webrtcbin: Avoid access of freed memory

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2256>

mp4mux: fix spelling

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2241>

va: h264enc: Do not let L0 number surplus the HW limitation.

The algorithm to calculate the L0/L1 number may let the L0 number
surplus the HW limitation. We should ensure that limitation after
that calculation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2202>

videoscale: Add test to ensure that non-scaled metas are kept

Just make sure that we're not breaking non-scaled metas

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1630>

videoscale: Add test for transform_meta function

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1630>

videoscale: Don't copy scaled metas

Returning TRUE from the `transform_meta` function tells
GstBaseTransform to copy the meta into the new buffer. If videoscale
has already transformed a meta by scaling it, it should always return
FALSE to avoid duplicating the meta.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1630>

x11: Fix unused variable warnings

These are emitted when XInput 2 is not available.

```
../subprojects/gst-plugins-base/sys/ximage/ximagesink.c: In function ‘gst_x_image_sink_handle_xevents’:
../subprojects/gst-plugins-base/sys/ximage/ximagesink.c:696:29: warning: unused variable ‘state’ [-Wunused-variable]
  696 |   GstNavigationModifierType state = GST_NAVIGATION_MODIFIER_NONE;
      |                             ^~~~~
../subprojects/gst-plugins-base/sys/ximage/ximagesink.c:694:35: warning: unused variable ‘touch_frame_open’ [-Wunused-variable]
  694 |   gboolean pointer_moved = FALSE, touch_frame_open = FALSE;
      |                                   ^~~~~~~~~~~~~~~~
../subprojects/gst-plugins-base/sys/xvimage/xvimagesink.c: In function ‘gst_xv_image_sink_handle_xevents’:
../subprojects/gst-plugins-base/sys/xvimage/xvimagesink.c:427:35: warning: unused variable ‘touch_frame_open’ [-Wunused-variable]
  427 |   gboolean pointer_moved = FALSE, touch_frame_open = FALSE;
      |                                   ^~~~~~~~~~~~~~~~
../subprojects/gst-plugins-base/sys/xvimage/xvimagesink.c:426:29: warning: unused variable ‘state’ [-Wunused-variable]
  426 |   GstNavigationModifierType state = GST_NAVIGATION_MODIFIER_NONE;
      |                             ^~~~~
```

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2093>

mingw: Remove deprecated section in example cross file

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2080>

meson: Add PYTHONPATH to load GDB helper module

Meson generates a gdbinit file that will automatically load gstreamer
script. However that script uses a helper python module that needs
PYTHONPATH to be pointing into the right location in the source
tree to be able to find gst_gdb.py.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1796>

multiudpsink: allow binding to IPv6 address

When the sink is configured to create sockets with an explicit bind
address, then the created socket gets set to the udp_socket field
irregardless of whether the bind address indicated that the socket
family should be IPv4 or IPv6.  When binding to an IPv6 address, this
results in the following error:

gstmultiudpsink.c:1285:gst_multiudpsink_configure_client:<rtcpsink>
error: Invalid address family (got 10)

This patch adds a check of the address family being bound to and sets
the created socket to used_socket or used_socket_v6, accordingly.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1551>

rtphdrextsdes: fixup test trying to g_free a local variable

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2235>

hls/m3u8: Fix starting segment for live playlist

RFC 8216 6.3.3 "Playing the Media Playlist File" : states that for live media
playlists "the client SHOULD NOT choose a segment that starts less than three
target durations from the end of the Playlist file"

This is an off-by-one error. Since we are looking for the "index" of the
segment, we need to subtract 1 from the searched position.

Ex: For a playlist with 12 entries, we want to start playback on the 9th segment
... which is at index 8.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2259>

test: Add test cases for the H265 bitwriter.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1831>

codecparsers: bitwriter: Add the common bit writer functions for H265.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1831>

va: vpp: Update the colorimetry and HDR fields when fixup src caps.

When we fixup src caps, the current way of handling the HDR fields is not
correct.
1. We trim the HDR fields only when the input caps is not a subset of the
fixup src caps. But in fact, the input caps with HDR fields such as the
"mastering-display-info" can possibly be the subset of the fixup src caps,
if they have all same other fields.
2. We always copy the colorimetry from input caps to src caps if it is
absent. But when hdr-tone-mapping is enabled, the HDR->SDR conversion makes
the colorimetry change. We should use downstream's setting, or just use the
default colorimetry of SDR.

We changes to:
1. If hdr-tone-mapping is enabled, we trim all HDR fields and add a correct
colorimetry.
2. Copy colorimetry from input if it is still absent.
3. Consider the subset replacement.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2244>

validate: Do not print Known error tests

Those are passing tests in practice and it doesn't give any useful information

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2264>

tests: convertscale: Do not log EOS events in validateflow

We already checked that we are getting an EOS before going back to ready
and adding the EOS at the end of the .validateflow is racy as the ERROR
message might reach the bus before it is logged.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2264>

ci: Trigger cerbero on user project if a branch with the same name exists there

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2264>

docs:examples: Fix building iOS examples

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2264>

validate:pipeline-monitor: Minor cleanup setting unused variable

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/898>

videoconvertscale: Add properties to disable scaling/converting in videoconvert/videoscale

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/898>

validate: scenario: Add a 'allow-error' configuration

This new configuration allows the scenario to keep running even
after receiving an "error" message on the bus.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/898>

videoconvertscale: Don't claim we can support any kind of memory

Since d0133a2d11566ff4c0cded7af8dfdff0046e0e8b "videoconvert: Allow
passthrough for ANY caps features" videoconvert will always claim that
it supports any kind of memory which is true in very specific case (when
it is running in passthrough mode). To get elements that autoplug
converters depending on the caps running in the pipeline (like
autovideoconvert), we need to have converters no lie about what they can
do when queried `accept_caps` or `query_caps`.

This still accepts any caps feature as before but it introduces
a restriction in the way we handle memory capsfeatures.

We keep previous behaviour in videoconvert and videoscale.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/898>

Introduce the videocolorscale element

Now that videoconvert and videoscale's are both based on
GstVideoConverter and are using the exact same code, it makes much more
sense to have one element doing the two operation, and it can be
more efficient in some cases (one single path for both operations).

This removes the `videoscale` and `videoconvert` plugins but keeps the element
but makes them also do both operations (adding some APIs to each element).

There is a small change in API for the `videoscale:dither` property which
was previously a totally unused boolean, it is now an enum and is used.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/898>

qsv: Add GPU description to element long-name on Windows

Would be useful for a multi-Intel-GPU system

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2249>

hls: Relax webvtt checks

If no hour field is present (which is allowed), the remaining data can be less
than 15 character.

Fix time translation failures if the hour field wasn't present

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2248>

vapostproc: Notify if hdr-tone-mapping property changes.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2247>

gstleaks: log the number of alive objects

Can help checking if the number of alive objects is growing over time or stays stable.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2245>

va: Expose a couple va surface/buffer methods.

And use them in va plugin.

Also, set as default logging category the vadisplay one.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2077>

rtpmanager: Move some duplicated constant and helper function to a single place

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2132>

gstreamer: Document various caps for the reference timestamp meta

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2132>

rtpbin/rtpjitterbuffer: Don't parse RTCP SRs twice unless needed

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2132>

rtpjitterbuffer: Add property to throttle handling of RTCP SR / NTP-64 syncing

This proxies the "rtcp-sync-interval" property of rtpbin.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2132>

rtpsession: Handle RTCP-SR-REQ (RFC6051) RTCP feedback message

This causes an RTCP SR to be sent at the earliest possible time.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2132>

rtpbin/rtpjitterbuffer: Allow syncing to an SR without CNAME if the CNAME is already known

The RTCP SR packet might be without SDES in case of a reduced-size RTCP
packet. For syncing purposes the CNAME is needed but it might be known
already from an earlier RTCP packet or out of band, via the SDP for
example.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2132>

rtpbin/jitterbuffer: Use inband 64-bit NTP timestamps according to RFC6051 for faster synchronization

When signalled via the caps that the header extension is used, it will
be read and used in the same way as the RTP/NTP time mapping from RTCP
SRs.

If the CNAME of the stream's SSRC is provided out of band via e.g. the
SDP then this allows streams to be synchronized immediately on the first
packet instead of having to wait for the first RTCP SR to arrive.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/383

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2132>

rtpsession: Only add send latency to the running time if it is actually known

Otherwise we can't know the running time yet if rtcp-sync-send-time is
set, and have to wait until the latency is known later.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2132>

rtpsession: Update 64-bit NTP header extensions with the actual NTP time in senders

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2132>

rtpmanager: Add header extension implementation for the 64-bit RFC6051 NTP header extension

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2132>

sdp: Parse the RFC5576 Source-specific media SDP attributes into caps

The format of the caps fields is
    ssrc-(SSRC_VALUE)-(ATTRIBUTE_NAME)=(ATTRIBUTE_VALUE)
.

Parsing of the attributes from the caps into the SDP is not implemented
as this depends not only a single stream's caps but on the whole rtpbin
configuration.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2132>

rtsp-server: Add RFC5576 Source-specific media attribute to the SDP media for signalling the CNAME

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2132>

dashsink: Unlock when failed to get content

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2242>

gstglmixerbin: minor refactor of _find_element_pad_template

As suggested by @slomo, make the loop more readable and prevent returning
a garbage value to the caller from the previous implementation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2232>

gstav1parse: fixup various possible logical errors

Found via an analyzed build for Clang. Specifically we had:

gstav1parse.c[1850,11] in gst_av1_parse_detect_stream_format: Logic error: The left operand of '==' is a garbage value
gstav1parse.c[1606,11] in gst_av1_parse_handle_to_small_and_equal_align: Logic error: The left operand of '==' is a garbage value

Also a couple of false-positives:

gstav1parse.c[1398,24] in gst_av1_parse_handle_one_obu: Logic error: Branch condition evaluates to a garbage value
gstav1parse.c[1440,37] in gst_av1_parse_handle_one_obu: Logic error: The left operand of '-' is a garbage value

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2230>

Always define ENABLE_NLS

GLib guarantees libintl API is always available, provided by
proxy-libintl as last resort. GLib itself unconditionally define
ENABLE_NLS.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2028>

Delete unused i18n headers

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2028>

Replace gst-i18n-*.h with gi18n-lib.h

GLib guarantees libintl is always present, using proxy-libintl as
last resort. There is no need to mock gettex API any more.

This fix static build on Windows because G_INTL_STATIC_COMPILATION must
be defined before including libintl.h, and glib does it for us as part
as including glib.h.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2028>

gst-play: Do not split translatable string

Concatenating N_() strings does not work with gi18n.h macro, was working
only with GStreamer's redefinition.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2028>

cuda: Fix typo in gst_cuda_buffery_copy_type_to_string

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1472>

cuda: Add a `gst_cuda_context_new_wrapped` constructor

So that application or people implementing cuda element from
outside GStreamer itself can ensure that their contexts are used
by GStreamer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1472>

cuda: Factor out a public GstCUDA library

So applications and elements implemented outside GStreamer can reuse
our infrastructure

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1472>

nvenc: Give more details about errors

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1472>

video: Fix possible overrun when iterating comp[] array

Fix 2 iterations that can overrun the array if the number of component is
equal to the size of the array.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2239>

Revert "video-frame: avoid possible out of bound memory access"

This reverts commit c4255f08f787ff6a41504b538fa4c2e911d40ec7.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2239>

meson_options: declare glib and gobject checks at top level

This makes it easier to specify this features for all subprojects.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2238>

meson: Add -Wl,-rpath,${libdir} on macOS

We made the gstreamer installation prefix relocatable by picking up
plugins relative to the location of libgstreamer-1.0.dylib, similar to
how it's done for Windows:

https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1627

This had a lot of side-effects:

https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1051
https://gitlab.freedesktop.org/gstreamer/cerbero/-/issues/363
https://gitlab.freedesktop.org/gstreamer/cerbero/-/issues/371
https://gitlab.freedesktop.org/gstreamer/cerbero/-/issues/362

A partial fix for the cerbero side of these was:

https://gitlab.freedesktop.org/gstreamer/cerbero/-/merge_requests/807

However, this relied on the consumers knowing that they need to add
`LC_RPATH` entries to the libdir of the prefix. This is done
automatically by build systems like Meson, but not by others, such as
Autotools, CMake, Cargo, XCode, etc. For those, we need to add the
RPATH entries to the gstreamer-1.0.pc file.

This also has the side-effect of fixing the loading of gstreamer rust
plugins on macOS:

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1159

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1149

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2218>

video-frame: avoid possible out of bound memory access

Although the components' initialization code would fill in -1 to all
unset components, make the code a bit more defensive and check for an
index bound first.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2231>

tests: dash_mpd: fix linker issues with non-optimizing compilers

undefined reference to `download_request_take_buffer'

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2117#note_1344646

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2228>

gst_plugin_load_file: force plugin reload if diff filename

If a file includes a new version of a plugin that exits in the
registry, the output of gst-inspect is incorrect. The output has the
correct version but incorrect filename, and element description.

This seems to have also fixed some documentation issues.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1344>

docs: design: fix up some document titles

These show up in the index.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2222>

docs: design: add adaptive streaming design docs to sitemap

.. and some small additions to make it clearer what exist
and what's new.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2222>

d3dvideosink: Remove outdated comment

d3dvideosink is not a primary video sink element

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2223>

h264decoder,h265decoder: Do not hold codec_data buffer

... also do not need to extract it from caps.

Decoder uses codec_data only once per set_format() and
baseclass will extract codec_data buffer already.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2212>

d3d11decoder: Do not preallocate texture using downstream d3d11 buffer pool

Our decoder implementation does not use downstream d3d11 pool for
decoding because of special requirement of D3D11/DXVA. So preallocation
using the downstream buffer pool will waste GPU memory in most cases.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2211>

gstdevicemonitor: added cleanup of signal handlers and hidden providers list

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2182>

New HLS, DASH and MSS adaptive demuxer elements

This provides new HLS, DASH and MSS adaptive demuxer elements as a single plugin.

These elements offer many improvements over the legacy elements. They will only
work within a streams-aware context (`urisourcebin`, `uridecodebin3`,
`decodebin3`, `playbin3`, ...).

Stream selection and buffering is handled internally, this allows them to
directly manage the elementary streams and stream selection.

Authors:
* Edward Hervey <edward@centricular.com>
* Jan Schmidt <jan@centricular.com>
* Piotrek Brzeziński <piotr@centricular.com>
* Tim-Philipp Müller <tim@centricular.com>

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2117>

v4l2videodec: copy colorimetry values to output_state caps

This is to avoid transcoding negotiation fail between v4l2h265dec
and v4l2h264enc caused by colorimetry mismatch.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2192>

h264parse: Include coded-picture-structure info in caps

This reverts commit 652773de366a98e155332bf8ffb83ee5e6bea9e5 and
modifies it to rename the caps field name to coded-picture-structure.

It was previously removed because it confuses the decoder and we didn't
have a valid use case for including it in the encoded caps at this
stage. We now do have such a use case but still don't want to confuse
the decoder, so the field is renamed.

However, it is still not accurate without looking at the SEI picture
structure of each frame, so it was named coded-picture-structure. If its
value is "frame" it is most likely progressive, if it's "field" it is
most likely interlaced or mixed.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2177>

libav: fix frame leak on negotiation error

The function owns a reference on the frame. Drop it if negotiation
failed as we are already doing for the other error cases.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2196>

decodebin3: fix collection leak

get_merged_collection() returns an owned stream collection and was
leaked in the else block.

Fix leak when running:
GST_TRACERS=leaks GST_DEBUG="GST_TRACER:7,leaks:6" gst-play-1.0 --use-playbin3 test.mkv

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/954>

va: h264enc: Fix the forgotten unlock() when getting device path property.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2215>

qsvdecoder: Use system memory for non-d3d11 downstream

QSV's memory download implementation seems to be more faster
than our implementation

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2206>

qsv: Add H.265 decoder

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2206>

qsv: Include single gstva.h

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2206>

qsvh264dec: Don't register element if it's not supported by device

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2206>

qsvh264dec: Add support for packetized format

Handle avc/avc3 packetized formats

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2206>

qsv: Move debug init into each source file

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2206>

qsvdecoder: Reset next_task_index per init_session()

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2206>

d3d11decoder: Copy HDR10 related caps field manually

If negotiate() is called from the set_format() chain, sinkpad may not
hold caps yet, so baseclass cannot copy it over to srcpad caps.
Copy them manually.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2203>

qsvencoder: Rename property name i-frames to idr-interval

... and clarify the meaning of the value for each h264 and h265
encoder since the usage of mfxInfoMFX::IdrInterval is different
per codec.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2204>

qsvh265enc: Write HDR10 metadata SEI

If device supports HDR10 metadata API, write the information
into bitstream per IDR.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2204>

gst_plugins_cache.json: Update va plugins.

* Update conditionally-available attribute in vah264enc properties.
* Update GstVaFeature usage.
* Update GstVaEncoderRatecontrol usage.
* Update vapostproc classification.
* Add `add-borders` property in vapostproc.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2169>

vah264enc: Use GstVaFeature enum for enabling MBBRC.

GstVaFeature is a common enumeration for auto/disabled/enabled kind of
property setting.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2169>

vah264enc: Move rate-control enum to encoder class helper.

Since it's a common enumeration used, as user setting property, for
most of codecs.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2169>

vah264enc: Add device-path property.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2169>

gstleaks: fix pthread_atfork return value check

pthread_atfork() returns 0 on success.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2195>

gst-python: fix build on systems without C++ compiler

Fix the following build failure on systems without C++ compiler:

The following exception(s) were encountered:
Running "/home/autobuild/autobuild/instance-0/output-1/host/bin/i686-buildroot-linux-gnu-g++ --version" gave "[Errno 2] No such file or directory: '/home/autobuild/autobuild/instance-0/output-1/host/bin/i686-buildroot-linux-gnu-g++'"

Fixes:
 - http://autobuild.buildroot.org/results/eebf65036f79d21d347714d62afecd0108393308

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2200>

cudautils: fix critical typo in gst_cuda_buffer_fallback_copy

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2194>

playbin3: fix missing lock when unknown stream type in pad-removed cb

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2193>

decodebin3: Don't duplicate stream selections

Make sure that the requested stream selection isn't identical to the current
one. If that's the case, just carry on as usual.

This avoids multiple `streams-selected` posting ... when the selection didn't
change.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2185>

x264enc: Don't try to fixate ANY allowed caps

Instead fall back to the template caps.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2197>

appsrc: fix annotations

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2188>

rtpbasepayload: Don't write header extensions if there's no corresponding input buffer for the packet

The GstRTPHeaderExtension API requires the input buffer to exist.

This can happen if the output packet is generated e.g. from a caps or
tags event like in the case for rtpgstpay.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2198>

tests: rename 'icles' subdir to be more descriptive

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2178>

qtdemux: fix leak of channel_mapping

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2179>

tsdemux: Fix AC-4 detection

This regression was introduced by
https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1102
and has been present since 1.18

* Stream Type 0x06 is defined in the base mpeg-ts specification as Private PES
Packets. Determining the content should be solely based on descriptors found
within the PMT.
* This was abused in that commit by defining a "bluray-only" stream type for AC4
: `ST_BD_AUDIO_AC4`
* This should be entirely handled in the regular private pes handling further
down in the code

Fixes #1154

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2175>

docs: Update for Windows specific parts

* glimagesink is not a recommended one on Windows
* Remove directdrawsink section
* d3dvideosink is legacy and should not be recommended
* Add d3d11videosink part
* directsoundsink should be deprecated
* Add wasapisink/wasapi2sink part

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2144>

ptpclock: Fix wrong condition order from last commit

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2171>

gst-rtsp-server: minor spelling fixes

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2170>

va: h264enc: Correct the hierarchical level name for pyramid-b frames.

The current way names the level by the number of B frames it contains, the
less it contains, the higher level it is. So the non ref B frames are in the
lowest layer and the B frames in the highest level refer to I/P frames.
But the widely used way is just the opposite, the ref B frames are in the
lower level and non ref B frames are at the highest level.
The is just a terminology change, and does not have any effect for compression
result and quality.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2149>