gst-play: update translated string

Make it more consistent with the other option descriptions.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4033>

qtmux: Implement writing of `av1C` version 1 box

Version 0 is ancient and not specified in any documents. Take it
directly from the `codec_data` if presents or otherwise try to construct
a reasonably looking `av1C` box.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4027>

qtdemux: Drop av1C version 0 parsing and implement version 1 parsing

The av1C box is optional so dropping parsing does not break anything
fundamentally, and there seems to be no historical record how version 0
even looks like while the comments and the parsing disagreed with each
other.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4027>

wpe: Logging fixes for the WebExtension

Using logging macros without a `GST_CAT_DEFAULT` in scope leads to critical
warnings.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4022>

appsrc: Don't chain up `BaseSrc::negotiate()`

If we have caps then we can only set exactly those caps, if we have no
caps yet then negotiating anything is not very meaningful because the
caps are defined by the application and not downstream.

Avoids, among other things, an unnecessary allocation query and spurious
useless caps being set before the first buffer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4020>

gst-play: Don't force accurate seeking

This crept in several years ago sadly :(

The usage of accurate seeking should be reserved to use-cases where it is
essential that we seek to that position. This should not be the default.

There is a new option `--acurate-seeks/-a` to be able to force that.

Furthermore, if accurate seeks aren't required, a player should be using the
GST_SEEK_FLAG_KEY_UNIT flag to seek to the closest keyframe and provide the most
reactive experience.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4017>

uridecodebin: Set source element to READY before querying it

Generating the source element is done when uridecodebin is doing the
READY to PAUSED state change, so it is reasonable to set the new source
element to that state.

This also allows detecting early failures with backing libraries or
hardware (checks done in NULL->READY).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4016>

ccconverter: don't debug a potentially freed filter caps

Fixes a use-after-free

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4013>

d3d11videosink: Fix rendering on external handle

Partial revert of the commit 068a5c1053ae05b3c7747243948447e764d25aa6.
That introduced size mismatch between internal and external HWND

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4010>

vah265enc: Use helper to update properties.

This is a continuation of the original patch by
 Mengkejiergeli Ba <mengkejiergeli.ba@intel.com>

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4009>

av1parser, h265parser: Fix some code defects

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4003>

webrtc: Calculate the jitter for remote-inbound-rtp stats

Populate the clock-rate in the internal stats structure, so
it can be used by the _get_stats_from_remote_rtp_source_stats()
method to calculate remote receivers' jitter.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3994>

webrtcbin: Report full codec-stats for source pads

Use the current caps for webrtcbin srcpads, as received_caps
are only stored for sink pads based on incoming caps events.

Makes it so that webrtcbin stats reports contain fuller
codec information.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3994>

msdkav1enc: fix the category for msdkav1enc debug

The msdkav1enc debug initialized with gst_msdkav1dec_debug,
fix this.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3991>

qtdemux: Don't emit GstSegment correcting start time when in MSE mode

When using qtdemux in a pipeline that should only work as a pure demuxer (not
for actual playback), qtdemux shouldn't emit new GstSegments to correct
the start time (jump to the future) to ensure that the user experiences no
playback delay. By doing so, it's generating the wrong segments when an append
of data from the past happens. When that happens, downstream elements such as
parsers (eg: aacparse) may clip those buffers laying before the GstSegment and
create problems on the GStreamer client app (eg: WebKit).

Getting buffers clipped out because of the wrong GstSegments started becoming
a problen when this commit was introduced:

ab6e49e9cc audioparsers: add back segment clipping to parsers that have lost it

This clipping makes test DASH shaka 35 from MVT tests[1] to fail in
WebKitGTK/WPE (at least) and can potentially cause a number of other problems
in the WebKit Media Source Extensions (MSE) code.

Note that this new behaviour of not emitting new GstSegments only makes sense
when qtdemux is being used as a pure demuxer and not as part of a regular
pipeline. This is why the variant field has been added. When equal to
VARIANT_MSE_BYTESTREAM, it will make qtdemux behave differently in push mode,
taking decisions that meet the expectations for an MSE-like processing mode.
This kind of tweaks have been done in the past for MSS streams, for instance.
That code has been refactored to use VARIANT_MSS_FRAGMENTED now, instead of
its own dedicated boolean flag.

Co-authored by: Alicia Boya García <ntrrgc@gmail.com>

...who suggested to use "variant: mse-bytestream" in the caps to identify that
mode, as proposed in her unmerged patch:

https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/467

[1] https://github.com/rdkcentral/mvt

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3990>

tests: Move srtp test known issue def to the 'check' testuite

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3986>

validate:launcher: try to bne more explicit about where to add known failure

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3986>

validate:launcher: Handle known return code issue in none validate based tests

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3986>

codectimestamper: Fix timestamping on sequence update

... and enhance debug logging.

Keep internal timestamp offsets on drain, no reason to reset them

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3987>

codecparsers: {h264,h265}bitwriter: Remove redundant condition checks

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3978>

validate: add elements_srtp.test_play_key_error to flaky test list

See https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1777

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3970>

tests: mark elements_srtp.test_play test as flaky

https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1777

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3970>

amfav1enc: Set stream-format on caps

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3961>

qsvav1enc: Set stream-format on caps

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3961>

avviddec: change AV_CODEC_CAP_AUTO_THREADS->AV_CODEC_CAP_OTHER_THREADS

This fixes a compile error with recent upstream FFmpeg.

The AV_CODEC_CAP_AUTO_THREADS was deprecated and renamed to
AV_CODEC_CAP_OTHER_THREADS in FFmpeg upstream commit
7d09579190de (lavc 58.132.100).

The AV_CODEC_CAP_AUTO_THREADS was finally removed in FFmpeg upstream
commit 10c9a0874cb3 (lavc 59.63.100).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3964>

glvideomixer: Keep a reference to the underlying pad

There was cases where we were accessing the pad while it was already
destroyed. Ensure it can't happen by owning a ref on it.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3963>

vtenc/vtdec: Fix typo in RGBA64_LE availability define

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3962>

vtenc: Disable ARGB/RGBA64 caps on M1 Pro/Max with macOS <13

Fixes #1358.
Passing ARGB64/RGBA64 to vtenc caused the encoding to fail
when running on M1 Pro/Max variants with macOS 12.x, so let's
remove these formats from caps when such scenario is detected.
This issue appears to have been fixed OS-side in macOS 13.0.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3962>

audio: channel-mix: Fix channel count limit to be able to equal 64

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3958>

qtdemux: Handle moov atom length=0 case by reading until the end

Previously it would fail to demux the file by trying to read G_MAXUINT64
bytes.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3938>

qtdemux: Fix guint vs gsize type confusion

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3938>

vtdec: Fix not waiting for async frames when flushing

This was causing incorrect output when seeking, especially
when used with a multithreaded source like `videotestsrc n-threads=2`.
It should now correctly wait for frames still being processed by VT
while vtdec is flushing.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3937>

win32ipcutils: Add missing include

We are using std::isspace() with one parameter. That function is defined
in the cctype header.

```
win32ipcutils.cpp(34): error C2672: 'std::isspace': no matching overloaded function found
win32ipcutils.cpp(34): error C2780: 'bool std::isspace(_Elem,const std::locale &)': expects 2 arguments - 1 provided
```

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3936>

pad: Don't leak user_data in gst_pad_start_task

When the task already exists, we forgot to free the passed `user_data`.
This wasn't an issue for most C code, which doesn't pass a
`GDestroyNotify`, but bindings such as gstreamer-rs do!

That said, allocating a trampoline in gstreamer-rs just for it to get
thrown away again is awkward. Maybe we need a `gst_pad_resume_task`?

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3925>

av1parser: Don't consider unknown metadata OBUs a bitstream error

Just don't parse them.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1774

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3924>

subparse: Properly forward segment seqnum

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3917>

closedcaption: Don't leak caps event

All events that we handle should be unreffed

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3906>

meson: Allow sysdeps to be forced as fallback subprojects

The original code was too complicated; likely created before the
provide section existed for wraps:

https://mesonbuild.com/Wrap-dependency-system-manual.html#provide-section

Now you can do --force-fallback-for=pygobject and it'll actually work.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3905>

validate-scenario: fix g-i warning in annotation

gst-validate-scenario.c:2485: Warning: GstValidate: missing ":" at column 28:

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3842>

gstreamer: bin: Don't unlock unlocked mutex in gst_bin_remove_func()

Calling `g_mutex_unlock(mutex)` leads to an undefined behavior if the
mutex is not locked by the current thread.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3910>

webrtc examples: Use webrtc.gstreamer.net

Actually just a CNAME to webrtc.nirbheek.in for now, but it allows
replacement / hosting without my involvement, so reduces the bus
factor.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3892>

nvencoder: Fix min buffers parameter of allocation query in auto GPU mode

At the time when propose_allocation() get called, encoder session
would not be initialized yet.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3891>

gl/cocoa: Return a strong ref to the parent GstGLContext

If the GstGLCAOpenGLLayer was initialized via a parent
context, make sure to ref the context before returning it
from getGLContext as all callers will unref it.

Follow up to !3729

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3885>

va: Avoid the array index overflow when filling 8x8 scaling list.

The VA API has not defined the scaling list entries for U/V planes
for the 4:4:4 stream. In fact, we do not meet the 4:4:4 format output
for H264 so far, and scaling list is not used frequently, so we just
print out some warning and ignore these scaling list values.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3877>

Fix gstreamer-validate-1.0 dependency name

The gst-devtools project generates gstreamer-validate-1.0.pc, this
must match the dependency in gst-editing-services for detection
to work properly.

Fixes:
Run-time dependency gst-validate-1.0 found: NO (tried pkgconfig and cmake)

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3872>

vtenc: Disable HW acceleration for interlaced ProRes

Due to a bug in the VT API, attempting to encode interlaced content
with ProRes results in an error, halting the pipeline instead of
gracefully falling back to software encoding.
Should be removed in the future if Apple ever fixes this issue.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3878>

aom: Include stream-format and alignment in the AV1 caps

The decoder does not work with arbitrary alignment and annexb stream
format and the encoder can give the information that it outputs
obu-stream/tu to downstream.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3873>

nvvp9dec: Fix return value

It should return GstFlowReturn value, not boolean

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3874>

rtspsrc: Also consider "Method Not Valid In This State" error in broken control URL handling workaround

Some servers send a 455 error instead of any reasonable error when using
a correctly constructed control URL.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3864>

ci: rebuild windows image

Needed for the backport of https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3805
to take effect, once cerbero is made to use the new image.

Should fix errors like

cp: will not create hard link `./cerbero-sources/sbc-1.3' to directory `./cerbero-sources/bzip2-1.0.8'

and similar.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3869>

rtpptdemux: set different stream-id on each src pad

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3866>

rtpssrcdemux: set different stream-id on each src pad

All the RTP src pads were sharing the same stream-id while each actually
carry a different stream.

This was causing problem for example when funneling the streams together
and then trying to split them using 'streamiddemux'.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3866>

gst-env: Handle installing python modules to dist-packages

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3848>

BaseSrc: fix transfer annotation for fixate virtual method

The fixate virtual method has the same semantics as gst_caps_fixate(),
so the caps parameter must be marked as (transfer full).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3839>

validate: Fix gst_validate_execute_action annotation

The action passed in is transfer full

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3838>

gldisplay: Mark `gst_gl_display_create_context()` `other_context` parameter as nullable

See https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/-/issues/438

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3835>

uritranscodebin: Fix unref of NULL

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3837>

mpegpsdemux: Ignore DTS if PTS < DTS

It's possibly timestamp rollover case. But PTS < DTS is already
invalid case anyway.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3836>

BaseSrc, PushSrc: add nullable annotations to virtual methods

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3833>

webrtc_sendrecv.py: Fix PEP8 warnings in CI lint

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3816>

webrtc_sendrecv.py: Handle LATENCY messages

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3816>

webrtc_sendrecv.py: Add bus message handling

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3816>

webrtc_sendrecv.py: Add support for using H264 encoding

Currently only works when we are creating the offer or the offer only
contains H264.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3816>

webrtc_sendrecv.py: Use sine wave for audio instead of red-noise

Makes it easier to notice when there's packet loss or other audio
distortion.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3816>

ci/windows: Remove cargo-vendor sources from the cerbero source cache

Copying the source cache to a running Cerbero CI job on Windows causes
strange / corrupted errors that talk about hard links like:

++ cp -a C:/cerbero/cerbero-sources .
cp: will not create hard link `./cerbero-sources/gstreamer-1.0/subprojects/gst-plugins-good/sys/osxaudio' to directory `./cerbero-sources/cargo-c-0.9.12/cargo-vendor/adler/benches'
[...]

So, delete it. The vendored sources are cached by cargo in its own
cache inside CARGO_HOME, which is build/rust/cargo/ and is preserved
in the image. We won't need to redownload any of this.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3821>

wasapi2src: Fix loopback capture on Windows 10 Anniversary Update

... or older. Work around an OS bug that loopback capture
device doesn't notify event.

Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1738
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3820>

aggregator: Warn about the right segment's format

We were checking the head segment here, not the current segment.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3819>

aggregator: Always lock aggpad around update_time_level

`aggpad->segment` is protected by the `aggpad`'s object lock. We need to
take the lock before calling `update_time_level`.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3819>

rtsp: gstrtspurl: gst_rtsp_url_get_request_uri: fix incorrect scheme for tls transport methods

gst_rtsp_url_get_request_uri returns rtsp://... url when requested url is rtsps://, this is not
in accordance with https://www.rfc-editor.org/rfc/rfc7826.html#section-19.2.
This also impedes setting up a rtsps session with a live555 rtsp server.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3818>

sdp: gstmikey: gst_mikey_message_to_caps: extract salt and append to srtp-key

A KEMAC can hold both key and salt, so if salt is present, we should extract it
and append it to the srtp-key cap because libsrtp expects the srtp-key to hold
the concantenated srtp master key and salt.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3818>

sdp: gstmikey: gst_mikey_message_to_caps: extract ROC from first crypto session

We need the ROC to decrypt a SRTP stream as the ROC is part of the AES IV.
So look for first crypto session, from which we can get the ROC and find corresponding
crypto policy.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3818>

gst: Fix gst_type_find_peek() return value annotation

It's not possible to annotate a in-parameter for a return value array as
the array length. Both are assumed to have the same direction and the
current annotation causes the size parameter to be considered an out
parameter.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3814>

sdp: Fix gst_mikey_payload_key_data_set_interval() array length annotation

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1744

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3814>

v4l2h264dec: Fix Raspberry Pi4 will not play video in application

Ensure object v4l2object->pool will be released by
correctly releasing the temporary thread-safety lock

Fixes issue #1729

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3815>

vulkan: memory: Flush non coherent memory after write.

Spec 7.1.3:

If a memory object does not have the VK_MEMORY_PROPERTY_HOST_COHERENT_BIT
property, then vkFlushMappedMemoryRanges must be called in order to guarantee
that writes to the memory object from the host are made available to the host
domain, where they can be further made available to the device domain via a
domain operation. Similarly, vkInvalidateMappedMemoryRanges must be called to
guarantee that writes which are available to the host domain are made visible to
host operations.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3817>

nvcodec: Log readable errors when initializing CUDA

It is really difficult for people to figure out why nvcodec has
0 features. Even the debug log is cryptic. Also make sure the errors
go to the ERROR log level, which is more likely to be enabled by
default.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3813>

nvcodec: Fix reporting of CuDeviceGetCount error

cuda_ret is was always going to be CUDA_SUCCESS in the error log.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3813>

redenc: fix setting of extension ID for twcc

1 was previously hardcoded in, and the bug went under the radar because
webrtcsink hardcodes the number too.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3812>

multiqueue: Handle use-interleave latency live pipelines

Due to the dynamic nature of multiqueue, when `use-interleave` is used we can't
report a maximum tolerated latency (when queried) since it is calculated
dynamically.

When in such live pipelines, we need to make sure multiqueue can handle the
lowest global latency (provided by this event). Failure to do that would
result in not providing enough buffering for a realtime pipeline.

Fixes #1732

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3810>

avvidenc: Don't take ffmpeg timestamps verbatim but only use them to calculate DTS

The ffmpeg timestamps are inaccurate and only in framerate granularity.
To avoid generating inaccurate output timestamps, especially with
variable framerate streams, only use the ffmpeg timestamps for
calculating the DTS.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1544
again.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3808>

Revert "avvidenc: Set timebase in the ffmpeg context to nanoseconds and set framerate"

This reverts commit 50db59449e9cdb637c5f0c1b7c5dccd582fac4ee.

This broke the MPEG-1 video encoder as it requires the framerate to be
used for the timebase.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1734

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3808>

vaapi: Skip plugin pc file for shared plugins

Following similar logic in the other subprojects.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3778>

udpsrc: GstSocketTimestampMessage only for SCM_TIMESTAMPNS

Deserialize socket control messages as GstSocketTimestampMessage only
if (level, type) is (SOL_SOCKET, SCM_TIMESTAMPNS).

Without this patch, messages with types SCM_RIGHTS or SCM_CREDENTIALS
could be deserialized as GstSocketTimestampMessage instead of
GUnixFDMessage or GUnixCredentialsMessage from gio.

Fixes #1736

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3777>

ci: update for 1.22 branch

Don't have validate do --check-bugs in the 1.22 branch, as
any issues fixed may only have been fixed in the main branch.

Back to development

Release 1.22.0

gst-plugins-ugly: update translations

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3773>

gst-plugins-bad: update translations

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3773>

gst-plugins-good: update translations

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3773>

gst-plugins-base: update translations

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3773>

gstreamer: update translations

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3773>

d3d11compositor: Workaround blending artifacts on Intel platform

Disable vendor implemented converter in d3d11compositor to workaround
artifacts

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3770>

glvideomixer: Fixing error with mouse navigation events when no pixel-aspect-ratio

Handling mouse navigation events in glvideomixer element, if no
pixel-aspect-ratio info in the caps, an assertion error is produced
inside gst_util_fraction_multiply because default denominator is zero.

Error fixed:
```
(gst-launch-1.0:102654): GStreamer-CRITICAL **: 00:47:51.598: gst_util_fraction_multiply: assertion 'b_d != 0' failed
```

Simple pipeline to reproduce the issue:
```
gst-launch-1.0 -v glvideomixer name=mix ! glimagesinkelement  gltestsrc  ! mix.sink_0
```

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3766>

playback: New playback elements are no longer experimental

The API is not expected to change and it has been used extensively since 1.18.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3761>

examples: webrtc: rust: Update dependencies

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3758>

examples: webrtc: multiparty-sendrecv: rust: Remove unnecessary macro recursion limit annotation

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3758>

examples: webrtc: sendrecv: rust: Implement `OFFER_REQUEST` handling

Allow requesting an offer from the peer if we're joining a call with a
peer, and allow the peer to request an offer from us if waiting for an
incoming call.

This implements all 4 variants the protocol allows for.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3758>

examples: webrtc: sendrecv: rust: Allow providing our ID via the commandline

Otherwise it continues to use a random ID as before.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3758>

examples: webrtc: sendrecv: rust: Implement TWCC support in both directions

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3758>

examples: webrtc: rust: Set keyframe-max-dist=2000 and picture-id-mode=15-bit for VP8 and perfect-timestamps=true for audio

This makes it in sync with the C sendrecv and generally behaves better.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3758>