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>
[helix] can use project specific configuration in the `.helix` directory under
the project root. For GStreamer development, this can be used to:
* point the lsp (clangd) to the folder where `compile_commands.json` is
generated,
* avoid automatic header insertion,
* define gst-indent as the formatter (from the gstreamer/scripts dir).
.helix/languages.toml:
```toml
[language-server.clangd]
args = [
"--compile-commands-dir=builddir",
"--header-insertion=never",
]
[[language]]
name = "c"
auto-format = true
formatter = { command = "gst-indent", args = ["-st"] }
```
[helix]: https://helix-editor.com/
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7612>
This commit fixes two issues:
- The event must be posted *after* calling stop, otherwise a race condition can occur and the app never stops
- isFinishedLaunching and applicationDidFinishLaunching are not always synchronized, causing sometimes
a deadlock on the g_cond_wait never catching the g_cond_signal
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7593>
In order to ensure all initial events (stream-start, caps, ..) are present on
pads that we expose, those various sticky events are propagated (from parsebin
to multiqueue output, from multiqueue output to exposed pads).
The problem was that the "hack" in `urisourcebin` to inform downstream elements
that the stream is parsed data and a collection will be present was only done in
one place : a probe on the output of parsebin ... but the stream-start could
potentially have already been propagated to the output pads before that.
In order to fix that, we make sure any pending sticky stream-start event is
updated before being propagated.
Fixes#3788
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7598>
This new LCEVC encoder plugin is meant to implement all LCEVC encoder elements.
For now, it only implements the LCEVC H264 encoder (lcevch264enc) element. This
element essentially encodes raw video frames using a specific EIL plugin, and
outputs H264 frames with LCEVC data. Depending on the encoder properties, the
LCEVC data can be either part of the video stream as SEI NAL Units, or attached
to buffers as GstMeta.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7330>
This new element wraps both the base H264 decoder and lcevcdec elements into a
bin so that LCEVC decoding works with auto-plugging elements such as decodebin.
By default, the H264 decoder element with higher rank is used as base decoder,
but any particular H264 decoder can be used by manually setting the base-decoder
property.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7330>
This new LCEVC decoder plugin is meant to implement all LCEVC decoder elements.
For now, it only implements the LCEVC enhancement decoder (lcevcdec) element.
This element essentially enhances raw video frames using the LCEVC metadata
attached to input buffers into a higher resolution frame. The element is only
meant to be used after any base decoder (eg avdec_h264).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7330>
There was an override to fake an IDR as soon as a SPS/PPS
is encountered, but that's not valid, at least an i-slice is needed.
Amend the visl result, as the output is slightly more correct, not
duplicating frame_num.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7330>
This improves the h264parse element to attach LCEVC enhancement data to buffers
using the new GstLcevcMeta API. This metadata will eventually be used downstream
by LCEVC decoders to enhance the RAW video frame.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7330>
This new metadata API allows elements to attach LCEVC enhancement data to video
buffers. Usually, the video parser elements are charged to parse the LCEVC
enhancement data from SEI Nal units (Supplemental enhancement Information).
However, other elements such as demuxers can also use this API if the LCEVC
enhancement data of the video is stored in a separate stream in the container.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7330>
Previously urisourcebin only allows stream-collections messages from adaptive
demuxers or sources to be posted.
This commit also allows the case where they come from a single parsebin. We
still want to prevent it in the case where they are multiple parsebins, since
that would require some form of aggregation to show a single/unified collection.
In order to avoid a regression with uridecodebin3 behavior, we also implement
support for GST_QUERY_SELECTABLE, so that uridecodebin3 can figure out whether
it should let GST_MESSAGE_STREAM_COLLECTION flow upwards (because app/user could
react on it) or whether it drops it in order for decodebin3 to do the collection
aggregation and posting.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7597>
The presence (or not) of a collection on an input will determine whether events
will be throttled so that there are only forwarded when that input gets a valid
collection.
Therefore the input lock should be used.
In addition to that, we want to ensure that the application/user has a chance to
reliably (i.e. synchronously) specify what streams it is interested in by
sending a GST_EVENT_SELECT_STREAMS.
But we cannot allow anything to go forward until that message posting has come
back, otherwise we run in various races.
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3872
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7594>
