vaav1dec: Use gst_va_base_dec_prepare_output_frame().

And simplify a bit the code flow.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3480>

va: Add and use gst_va_base_dec_prepare_output_frame().

This helper will do downstream negotiation and later will
allocate the output frame.

H265 and AV1 decoders don't use this approach since their output
frame allocation is different.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3480>

va: Remove last_ret error handling in decoders.

It was used in the early development of the base classes. Now it
shouldn't be needed.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3480>

va: Add and use gst_va_base_dec_process_output().

This function will copy the frame, if it's needed, and will apply buffer flags.

The function is used by all the decoders.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3480>

va: Fix the caps memory leak by gst_va_pool_new_with_config().

The gst_va_pool_new_with_config() will ref the caps, and so we need
to unref after that.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3503>

va: Add render node name in non-first devices.

There could be multi-GPU setups where the non-first has more
entrypoints than the first one, and the elements names are not
homogeneous, leading to pipeline building error.

This patch add the render node in the elements names when they belong
to the non-first device.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3491>

vafilter: Increase the caps for HDR.

As they might be other medatadata types.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3473>

vafilter: Increase the number of 3DLUT caps to 16.

To fix the warning on Alderlake

vafilter gstvafilter.c:534:gst_va_filter_ensure_filters:<vafilter0>
vaQueryVideoProcFiltersCaps: list argument exceeds maximum number

Increase the number of caps to 16 as vadumpcaps does.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3473>

qt: deactivate context if fill_info fails

Now the OpenGL context is deactivated if call to gst_gl_context_fill_info()
fails in gst_qt_get_gl_wrapcontext(), preventing that the context is left
activated, which could lead to invalid memory reads.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3492>

av{aud,vid}dec: Report decoding errors to the base class

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1889>

v4l2: Fix SIGSEGV on 'change state' during 'format change'

Ensure all access to v4l2object->pool imply taking a lock and a hard ref on the pool

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3481>

gst-inspect: Don't leak list

Just iterate the list instead of trying to be smart...

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3497>

example: vaenc-dynamic-reconfigure: Support H265.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2036>

va: Add H265 SCC profile support.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2036>

va: Add extended formats support such as 10/12 bits, 4:2:2 and 4:4:4.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2036>

va: enable multi tile support for H265 encoder.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2036>

va: enable vah265lpenc for low power mode H265 encoder.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2036>

va: Add the low-delay-b frame support for H265.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2036>

va: Add prediction direction attribute support for H265 encoder.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2036>

va: Register and enable and the vah265enc plugin.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2036>

va: Implement the vah265enc plugin for va HEVC encoding.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2036>

opusdec: Use proper guint/guint8 type conversion

Do not cast, that might yield wrong results.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3494>

opusdec: Negotiate default to 2 channels

In that place, dec->n_channels can still theoretically be 0. Default to
2 in that case.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3494>

opusenc: Reverse channel order in template caps

We want to make it so that we prefer a higher, not lower, number of
channels. Otherwise, this pipeline would convert from 2 to 1 channels:

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/3494>

avviddec: Report flow error when decoder refused packet

In cases where an invalid input packet is submitted to the decoder we emit a
warning but reporting the flow error upstream would also be useful. This came up
with a case were the application interacts directly with the decoder, using a
mechanism similar to GstHarness.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3463>

audiodecoder: Make data processing errors non-fatal by default

The previous default value of `max-errors` was too small and would potentially trigger the
decoder to emit errors too often for most cases.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3478>

videodecoder: Make data processing errors non-fatal by default

The previous default value of `max-errors` was too small and would potentially trigger the
decoder to emit errors too often for most cases.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3478>

sdpmessage: Annotate SDP message parameters for `parse_uri()` and `parse_buffer()` correctly

They are not actually `out` parameters but must be allocated and
initialized to a valid SDP message by the caller. Just allocating them
without initialization will cause memory corruptions because the
functions will try to clear/append to the existing SDP message.

As such they should be just marked as `transfer none`.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1605

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3484>

mfvideosrc: Fix buffer leak

The allocated buffer should be released

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3479>

mediafoundation: Chain up from GObject::constructed

... so that GstTracer can trace it

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3479>

d3d11: Use correct ref/unref methods

Those objects are GstObject subclasses

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3479>

sdp: use g_strdup instead of strdup

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3475>

glimagesink: Fix render rect assertion

Whenever the surface is resized before the stream is negotiated, we endup
with an assertion in libgstvideo.

  gst_video_center_rect: assertion 'src->h != 0' failed

This fixes it, by following the style aready in place, which is to ensure
surfaces have a minimum size of 1x1.

Fixes #1139

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3467>

audioconvert, audioresample, audiofilter: fix divide by 0 for input buffer without caps

gst-launch-1.0 audiotestsrc ! udpsink host=127.0.0.1
gst-launch-1.0 udpsrc ! audioconvert ! autoaudiosink

would crash with a floating point exception when clipping the input
buffer owing to a division by zero because no caps event was received.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3469>

win32ipcvideosrc: Fix property name and enum value mismatch

Make them consistent

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3465>

win32ipcvideosrc: Protect pipe from flush thread

The pipe object must be protected against GstBaseSrc::unlock()
thread.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3465>

h264parser: Fix a typo in pred_weight_table parsing.

When setting default values, the reference list number of l1 is wrong.

Fix: https://gitlab.freedesktop.org/gstreamer/gstreamer-vaapi/-/issues/336
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3464>

msdkdec: Update decoding pts

Update frame pts before calling gst_video_decoder_finish_frame, or frames may be lost when transcoding in specific stream

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3449>

Install meson from git when MESON_COMMIT variable is set

When starting manually a CI pipeline with MESON_COMMIT variable defined,
install the specified version of Meson. This can be used to help the
Meson project smoke testing RC releases or investigate eventual
regressions.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3036>

win32ipc: Add WIN32 shared memory videosrc/sink elements

Windows supports various IPC methods but that's completely
different form that of *nix from implementation point of view.
So, instead of adding shared memory functionality to existing
shm plugin, new WIN32 shared memory source/sink elements
are implemented in this commit.

Each videosink (server) and videosrc (client) pair will communicate
using WIN32 named pipe and thus user should configure unique/proper
pipe name to them (e.g., \\.\pipe\MyPipeName).
Once connection is established, videosink will create named shared memory
object per frame and client will be able to consume the object
(i.e., memory mapped file handle) without additional copy operation.

Note that implementations under "protocol" directory are almost
pure C/C++ with WIN32 APIs except for a few defines and debug functions.
So, applications can take only the protocol part so that the application
can send/receive shared-memory object from/to the other end
even if it's not an actual GStreamer element.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3441>

rtpsession: Support disabling late adjustment of ntp-64 header ext

Currently in rtp_session_send_rtp(), the existing ntp-64 RTP header
extension timestamp is updated with the actual NTP time before sending
the packet. However, there are some circumstances where we would like
to preserve the original timestamp obtained from reference timestamp
buffer metadata.

This commit provides the ability to configure whether or not to update
the ntp-64 header extension timestamp with the actual NTP time via the
update-ntp64-header-ext boolean property. The property is also exposed
via rtpbin. Default property value of TRUE will preserve existing
behavior (update ntp-64 header ext with actual NTP time).

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1580

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3451>

add new plugin for Qt 6 rendering inside a QML scene

- Based heavily on the existing Qt5 integration however:
  - The sharing of OpenGL resources is slightly different
  - The integration with the scengraph is a bit different
- Wayland, XCB and KMS have been smoke tested.  Android, MacOS/iOS,
  Windows may or may not work.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3281>

mfvideosrc: check HRESULT of CreateClassEnumerator correctly

Fixes a crash during direct show device enumeration

Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1599
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3460>

v4l2: bypass check some transfer types in colorimetry

v4l2 will report fail for some streams whose colorimetry value such as 2:4:8:3.
Can bypass check these transfer types in colorimetry to avoid error.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3440>

docs/design: Add document detailing the new gapless/instant-uri changes

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3457>

uridecodebin3: propagate about-to-finish for new inputs

If the pending next urisourcebin is already draining, make sure we propagate it
when activating it.

Fixes gapless playback of small entries

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1588

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3457>

uridecodebin3: Remove old fixme

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3457>

(uri)decodebin3: Ensure group-id consistency

Make sure that group-id of a given play item are made consistent from the
start (sources) and all the way through the output.

This ensures that we can reliably detect that we have switched to the next play
item on the output of decodebin3 (and we can therefore properly free/release it)

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3457>

uridecodebin3: Fix shutdown procedures in probe

When shutting down, we want to remove the urisourcebin blocking probes ... but
we also want to propagate a GST_FLOW_FLUSHING upstream (and not
GST_FLOW_NOT_LINKED) to make the upstream task gracefully stop instead of
posting an error message.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3457>

decodebin3: Properly reset when going back to READY

Clear the remaining stream-related fields when going from PAUSED to READY, and
use when disposing.

Fixes various issues when re-using decodebin3/playbin3

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3457>

decodebin3: Don't output bogus GST_MESSAGE_STREAMS_SELECTED

When `is_selection_done` is called, it checks that all the requested streams are
present in the active stream list ...

... except there could very well be a (about to be removed) stream from the
previous selection present.

Therefore filter the list of streams we add to the message by the streams which
are actually requested.

Fixes issues when switching between different stream types (ex: video-only to
audio-only).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3457>

Use the correct SSRC(s) when routing a RTCB FB FIR

Previously we tried to route an incoming RTCP FB FIR to the correct ssrc
using the "media source" component of the RTCP FB message. However,
according to RFC5104 (section 4.3.1.2) the "media source" SHALL be set
to 0. Instead the ssrc(s) in use are propagated via the FCI data. Now
a specific GstForceKeyUnit event is sent for every ssrc.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3292>

rtpsource: Track the seqnum for senders

RTP source statistics are tracked for local senders by
treating them as a receiver of their own outbound packets.

Accordingly, track the highest packet seqnum so that the
packets-lost calculation generates a sensible number instead
of always reporting -$number_of_packets_sent

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3454>

adaptivedemux2: Add GStreamer to the deps list

Explicitly dep on GStreamer so as not to accidentally
link to the system version in a git build

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3453>

kmssink: Fix compilation without kernel headers

There was a drm/drm_mode.h included added recently, drm/ is usually
referencing the linux kernel header, but we only requires the libdrm
headers to be installed. On top of this, including drm_mode.h is never
needed as its already included by drm.h.

Fixes #1596

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3452>

onnx: Update to OnnxRT >= 1.13.1 API

- Replace deprecated methods
- Add a check on ORT version we are compatible with.
- Add clarification to the example given.
- Add the url to retrieve the model mentioned in the example.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3388>

h265parse: Add the missing timestamp when splitting a frame.

When splitting a frame, the gst_buffer_copy_region() does not copy
the timestamp correctly for sub frames when the offset is not 0.
We still need those timestamps for each output sub frame.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3448>

h264parse: Add the missing timestamp when splitting a frame.

When splitting a frame, the gst_buffer_copy_region() does not copy
the timestamp correctly for sub frames when the offset is not 0.
We still need those timestamps for each output sub frame.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3448>

dvbsubenc: Write Display Definition Segment if a non-default width/height is used

Otherwise it can't be rendered by dvbsuboverlay or ffmpeg at least.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3446>

inputselector: Add drop-backwards property

When sync-streams=true, drop backwards buffers on pad switch.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3256>

inputselector: Fix waiting on sync-mode=clock

Basically copy over what clocksync does, but taking into account that we
have multiple upstream latencies.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3256>

av1dec: Demote rank to secondary

cerbero does not build this plugin for now, and there's altanative
dav1ddec

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3287>

rtspsrc: Don't replace 404 errors with "no auth protocol found"

When getting a "404 Not Found" response from the DESCRIBE request, the
source produced a "No supported authentication protocol was found" error
instead of passing on the 404, which was confusing.

Only produce this error message when we're handling a response of "401
Unauthorized" without a compatible WWW-Authenticate header.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3414>

vaapipostproc: Fix the negotiation failure of some formats

This patch fixes issue https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1565

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3399>

kmssink: add HDR10 infoframe support

If stream has HDR10 metadata and HDMI device EDID supports it, this patch
will set the DRM properties accordingly.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3303>

kmssink: Add skip-vsync property

The legacy emulation in DRM/KMS drivers badly interact with GStreamer and
may cause the framerate to be halved. With this property, users can disable
vsync (which is handled internally by the emulation) in order to regain the
full framerate.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3303>

adaptivedemux2: Don't leak caps in debug statements

Instead just directly use the stream object (which will report the caps)

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3443>

adaptivedemux2: Don't leak tags

If we got them from GstStream, we should unref them when done

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3443>

adaptivedemux: Use gst_clear_tag_list_where applicable

Clearer and ensures fields are reset

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3443>

oggdemux: Don't leak pending seek event

Make sure any pending seek event is released when going back down to READY.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3443>

rtspsrc: Don't leak sticky events

We have incremented the reference 2 lines above, and
gst_pad_store_sticky_event() does not take a reference, therefore release it

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3443>

parsebin: Don't leak parsepad list on shutdown

Free it as it is down in other cases

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3443>

textrender: Don't pass plaintext as pango markup to Pango

Otherwise e.g. & in the text will cause Pango to complain about invalid
markup and render the text incorrectly.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3445>

textrender: Don't blindly forward all events

Use gst_pad_event_default(), which does the right thing by default.
Especially it does not forward text/x-plain caps downstream.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3445>

event: Add transfer none annotation to gst_event_new_stream_collection()

Update the documentation

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3387>

adaptivedemux2: Fix sticky event storage.

Use the new gst_event_type_to_sticky_ordering() method to retrieve
the order that sticky events should be sent / stored in, instead
of assuming it's the event type value.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3387>

pad: Fix sticky event ordering for instant-rate-change

The event type for instant-rate-change events was poorly chosen,
leading to them being re-sent too late and even after EOS.

Add a mechanism in GstPad for the sticky event order to be
different to the value of the event type to fix that up.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3387>

av1decoder: Clear highest_spatial_layer per sequence header

Clear the value to default zero, indicating that no spatial scalability
layer is used.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3430>

av1decoder: Don't error out by dropped OBU

OBU can be dropped if the current layer is not in selected operation
point

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3430>

av1decoder: Fix wrong spatial layer validation

Highest spatial id and temporal id is independent, and should not drop
temporal enhance layer by the previous condition. Note that
the decision for dropping OBU based on operation point is being
handled in gst_av1_parser_identify_one_obu() already.

Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1585
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3430>

av1parser: Don't print warning for expected OBU drop

Dropping an OBU which is not in selected operation point is an
expected condition.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3430>

av1parser: Remove impossible condition

unsigned integer cannot be negative

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3430>

fakesrc: avoid time overflow with datarate

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3433>

webrtcbin: Remove queue after rtpfunnel

The original BUNDLE support commit placed a queue after the
rtpfunnel that combines streams, but I don't see a good reason for
it. It has default settings, so if network output is slow might
accidentally store up to 1 second of pending data, increasing
latency.

Remove it in favour of doing any necessary buffering before
webrtcbin. If it turns out that there is a reason for it to
exist, the limits should probably be configurable and small.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3437>

examples/sendrecv: Remove extra unref of webrtcbin

The code now constructs webrtcbin with a floating ref and then
gives it to the pipeline. The extra unref is one too many.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3436>

rtpbasedepayload: Drop redundant reference timestamp buffer meta in RTP depayloaders

Currently, when rtspsrc property add-reference-timestamp-metadata=true,
a downstream rtph264depay element will attach multiple copies of the
same GstReferenceTimestampMeta to the depayloaded media buffers. This
can have signficant performance impacts further downstream in a pipeline
like the following:

    rtspsrc add-reference-timestamp-metadata=true ! rtph264depay ! h264parse ! ... ! rtph264pay ! ...

For example, if there are 10 packet buffers for a frame of RTP H.264
video, each of those packet buffers will contain the same reference
timestamp meta. The rtph264depay element will then attach all 10
metadata to the depayloaded frame. And then later when we payload the
frame buffer again for proxying, we now have 10 more buffers each with
10 instance of the same metadata. Allocating/deallocating 100+ instances
of metadata @ 30fps for multiple streams has a pretty large performance
impact.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1578

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3431>

webrtc: Improve GstWebRTCStatsType docstring

Fix a typo of peer-connectiion -> peer-connection

Add a link to the w3c RTCStats type for a description
of what each statistics type is.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3434>

webrtcbin: Fix a typo in debug log

transceiever -> transceiver

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3434>

examples/webrtc: Configure payload types

MR 2398 broke the webrtc sendrecv example
by not configuring the payload types, so both audio and video streams
get sent on payload 96.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3434>

qsv: Promote encoder rank to PRIMARY on Windows

QSV is very well integrated with GstD3D11 infrastructure on Windows,
and this is the recommended H/W encoder element over the MediaFoundation
plugins on Intel GPU system.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3393>

video-frame: Avoid using tile width

The tile width in pixel is not always available. Notably for
8L128 10bit format, the tile is 8x128 bytes, and the pixel
format is fully packed. That means that the tile contains at
least 6 pixels per line, but it also hold some bits of the
pixel from the same line on the previous or next tile.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3424>

video: Add arbitrary tile dimensions support

In current tile representation, only tiles with power of two
width and height in bytes are supported. This limitation
prevents adding more complex tiles formats.

In this patch, we deprecate tile_ws and tile_hs from GstVideoFormatInfo and
replace if with an array of GstVideoTileInfo. Each plane tiles are then
described with their pixels width/height, line stride and total size.
The helper gst_video_format_info_get_tile_sizes() that depends on the
deprecated API is also being removed. This can simply be removed as it wasn't
in any stable release yet.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3424>

gst: serialization of GLibDateTime

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2980>

aggregator: Implement force_live API

Setting force_live lets aggregator behave as if it had at least one of
its sinks connected to a live source, which should let us get rid of the
fake live test source hack that is probably present in dozens of
applications by now.

+ Expose API for subclasses to set and get force_live
+ Expose force-live properties in GstVideoAggregator and GstAudioAggregator
+ Adds a simple test

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3008>

splitmuxsink: Avoid assertion when WAITING_GOP_COLLECT on reference context

I have seen a backtrace out in the wild where this happened. Maybe after
receiving EOS and stream-start on the reference context.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3005>

webrtcbin: Add settings for HTTP proxy

Pass this to libnice which has a simple HTTP 1.0 proxy with basic
authentication only.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2867>

tsdemux: Add pad-name to warning for continuity mismatch

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3141>

decodebin2: Minor debug fix for decodepad

decodedad might have their name changed when exposing, causing a race when
trying to get their name without taking a lock. Just use GST_PTR_POINTER in
debug statements instead

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3428>

imagesequencesrc: Don't leak caps

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3428>

ges-launcher: Don't leak string

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3428>