUFFERS create
the buffers one per one instead of a range of them. This way
it can, in the futur, fill the holes.
gst_v4l2_decoder_request_buffers() is stil used to remove all
the buffers of the queue.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7684>
When a datachannel within a session is removed after proper close,
reference to the error_ignore_bin elements of the datachannel
appsrc/appsink were left in webrtcbin.
This caused the bin-objects to be left and not freed until the whole
webrtc session was terminated. Among other things that includes a thread
from the appsrc.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7675>
We want to ensure the stream-collection is present on the pad (as a sticky
event) before we expose the pad.
This is more reliable since it will ensure it is present before any other event
is pushed through.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7609>
- Add mtd_meta_clear to allow specific analytics-meta to handle their clear
operation specific to their type.
- Clear mtd's attached when analytic-meta is freed. When the buffer where
analytics-meta is attached is not from a buffer pool
gst_analytics_relation_meta_clear will not be called unless we explicitly call
it in _free. This important otherwise _mtd_clear are not called and lead to
leak if embedded mtd's allocated memory
- Un-ref in transform if it's a copy
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6026>
FLUSH_STOP is meant to clear the flushing state of pads and elements
downstream, not to process data. Hence, a FLUSH_STOP should not
propagate sticky events. This is also consistent with how flushes are a
special case for probes.
Currently this is almost always the case, since a FLUSH_STOP is
__usually__ preceded by a FLUSH_START, and events (sticky or not) are
discarded while a pad has the FLUSHING flag active (set by FLUSH_START).
However, it is currently assumed that a FLUSH_STOP not preceded by a
FLUSH_START is correct behavior, and this will occur while autoplugging
pipelines are constructed. This leaves us with an unhandled edge case!
This patch explicitly disables sending sticky events when pushing a
FLUSH_STOP, instead of relying on the flushing flag of the pad, which
will break in the edge case of a FLUSH_STOP not preceded by a
FLUSH_START.
If sticky events are propagated in response to a FLUSH_STOP, the
flushing thread can end up deadlocked in blocking code of a downstream
pad, such as a blocking probe. Instead, those events should be
propagated from the streaming thread of the pad when handling a
non-flushing synchronized event or buffer.
This fixes a deadlock found in WebKit with playbin3 when seeks occur
before preroll, where the seeking thread ended up stuck in the blocking
probe of playsink:
https://github.com/WebPlatformForEmbedded/WPEWebKit/issues/1367
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7632>
H.266 NAL unit header syntax [1] is similar to H.265 NAL unit header syntax[2]:
```
H.265 H.266
+---------------+---------------+ +---------------+---------------+
|0|1|2|3|4|5|6|7|0|1|2|3|4|5|6|7| |0|1|2|3|4|5|6|7|0|1|2|3|4|5|6|7|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|F| NALType | LayerId | TID | |F|U| LayerId | NALType | TID |
+-------------+-----------+-----+ +-------------+-----------------+
Where
* F: `forbidden_zero_bit`: f(1)
* U: `nuh_reserved_zero_bit`: u(1) only H.266
* LayerId: `nuh_layer_id`: u(6)
* NALType: `nal_unit_type`: u(6) in H.265 and u(5) in H.266
* TID: `nuh_temporal_id_plus1`: u(3)
```
NAL unit types have different values:
| NALType | H.265 | H.266 |
|----------|------------------------------------|---------------------------|
| VPS | HEVC_NAL_VPS(32) | VVC_VPS_NUT(14) |
| SPS | HEVC_NAL_SPS(33) | VVC_SPS_NUT(15) |
| PPS | HEVC_NAL_PPS(34) | VVC_PPS_NUT(16) |
| IRAP | BLA_W_LP(19)..HEVC_NAL_CRA_NUT(21) | IDR_W_RADL(7)..CRA_NUT(9) |
Implementation of `h266_video_type_find` is based on `h265_video_type_find` with
next differences:
- NAL unit header syntax for H.265 and H.266
- Diff NAL unit types values
- Avoid checking nuh_layer_id is zero. H.266 conformance test suite[3] contains examples with more than one layer.
This typefind was tested with H.266 conformance test suite [3]. Also, with the help of fluster[4],
with H.264 and H.265 conformance test suites to avoid regresions. Pending test vectors to fix:
- 8b422_H_Sony_4
- DEBLOCKING_E_Ericsson_3
[1] https://www.itu.int/rec/T-REC-H.266
[2] https://www.itu.int/rec/T-REC-H.265
[3] https://www.itu.int/wftp3/av-arch/jvet-site/bitstream_exchange/VVC/draft_conformance/draft6/
[4] https://github.com/fluendo/fluster/
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7339>
I wanted to check if an element had the SINK flag and realized it was
not displayed in gst-inspect.
The clock flags were already reported as part of the "clocking
capabilities" info but best to have them explicitly listed here as well.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7641>
In my tests with the new GCC 14 compiler for Cerbero, I got the
following error:
> In file included from include/directxmath/DirectXMath.h:2275,
> from ../gst-libs/gst/d3d11/gstd3d11converter.cpp:46:
> include/directxmath/DirectXMathMatrix.inl: In function 'bool
> DirectX::XMMatrixDecompose(XMVECTOR*, XMVECTOR*, XMVECTOR*, FXMMATRIX)':
> include/directxmath/DirectXMathMatrix.inl:1161:16:
> error: variable 'aa' set but not used [-Werror=unused-but-set-variable]
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7658>
Check and generate remote reception statistics from the info stored on
internal sources, as they are stored there when running against newer rtpbin
since MR !7424
This fixes cases where statistics are incomplete when
peers send RR reports from a single remote ssrc, which GStreamer does
when bundling is enabled and other RTP stacks may too.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7425>
In some cases, decodebin3 will send us incomplete caps (not containing
codec_data), and then a GAP event, which will force a negotiation.
This segfaults due to a null pointer deref because self->input_state
is NULL.
The only possible fix is to avoid negotiating when we get incomplete
caps (to avoid re-negotiationg immediately afterwards, which isn't
supported by some muxers), but also set as much input state as
possible so that a renegotiation triggered by a GAP event can complete
successfully.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7634>
Move RB info from receiver reports into the internal source that the RR
are about, and deprecate (but retain) the old mapping where each
external source has only a single RB entry in the rtp statistics.
The old method is broken if a remote peer uses a single ssrc to send
receiver reports for more than one of our internal sources, other
as multiple RB in a single packet, or alternate RB in different reports.
In each case only the most recent entry was kept, overwriting data for
other internal sources.
In multicast scenarios each internal source may receive multiple
receiver reports from different peers. To support that, all received
RR's are now stored into a hash table indexed by the sender's SSRC,
and all RRs are placed into an array when generating statistics, so
that the information from all peers is retrievable.
The current deficient behaviour (adding RB info into non-internal RTPSources) is
deprecated but kept in order to be backward compatible, and retained
that way in the generated statistics structure.
Refs
[1] https://tools.ietf.org/html/rfc3550#section-6.4.1
Based on a patch by Fede Claramonte <fclaramonte@twilio.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7424>
