webrtc/nice: Enable nice debug log

It has been omitted since it was updated to gst 1.22 version.

Change-Id: I721c195ce30116dade93ef5b23ad9f060c57c026

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 #1544(https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1544)
again.

Part-of: <!3808>

Change-Id: I7bf166986e5fc9b9a43a62eb32c496d4459cd445

Merge branch 'tizen_gst_1.22.0' into tizen

Change-Id: I1e1e80c68f113d92fc502d6e23ba151ce941fdef

videoconvertscale: Support I420-to-SN12 converting

- Additional change
 : Enable videoconvertscale.
 : Remove unused directory(videoconvert, videoscale) which are already removed from upstream.

[Version] 1.22.0-8
[Issue Type] Update

Change-Id: I6d8635c0717b81d70a6eaa39cedb827111cf0f5b
Signed-off-by: Jeongmo Yang <jm80.yang@samsung.com>

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

Change-Id: Ib28d0eb58fb6f280542af1e2fb9cc43142120336
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3786>

good: enable adaptivedemux2

- good: enable adaptivedemux2 to support hls playback
  with uridecodebin3 which does not use old hlsdemux
- good/adaptivedemux2: make post variant info at new hls plugin
- base/uridecodebin3: make post buffering msg without drop
  to control buffering at libmm-player

Change-Id: I94c310ceefe57efeba439112d1d0fe4f3e38553f

gst-plugins-bad: Fix pkgconfig file issue

pc files from this project could not be found when compiling other
packages using these pc files as BuildRequires.
As a result, some Tizen native packages could not be built well.
e.g.) capi-media-webrtc, capi-media-streamer

This issue is related to the meson version. See below.
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3061

It is fixed as per the current circumstance of Tizen dev.

Change-Id: Iad46fa21d87416b4b3bec0d90f538dfff74a83d1

fixup! Apply GStreamer 1.22.0 into Tizen

A crash issue related to webrtcbin is fixed.

Change-Id: Ie91cad999bb7ec1168c98cb7f5179d81e23c60ec

nice: Sync with libnice 0.1.21's

Change-Id: I9eb5533bff65b614045d08bd8f4d5bf6551f536e

avenc_mpeg2video:Apply the patch for breaking some encoders

- https://github.com/GStreamer/gstreamer/commit/5372b77987cbdf16f681526152325bee8388ab62
Revert "avvidenc: Set timebase in the ffmpeg context to nanoseconds a…
…nd set framerate"

This reverts commit 50db594.

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/3780>

Change-Id: I40383d874e929d1eade6f586e5a977631d4d556e

v4l2videodec: Add removed tizen feature code by conflict

- It fixes failure of player API.

Change-Id: Id9e0820daf65fa398755c24b39945e1a8e399142
Signed-off-by: Jeongmo Yang <jm80.yang@samsung.com>

webrtc: Use nice implementation for WebRTCICE

Change-Id: I414d9dfca5e679efc2d93ca7abd1b82a91f6f3aa

Apply GStreamer 1.22.0 into Tizen

- gbs build works : both full and each plugin

Change-Id: I7d4948119a07a67efd66b129e9dd65d70c4230ce

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>

examples: webrtc: sendrecv: rust: Use the correct payload types if the remote is the offerer

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

wpe: Add 'run-javascript' action signal

Introduce way of running a script in the context of the internal
webView.

Fixes #1722

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3753>

decodebin3: Remove failing stream from selection

If no decoder is present for a given stream, remove it from the requested
selection.

Fixes #1581 and #1662

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3754>

decodebin3: Don't intercept queries if no parsebin present

If we don't use a parsebin, we forward the queries as-is from upstream. There is
no reconfiguration possible within identity.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3754>

decodebin3: Don't insert parsebin if input is already parsed

This is a temporary workaround until we find a generic solution to indicate that
a stream has already been "parsed".

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3754>

win32ipcvideosrc: Fix leak in fallback copy path

Clear shared memory after use

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3751>

va: Fix path leak

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3755>

gst: Mark caps parameters in ALLOCATION query API as nullable

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3752>

gst-play: add --no-position

It can be useful to reduce or remove the clutter in logs due to those
queries when debugging.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3698>

matroska: Add `stream-format = (string) obu-stream` to AV1 caps

Anything else is not allowed in Matroska/WebM.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3740>

isomp4: Add `stream-format = (string) obu-stream` to AV1 caps

Anything else is not allowed in MP4.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3740>

gst-examples: Update Rust dependencies

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3750>

gleglimage: Fix GL_OES_EGL_image_external name in debug trace

There is a debug log when this extension is missing, but it was
reporting the wrong extension name, which was confusing.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3748>

input-selector: Take the object lock while iterating sinkpads

Otherwise we can race with pad removal and crash from use-after-free.

Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1717
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3745>

glvideomixer: don't clear pad->geometry_change when setting an identical value

Closes #1715

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3733>

gst-docs: include dmabuf and gapless playback design docs

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3744>

uridecodebin3: clear subitem if suburi is set to NULL

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1649

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3730>

qtdemux: Add basic support for AVC-Intra video

AVC-Intra is a range of H.264-compliant intra-only codecs from
Panasonic. The codes and descriptions have been taken from VLC.

The (encumbered) sample I have here produces byte-stream H.264,
including SPS and PPS and no `avcC` box.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3739>

gl: Fix crash in cocoa due to race in gst_gl_context_activate()

We create a new context in `gst_gl_context_create_thread()` and then
activate it on the current thread. Thereafter we assume that the
current thread continues to be the active thread for that context and
call `gst_gl_context_fill_info()` which asserts that the current
thread is the active thread.

However, if at the same time a different thread calls
`send_message_async()`, it will call into
`gst_gl_window_cocoa_send_message_async()` which will schedule the
message to be invoked using GCD. That anonymous function will also
call `gst_gl_context_activate()`, which creates a race, which can lead
to:

```
gst_gl_context_fill_info: assertion 'context->priv->active_thread == g_thread_self ()' failed
```

Fix it by using `gst_gl_context_thread_add()` to invoke `fill_info()`
on the context.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3732>

ges: smart-mixer: Avoid video freeze on NLE stack changes

Since we support "nested compositors"[0] we were not finalizing the pads_infos
and thus not releasing the mixer pad on stack change, meaning that
the following stacks were never prerolling.

[0] https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2094

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3738>

ges: Fix issues avoiding to use `operator` when unavailable on mixer

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3738>

gl/cocoa: Store a weak reference to the `GstGLWindow` instead of the `GstGLContext`

We can't rely on the `GstGLContext` to stay alive and need to keep track
of it. For that we keep track of the `GstGLWindow` in a weak reference
to avoid a reference cycle, and get the corresponding `GstGLContext`
whenever needed.

With contributions from Nirbheek Chauhan.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1697

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3729>

Release 1.21.90

avvidenc: Offset PTS to zero to fix bitrate control

Otherwise ffmpeg's rate control algorithm will not work correctly as
it is based on the absolute PTS values.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-libav/-/issues/91

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3718>

urisourcebin: Cleanup on failure

This ensures there is a source present/valid *ONLY* in PAUSED and above.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3720>

urisourcebin: Unset locked state

Some elements still had it in failure cases

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3720>

uridecodebin3: Handle state synchronization failures

Properly reset and remove the failing urisourcebin if we failed bringing them up
to the same state as ourselves.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1708

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3720>

rtopuspay: Use GstStaticCaps to cache parsed caps

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3674>

rtopuspay: Ignore the stereo parameter in multiopus caps

Also add unit tests for the various variants

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3674>

rtpopuspay: Leave original caps as-is

This should make it work if someone specifies stereo with MULTIOPUS
somehow.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3674>

rtpopuspay: Return upstream channel filter based on OPUS vs MULTICAPS

Only allow 1 or 2 channels if the caps are OPUS, or 3+ if they are
MULTIOPUS.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3674>

rtpopus: Put MULTIOPUS in all caps

The RTP payload encoding-name are always in caps in GStreamer.
In SDP, they are not case-sensitive, but since caps are, we need to pick
a caps, and we picked upper-case along time ago.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3674>

android: Update minimum SDK version to Android 21

Otherwise we can't bump the minimum version of the cerbero build without
it breaking linking of the applications.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3717>

d3d11memory: Fix potential crash in GstD3D11PoolAllocator

Allocator object can be the final refcount holder, when a GstD3D11Memory
is being released. Do unref allocator once there's no more
remaining operation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3715>

d3d11memory: Remove GstD3D11PoolAllocator::dispose

Clear GstD3D11Device object in finalize method as well

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3715>

glbasefilter: Copy metas when they only contain they "video" tag

Same logic as in `GstVideoFilter`

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3716>

decodebin3: Fix global group-id handling

The goal of the "global" group-id is to fix new inputs that do not come from the
same "source" as others. In order to ensure all "current" streams have the same
group-id we distribute the first valid group-id to all streams.

This commit fixes two issues with that:

* When inputs are unlinked they weren't always properly resetted (it would only
  work if parsebin is used, which is no longer the default in
  uridecodebin3/playbin3).

* When computing the global group-id, take into account unset
  group-id (i.e. GST_GROUP_ID_INVALID).

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1698

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3712>

gst-plugins-ugly: update translations

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1575

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3711>

gst-plugins-bad: update translations

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3711>

gst-plugins-good: update translations

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3711>

gst-plugins-base: update translations

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3711>

gstreamer: update translations

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3711>

Fix translation pot files when creating dist tarballs

Add version as per Translation Project requirements and
also add a .pot file without the ABI suffix.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3711>

decodebin3: ensure recalculate_group_id() is called with INPUT lock

This function uses main_input and current_group_id which are protected
by the input lock.
Some calls were already safe but not all.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3702>

uridecodebin3: fix stream leak

GstSourcePad owns the GstStream but was not unreffing it when being
disposed.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3617>

decodebin3: fix dead lock when removing pad

gst_element_remove_pad() is triggering a call to
gst_decodebin3_input_pad_unlink() which needs the input lock as well,
resulting in a dead lock.

Fix #1667

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3614>

tools: Use `gst_macos_main()` on macOS

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1673

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3709>

docs: Need to use versioned paths on macOS

Especially with pkg-config, it needs to be called with the right path
otherwise it might not pick up the prefix correctly.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1677

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3706>

meson: gst-plugins-bad: Disable spammy MSVC warning

warning C5051: attribute 'noinline' requires at least '/std:c++20';
ignored

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3705>

tests: interactive: test-effect-switch: use autovideosink

Also works on Windows.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3703>

mxfmux: Add all local tags to the primer

No matter if they're allocated via GSlice or malloc(). The allocator is
completely irrelevant, all local tags need to be in the primer so they
can be handled.

This didn't have any effect in practice because all local tags that
appear in the muxer are allocated via GSlice. Only from the demuxer they
might be allocated via malloc().

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3699>

jpegdec: Disable libjpeg-turbo SIMD acceleration support for now

The libjpeg-turbo SIMD acceleration support suffers from multiple
unresolved cornercases. Disable the libjpeg-turbo for now until
those cornercases are resolved.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3694>

ci/README: basic instructions for running locally

These should be extended upon by someone more knowledgeable with the
topic, but that's the basic information I would want to find there
in order to start debugging CI issues locally.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3681>

libav: avaudec: display flow return name instead of value

More convenient to read in logs. gstavviddec already does it.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3700>

gst-play: fix command line option string formatting

Add missing space to translated string.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3696>

av1enc: Add property for controlling max distance between 2 keyframes

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2743>

aom: av1enc: Expose more properties

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2743>

aom: av1enc: Remove redundant enum variants from header

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2743>

aom: av1enc: Ensure that input pts is strictly increasing

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2743>

aom: av1enc: Fix pts unit

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2743>

dfbvideosink: Fix compile error

Fix some compile errors

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3670>

webrtc nice: Let libnice handle STUN/TURN DNS resolution

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3676>

webrtc: Change requirement to libnice 0.1.20

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3676>

rtspclientsink: Add `publish-clock-mode` property

This allows modifying the behaviour how/if the pipeline clock is
published according to RFC7273, similar to the same API on
`GstRTSPMedia`.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3581>

rtspclientsink: Fix docs for various properties

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3581>

docs: explicitly declare gir build dependencies

As the path to the gir file is passed to hotdoc.generate_doc() and
not the build target itself, meson doesn't know about the dependency.

In turn, as the CI doesn't build everything before building the
documentation target, some gir files might not exist, for instance
in the case of gst-rtsp-server, causing the output documentation to
be empty.

The error occurred silently because hotdoc accepts wildcards for
*-sources arguments, thus it won't warn about a missing gir file as
it is legitimate for glob matching to resolve to nothing.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3686>

typefindfunctions: Add missing length check to XML typefinder

Fixes https://bugs.chromium.org/p/oss-fuzz/issues/detail?id=54811

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3690>

typefindfunctions: Reduce XML typefinder minimum data size from 64 to 32 bytes

64 bytes makes the typefinder fail on very small XML documents on which
it succeeded previously.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3690>

opusdec: Try harder to negotiate the upstream channels/rate preferences

It might be possible to fulfill those but not with the first caps
structure. Instead of just fixating the first caps structure, check if
the preference can be fulfilled by any of the structures as the first
step.

Without this the following pipeline negotiates to mono after the
decoder because opusenc only has a single channel in its first caps
structure.

    gst-launch-1.0 audiotestsrc ! audio/x-raw,channels=2 ! opusenc \
        ! queue ! opusdec ! queue ! opusenc ! fakesink

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3689>

Revert "opusenc: Reverse channel order in template caps"

This reverts commit cdf411e82d7e437e5de138492b6e75e5513686ee. It causes
pipelines to unnecessarily negotiate to 8 channels.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1688

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3689>

dvbbasebin: don't rely on g_key_file_get_(integer|uint64) return

instead check if an error was returned, and fail if any

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3688>

hlsdemux: Consider starting stream time in presentation offset

When calculating the presentation offset for CMAF input in live
playback, subtract the stream_time of the fragment from the
calculated presentation offset, so that the first fragment
is played at running time zero.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3680>