id3tag: Map GST_TAG_MUSICBRAINZ_RELEASETRACKID

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1331>

id3tag: Map GST_TAG_MUSICBRAINZ_RELEASEGROUPID

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1331>

id3tag: Remove trailing whitespace

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1331>

qtdemux: Fix can not demux Opus track made by qtmux

Opus stream info is read from dOps box [1]. The offset of dOps box in Opus box is different in mp4a version 1 and 0 [2]. Calculate the offset of dOps box according to mp4a version.

[1] https://opus-codec.org/docs/opus_in_isobmff.html

[2] subprojects/gst-plugins-good/gst/isomp4/atoms.c:sample_entry_mp4a_copy_data:2146

Fixed: https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/918
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1329>

docs: app-dev: events: seeking: use CLOCK_TIME_NONE instead of -1 and fix parameter names

to match the parameter names in the gst_element_seek() declaration.

Closes https://gitlab.freedesktop.org/gstreamer/gst-docs/-/merge_requests/34/

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1324>

docs: installing-on-macos: flesh out instructions what to download

Closes https://gitlab.freedesktop.org/gstreamer/gst-docs/-/merge_requests/106/

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1324>

plugin-development: basics-boilerplate: Remove unneeded meson.build edit step

Closes https://gitlab.freedesktop.org/gstreamer/gst-docs/-/merge_requests/157/

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1324>

docs: installing-on-windows: mention packages to download

Closes https://gitlab.freedesktop.org/gstreamer/gst-docs/-/merge_requests/158/

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1324>

docs: tutorials: add link to Rust version of the tutorials

Closes https://gitlab.freedesktop.org/gstreamer/gst-docs/-/merge_requests/159/

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1324>

docs: tutorials: playback-3: flesh out build instructions

Add information about how to compile this file. The code in this
tutorial also requires `gstreamer-audio-1.0`, so we cannot use
the commands from the previous tutorial.

Closes https://gitlab.freedesktop.org/gstreamer/gst-docs/-/merge_requests/164/

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1324>

docs: plugin-dev: basics-boilerplate: fix typo in variable name

Closes https://gitlab.freedesktop.org/gstreamer/gst-docs/-/merge_requests/166/

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1324>

avtpsrc: Use correct size for provided buffers

Without this patch the following pipeline would send packets containing
garbage in the data section.
$ gst-launch-1.0 avtpsrc ! avtpsink

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1077>

appsrc: log when segment changes

We were logging when it does not change but not when it does, which is
confusing when reading logs.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1327>

avtp: crfsync: Warn when CRF package not yet received

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1075>

v4l2bufferpool: set video alignment of video meta

need apply video alignment info on video meta, downstream
element can get buffer alignment from video meta

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1326>

test: avtp: crf: Check for rounding errors

on average period calculation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1073>

avtp: crf: Use double for average period calculation

to also support CRF intervals like every 1,333,333ns 64 events

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1073>

tests: avtp: crf: Test for timestamp_interval > 1

in case of CRF AVTPDUs with single CRF timestamp.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1076>

avtp: crf: Properly handling one timestamp per PDU

The average_period should always represent the time between two
events. The specification defines the event time as the time
between audio samples, video frame sync, video line sync, etc.
In case of one timestamp per PDU the timestamp_interval identifies
the amount of events between the timestamp of one PDU and the
timestamp of the next PDU.
As described in IEEE 1722-2016 chapter
"10.4.12 timestamp_interval field" timestamp_interval shall be
nonzero.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1076>

shm: NetBSD build fix

shm_unlink() and friends live in librt on NetBSD. Adapt build system.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1323>

bin: Don't check twice for adding a bin to itself or removing it from itself

This is already covered by a `g_return_val_if_fail()` in the calling
function.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1320>

bin: Switch `g_warning()`s to `GST_WARNING_OBJECT()`s when adding/removing an element to a bin fails

The failure conditions can be overidden by subclasses, and a boolean
return value is provided to the caller whether adding/removing the child
element has actually worked. The caller can then handle this
accordingly but flooding stderr with this is not very useful.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1320>

vpxdec: Skip check of key frame when open GOP

Valid stream [1] might has no key frame.

Fixed: https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/890

[1] https://storage.googleapis.com/downloads.webmproject.org/test_data/libvpx/vp90-2-16-intra-only.webm

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1316>

gst-integration-tests: medias: update git module to latest commit

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1184>

osxaudio: Increase max channels to 64

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1059>

gstinfo: Fix leak in generate_backtrace_trace

Spotted by Laurent Pinchart.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/901>

aom: Set fixed_qp_offsets to a deactivated value

aom only uses fixed_qp_offsets with the
Constant Quality (Q) Rate Control mode,
previously this was locking any usage
with another Rate Control mode.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1198>

meson:ges: Fix typo adding ges gir to libs list

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1322>

gst-ptp-helper: Do not disable multicast loopback

Otherwise we cannot run gst-ptp-helper if the PTP master is on the
same device.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1306>

tests:ges: Take TIMEOUT_FACTOR into account

This env var is set in the CI so we grow the timeout as required
(when running in valgrind for example).

See https://gitlab.freedesktop.org/gstreamer/gstreamer/-/jobs/15449209

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1315>

magicleap: update lumin_rt libraries names to the latest official version

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1314>

codecs: h264dec: Calculate the latency by its bump mode.

The current latency calculation just uses the num_reorder_frames,
which is not very precise. We should consider the bump mode of the
DPB, the faster it bumps, the lower latency we will have.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1046>

doc: Update nvdec documentation

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/870

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1309>

imagefreeze: Only set caps if they do not match current caps

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1311>

h264parse: Don't insert extra AUD if exists in bitstream already

AUD nalu in packetized format is completely valid and therefore we should not
assume that we should insert AUD for packetized -> bytestream
conversion.

Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/862
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1296>

tests: h264parse: Add test for AUD insertion

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1296>

v4l2: Support bools for setting extra-controls

They're just mapped to 0 / 1 and can already be set as integers, but
being able to set them as booleans seems useful.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1308>

wasapi2ringbuffer: Fix client object leak

Check whether ringbuffer is holding client object already since
open_device() may be called multiple times

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1307>

webrtcbin: Clear errors from finding codec preferences before the next iteration

The media is just skipped and the error is not propagated to the caller,
so keeping it around here would cause assertions a bit later when trying
to set a new error over the old one.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1291>

webrtcbin: Move addition of attributes to the caps after making sure they're not empty or any

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1291>

webrtcbin: Don't require fixed caps when querying caps for a transceiver pad to match it with a media

Upstream caps might for example be
  application/x-rtp,media=audio,encoding-name={OPUS, X-GST-OPUS-DRAFT-SPITTKA-00, multiopus}
and while that is not fixed caps it is enough to match it with a media.

Only caps structures that have the correct structure name and that have
the media and encoding-name field are preserved, but if both are present
then these caps are used as "codec preferences".

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1291>

typefind: fix reading file extension from URI

Currently reading extension relies on the fact that everything after the
last"." character is a file extension. Whereas that works fine for most
of the cases, it breaks when the URI contains a query part.

E.g.: `http://url.com/file.mp4?param=value` returns `mp4?param=value`
instead of `mp4`.

In this commit we use URI parser to read the path of the URI (in the example
above, that is `/file.mp4`) and read extension from that path.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1305>

Back to development

Release 1.19.3

Update ChangeLogs for 1.19.3

ci: Do not clone in the trigger job

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1298>

multifilesink: Make minimum distance between keyframes in next-file=key-frame mode configurable

Previously this was hardcoded to 10s, which is not necessarily the
desired behaviour.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1292>

qtmux: Fix deadlock in gst_qt_mux_prepare_moov_recovery

Regression from 5766731bd4200c3a374522a749386f740347661a

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1288>

egl: handle configless contexts

With EGL_KHR_no_config_context, EGL contexts may just not
have an EGLConfig to give you. Deal with it.

Fixes: #858
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1289>

examples: win32-videooverlay-playbin: Add a repeat option

... in order to test pipeline/element reuse cases.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1275>

codecs: mpeg2: Drain before a new_sequence get signalled

The decoder may need to re-allocate the output buffer, it is easier if all
pictured have been outputed.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1013>

v4l2codecs: gstv4l2codecsmpeg2dec: implement a render delay

The v4l2 backend support delayed output for performance reasons.
It is then possible to use render delays to queue multiple requests
simultaneously, thus increasing performance.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1013>

codecs: gstmpeg2decoder: add support for render delay

Some decoding APIs support delayed output for performance reasons.
One example would be to request decoding for multiple frames and
then query for the oldest frame in the output queue.
This also increases throughput for transcoding and improves seek
performance when supported by the underlying backend.

Introduce support in the mpeg2 base class, so that backends that
support render delays can actually implement it.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1013>

v4l2codecs: Implement a MPEG2 V4L2 decoder element

Implement a MPEG2 V4L2 decoder element based on the previous h264
implementation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1013>

v4l2codecs: mpeg2: update to the new uAPI

The mpeg2 stateless api has undergone changes as it is being
destage. Update the v4l2-controls header to match.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1013>

sys: va: GstVaMpeg2Dec: use slice sc_offset and size

Seeing as how GstMpeg2Slice will now record the start code offset
as well as its size with the above field taken into account, the
manual computation in this class is not needed.

Remove it.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1013>

codecs: GstMpeg2Slice: add field for sc_offset and size

Downstream might need the start code offset when decoding.

Previously this computation would be scattered in multiple sites. This
is error prone, so move it to the base class. Subclasses can access
slice->sc_offset directly without computing the address themselves
knowing that the size will also take the start code into account.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1013>

v4l2codecs: Handle allocator creation failure

If `VIDIOC_REQBUFS` doesn't return enough buffers the allocator creation
function can fail and return `NULL`. Handle this by generating an error
and returning instead of segfaulting.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1220>

v4l2codecs: Fix segfault when destroying non-detached allocator

The GstV4l2CodecAllocator dispose function clears `self->decoder` but
the finalize function then tries to use it if the allocator has no been
detached yet.

Fix by detaching in the dispose function before we clear
`self->decoder`.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1220>

re-add gst-plugins-rs.wrap

It's been lost during the monorepo migration.

Note that building Rust plugins as a subproject is currently broken,
https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/171
but I'm going to fix that.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1290>

alphacombine: use the same allocation query data for both decoders

This allows downstream elements to set allocation query parameters for both
decoders, which should be always the same.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1277>

applemedia: Add ARGB64_BE, RGBA64_LE support to vtenc/vtdec

We can add this now that ARGB64_BE videoconvert support was added in:
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1247

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1214>

vtenc: Add FieldDetail properties for interlaced input

Standard interlace handling:
* If we have interlace-mode=interleaved and the field order, we just
  set it when creating the session
* If we have interlace-mode=(interleaved|mixed) and no field order, we
  set the field order on the first buffer

The encoder session does not support changing the FieldDetail after it
has started encoding frames, so we cannot support mixed streams
correctly.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1214>

vtenc: Add a property to forcibly ignore alpha values

This PropertyKey is not documented in any headers anywhere, so we need
to define it ourselves.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1214>

vtenc: Set colorimetry information

It looks like VideoToolbox doesn't support all our colorimetries.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1214>

applemedia: Add ProRes support to vtenc and vtdec

For vtdec, we continue to prefer NV12; else we pick whatever
downstream wants. In the special case where we're decoding 10-bit or
12-bit ProRes formats, we will prefer AYUV64.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1214>

vtenc: Improve error reporting in chain function

Otherwise it is quite difficult to figure out why the chain function
failed. Also assert not reached for case statements that should not be
hit.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1214>

vtdec: Remove dead code in switch statement

We never advertise these formats, so these cases will never be hit.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1214>

mediatype-video-raw: add ARGB64_LE/BE and variants

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1286>

Allow Strawberry Perl on Windows with Meson >= 0.60.0

Meson now explicitly blacklists pkg-config.bat from Strawberry Perl and
thus building GStreamer on Windows has been fixed.

This allows for instance to build GStreamer on GitHub Action Windows
runners that have Strawberry Perl by default.

It is also useful when needing to build OpenSSL on the same Windows
machine because it requires Perl.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1263>

video: Fix order of new video formats

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1282>

base: Fix some annotations

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1281>

Couple more g_memdup() -> g_memdup2() fixes

Fixes deprecation warnings with newer GLib versions.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1279>

dtls: don't use deprecated g_binding_get_source() with newer GLib versions

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1279>

Use g_pattern_spec_match() instead of g_pattern_match() which is deprecated since glib 2.70

Fixes compiler warnings with glib 2.70

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1279>

codecs: h265decoder: Fix per-slice leak

As documented, slice header parsed via gst_h265_parser_parse_slice_hdr()
should be cleared, otherwise it would result in memory leak.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1274>

va: Delay decoders downstream negotiation.

Delay decoders downstream negotiation just before an output frame
needs to be allocated.

This is required, are least for H.264 and H.265 decoders, since
codec_data might trigger a new sequence before finishing upstream
negotiation, and sink pad caps need to set before setting source pad
caps, particularly to forward HDR fields. The other decoders are
changed too in order to keep the same structure among them.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1257>

vabasedec: Move warning message to decoder's category.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1257>

va: Move common variable need_negotiation to GstBaseDec.

This is a common variable to all decoders, so it's sound to move it to
the base helper.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1257>

va: Move back parent_object to each element.

Using GstBaseDec hack to access the parent_object of each element in
the element itself is a bit fragile. It would be better to keep its
own parent object as the usual global variable. It would make it
resistant to code changes.

The GstBaseDec macro to access the parent object now it's internal to
base decoder.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1257>

video-format: Add support for ARGB64 LE/BE and similar variants

Co-authored-by: Sebastian Dröge <sebastian@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1247>

identity: Fix crash when receiving a gap event outside the current segment

We were checking if the start time of the gap event was
GST_CLOCK_TIME_NONE, which is superfluous because that cannot happen,
and then not checking if it was NONE after gst_segment_to_running_time,
which caused a crash if an identity received a gap event fully or
partially outside the current segment.

This patch was done in cooperation with:
Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1269>

rtptwcc: don't assume all PacketInfo->data are buffers

They can also be buffer lists

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1268>

neon: Allow building against neon 0.32.x

No API/ABI changes: https://github.com/notroj/neon/blob/0.32.0/NEWS#L3

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1267>

meson: require matching GStreamer dep versions for unstable development releases

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/929

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1244>

subprojects: fdk-aac: add fallback_url

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1244>

webrtcbin: fix check_negotiation computing on caps event

It seems logical that check_negotiation be true if received_caps
is *not* equal to the new caps.

Also clean up handling of transceivers' ssrc events, as this
patch triggered a leaky code path.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1233>

webrtcbin: connect input stream when receiving caps

.. if a current direction has already been set

When `webrtcbin` has created an offer based on codec_preferences,
it might not have received caps on its sinkpads by the time a
remote description is set, in which case we want to connect the
input stream upon actual reception of the caps instead.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1233>

webrtcbin: consider pads with trans->codec_preferences ready

.. when determining whether we can emit on-negotiation-needed

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1233>

mpeg4videoparse: fix criticals trying to insert configs that don't exist yet

With mpeg4videoparse drop=false config-interval=N|-1 we might be
trying to insert a config before we have actually received one,
in which case we'll try to map a NULL buffer which will generate
lots of criticals.

Fixes #855

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1265>

qtdemux: Fix text and closed-caption handling.

https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1182
broke text and and closed caption extraction when introducing WebVTT
support, by making the output buffers not have timestamps any more.

Fix that by making the process functions copy buffer metadata
when generating new output buffers.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1262>

msdk: Insert hdr sei at hevc encoder

There are two HDR SEIs defined in spec: mastering display colour volume and
content light level. Add insertion of HDR SEIs when they are available
during encoding.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1242>

webrtcbin: Also check data channel transport when collating connection state

https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/838

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1224>

vapostproc: Color fixation will choose othercaps' structure.

gst_va_fixate_format() will iterate all othercaps' structures to find
the one with less information lost at color conversion. If a structure
with same color format is found, the iteration stops. It's like a
smart truncation. Then, this function also will choose the caps
feature.

Later this structure is used fixate its size and no further truncation
is needed.

Don't intersect at fixate, since it kills possible resizing.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1261>

vapostproc: Ignore direction at orientation swapping.

It doesn't matter the direction of the negotiation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1261>

vapostproc: Consider video orientation for border calculation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1261>

decodebin3: Only unblock for GAP events and buffers.

An alternate fix for
https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1239
and https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1236
that makes it clear the intent is to only unblock on a GAP event, and
not any others.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1260>

python: Update audiotestsrc to return correct tuple on FlowReturn Errors

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1259>

avtpsrc: Retry receive with same buffer size

Without this patch in case of a retry recv() will be called with a
negative size argument.

Signed-off-by: Timo Wischer <timo.wischer@de.bosch.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1078>

parsebin: fix critical when sorting pads

If the pad does not have a current caps, get_pad() returns the query
caps which can be ANY. In such case the caps does not have any structure
resulting in a critical warning when calling gst_caps_get_structure().

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1254>