Release 1.19.901.19.90

70 files changed, 20784 insertions, 1774 deletions

diff --git a/meson.build b/meson.build
index 57c3f0070e..0ff3e479da 100644
--- a/meson.build
+++ b/meson.build
@@ -1,5 +1,5 @@
 project('gstreamer-full', 'c',
-  version : '1.19.3.1',
+  version : '1.19.90',
   meson_version : '>= 0.59',
   default_options : ['buildtype=debugoptimized',
                      # Needed due to https://github.com/mesonbuild/meson/issues/1889,
diff --git a/subprojects/gst-devtools/ChangeLog b/subprojects/gst-devtools/ChangeLog
index b5e81aa7c0..6644064833 100644
--- a/subprojects/gst-devtools/ChangeLog
+++ b/subprojects/gst-devtools/ChangeLog
@@ -1,3 +1,18 @@
+=== release 1.19.90 ===
+
+2022-01-28 14:28:35 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* NEWS:
+	* RELEASE:
+	* gst-devtools.doap:
+	* meson.build:
+	  Release 1.19.90
+
+2022-01-28 14:28:28 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* ChangeLog:
+	  Update ChangeLogs for 1.19.90
+
 2022-01-27 11:25:00 +0530  Nirbheek Chauhan <nirbheek@centricular.com>
 
 	* meson_options.txt:
diff --git a/subprojects/gst-devtools/NEWS b/subprojects/gst-devtools/NEWS
index e9f5227c15..1a512c1848 100644
--- a/subprojects/gst-devtools/NEWS
+++ b/subprojects/gst-devtools/NEWS
@@ -1,23 +1,19 @@
 GStreamer 1.20 Release Notes
 
 GStreamer 1.20 has not been released yet. It is scheduled for release in
-November 2021.
+late January / early February 2022.
 
 1.19.x is the unstable development version that is being developed in
-the git main branch and which will eventually result in 1.20, and 1.19.3
-is the current development release in that series
-
-It is expected that feature freeze will be in early October 2021,
-followed by one or two 1.19.9x pre-releases and the new 1.20 stable
-release around October/November 2021.
+the git main branch and which will eventually result in 1.20, and
+1.19.90 is the first release candidate in that series (1.20rc1).
 
 1.20 will be backwards-compatible to the stable 1.18, 1.16, 1.14, 1.12,
-1.10, 1.8, 1.6,, 1.4, 1.2 and 1.0 release series.
+1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series.
 
 See https://gstreamer.freedesktop.org/releases/1.20/ for the latest
 version of this document.
 
-Last updated: Monday 1 November 2021, 01:00 UTC (log)
+Last updated: Wednesday 26 January 2022, 01:00 UTC (log)
 
 Introduction
 
@@ -30,25 +26,971 @@ fixes and other improvements.
 
 Highlights
 
--   this section will be completed in due course
+-   Development in GitLab was switched to a single git repository
+    containing all the modules
+-   GstPlay: new high-level playback library, replaces GstPlayer
+-   WebM Alpha decoding support
+-   Encoding profiles can now be tweaked with additional
+    application-specified element properties
+-   Compositor: multi-threaded video conversion and mixing
+-   RTP header extensions: unified support in RTP depayloader and
+    payloader base classes
+-   SMPTE 2022-1 2-D Forward Error Correction support
+-   Smart encoding (passthrough) support for VP8, VP9, H.265 in
+    encodebin and transcodebin
+-   Runtime compatibility support for libsoup2 and libsoup3 (libsoup3
+    support experimental)
+-   Video decoder subframe support
+-   Video decoder automatic packet-loss, data corruption, and keyframe
+    request handling for RTP / WebRTC / RTSP
+-   MP4 and Matroska muxers now support profile/level/resolution changes
+    for H264/H265 input streams (i.e. codec data changing on the fly)
+-   MP4 muxing mode that initially creates a fragmented mp4 which is
+    converted to a regular mp4 on EOS
+-   Audio support for the WebKit Port for Embedded (WPE) web page source
+    element
+-   CUDA based video color space convert and rescale elements and
+    upload/download elements
+-   NVIDIA memory:NVMM support for OpenGL glupload and gldownload
+    elements
+-   Many WebRTC improvements
+-   The new VA-API plugin implemention fleshed out with more decoders
+    and new postproc elements
+-   AppSink API to retrieve events in addition to buffers and buffer
+    lists
+-   AppSrc gained more configuration options for the internal queue
+    (leakiness, limits in buffers and time, getters to read current
+    levels)
+-   Updated Rust bindings and many new Rust plugins
+-   Improved support for custom minimal GStreamer builds
+-   Support build against FFmpeg 5.0
+-   Linux Stateless CODEC support gained MPEG2 and VP9
+-   Windows Direct3D11/DXVA decoder gained AV1 and MPEG2 support
+-   Lots of new plugins, features, performance improvements and bug
+    fixes
 
 Major new features and changes
 
 Noteworthy new features and API
 
--   this section will be filled in in due course
+-   gst_element_get_request_pad() has been deprecated in favour of the
+    newly-added gst_element_request_pad_simple() which does the exact
+    same thing but has a less confusing name that hopefully makes clear
+    that the function request a new pad rather than just retrieves an
+    already-existing request pad.
+
+Development in GitLab was switched to a single git repository containing all the modules
+
+The GStreamer multimedia framework is a set of libraries and plugins
+split into a number of distinct modules which are released independently
+and which have so far been developed in separate git repositories in
+freedesktop.org GitLab.
+
+In addition to these separate git repositories there was a gst-build
+module that would use the Meson build systems’s subproject feature to
+download each individual module and then build everything in one go. It
+would also provide an uninstalled development environment that made it
+easy to work on GStreamer and use or test versions other than the
+system-installed GStreamer version.
+
+All of these modules have now (as of 28 September 2021) been merged into
+a single git repository (“Mono repository” or “monorepo”) which should
+simplify development workflows and continuous integration, especially
+where changes need to be made to multiple modules at once.
+
+This mono repository merge will primarily affect GStreamer developers
+and contributors and anyone who has workflows based on the GStreamer git
+repositories.
+
+The Rust bindings and Rust plugins modules have not been merged into the
+mono repository at this time because they follow a different release
+cycle.
+
+The mono repository lives in the existing GStreamer core git repository
+in GitLab in the new main branch and all future development will happen
+on this branch.
+
+Modules will continue to be released as separate tarballs.
+
+For more details, please see the GStreamer mono repository FAQ.
+
+GstPlay: new high-level playback library replacing GstPlayer
+
+-   GstPlay is a new high-level playback library that replaces the older
+    GstPlayer API. It is basically the same API as GstPlayer but
+    refactored to use bus messages for application notifications instead
+    of GObject signals. There is still a signal adapter object for those
+    who prefer signals. Since the existing GstPlayer API is already in
+    use in various applications, it didn’t seem like a good idea to
+    break it entirely. Instead a new API was added, and it is expected
+    that this new GstPlay API will be moved to gst-plugins-base in
+    future.
+
+-   The existing GstPlayer API is scheduled for deprecation and will be
+    removed at some point in the future (e.g. in GStreamer 1.24), so
+    application developers are urged to migrate to the new GstPlay API
+    at their earliest convenience.
+
+WebM alpha decoding
+
+-   Implement WebM alpha decoding (VP8/VP9 with alpha), which required
+    support and additions in various places. This is supported both with
+    software decoders and hardware-accelerated decoders.
+
+-   VP8/VP9 don’t support alpha components natively in the codec, so the
+    way this is implemented in WebM is by encoding the alpha plane with
+    transparency data as a separate VP8/VP9 stream. Inside the WebM
+    container (a variant of Matroska) this is coded as a single video
+    track with the “normal” VP8/VP9 video data making up the main video
+    data and each frame of video having an encoded alpha frame attached
+    to it as extra data ("BlockAdditional").
+
+-   matroskademux has been extended extract this per-frame alpha side
+    data and attach it in form of a GstVideoCodecAlphaMeta to the
+    regular video buffers. Note that this new meta is specific to this
+    VP8/VP9 alpha support and can’t be used to just add alpha support to
+    other codecs that don’t support it. Lastly, matroskademux also
+    advertises the fact that the streams contain alpha in the caps.
+
+-   The new codecalpha plugin contains various bits of infrastructure to
+    support autoplugging and debugging:
+
+    -   codecalphademux splits out the alpha stream from the metas on
+        the regular VP8/VP9 buffers
+    -   alphacombine takes two decoded raw video streams (one alpha, one
+        the regular video) and combines it into a video stream with
+        alpha
+    -   vp8alphadecodebin + vp9alphadecodebin are wrapper bins that use
+        the regular vp8dec and vp9dec software decoders to decode
+        regular and alpha streams and combine them again. To decodebin
+        these look like regular decoders which ju
+    -   The V4L2 CODEC plugin has stateless VP8/VP9 decoders that can
+        decode both alpha and non-alpha stream with a single decoder
+        instance
+
+-   A new AV12 video format was added which is basically NV12 with an
+    alpha plane, which is more convenient for many hardware-accelerated
+    decoders.
+
+-   Watch Nicolas Dufresne’s LCA 2022 talk “Bringing WebM Alpha support
+    to GStreamer” for all the details and a demo.
+
+RTP Header Extensions Base Class and Automatic Header Extension Handling in RTP Payloaders and Depayloaders
+
+-   RTP Header Extensions are specified in RFC 5285 and provide a way to
+    add small pieces of data to RTP packets in between the RTP header
+    and the RTP payload. This is often used for per-frame metadata,
+    extended timestamps or other application-specific extra data. There
+    are several commonly-used extensions specified in various RFCs, but
+    senders are free to put any kind of data in there, as long as sender
+    and receiver both know what that data is. Receivers that don’t know
+    about the header extensions will just skip the extra data without
+    ever looking at it. These header extensions can often be combined
+    with any kind of payload format, so may need to be supported by many
+    RTP payloader and depayloader elements.
+
+-   Inserting and extracting RTP header extension data has so far been a
+    bit inconvenient in GStreamer: There are functions to add and
+    retrieve RTP header extension data from RTP packets, but nothing
+    works automatically, even for common extensions. People would have
+    to do the insertion/extraction either in custom elements
+    before/after the RTP payloader/depayloader, or inside pad probes,
+    which isn’t very nice.
+
+-   This release adds various pieces of new infrastructure for generic
+    RTP header extension handling, as well as some implementations for
+    common extensions:
+
+    -   GstRTPHeaderExtension is a new helper base class for reading and
+        writing RTP header extensions. Nominally this subclasses
+        GstElement, but only so these extensions are stored in the
+        registry where they can be looked up by URI or name. They don’t
+        have pads and don’t get added to the pipeline graph as an
+        element.
+
+    -   "add-extension" and "clear-extension" action signals on RTP
+        payloaders and depayloaders for manual extension management
+
+    -   The "request-extension" signal will be emitted if an extension
+        is encountered that requires explicit mapping by the application
+
+    -   new "auto-header-extension" property on RTP payloaders and
+        depayloaders for automatic handling of known header extensions.
+        This is enabled by default. The extensions must be signalled via
+        caps / SDP.
+
+    -   RTP header extension implementations:
+
+        -   rtphdrextclientaudiolevel: Client-to-Mixer Audio Level
+            Indication (RFC 6464) (also see below)
+        -   rtphdrextcolorspace: Color Space extension, extends RTP
+            packets with color space and high dynamic range (HDR)
+            information
+        -   rtphdrexttwcc: Transport Wide Congestion Control support
+
+-   gst_rtp_buffer_remove_extension_data() is a new helper function to
+    remove an RTP header extension from an RTP buffer
+
+-   The existing gst_rtp_buffer_set_extension_data() now also supports
+    shrinking the extension data in size
+
+AppSink and AppSrc improvements
+
+-   appsink: new API to pull events out of appsink in addition to
+    buffers and buffer lists.
+
+    There was previously no way for users to receive incoming events
+    from appsink properly serialised with the data flow, even if they
+    are serialised events. The reason for that is that the only way to
+    intercept events was via a pad probe on the appsink sink pad, but
+    there is also internal queuing inside of appsink, so it’s difficult
+    to ascertain the right order of everything in all cases.
+
+    There is now a new "new-serialized-event" signal which will be
+    emitted when there’s a new event pending (just like the existing
+    "new-sample" signal). The "emit-signals" property must be set to
+    TRUE in order to activate this (but it’s also fine to just pull from
+    the application thread without using the signals).
+
+    gst_app_sink_pull_object() and gst_app_sink_try_pull_object() can be
+    used to pull out either an event or a new sample carrying a buffer
+    or buffer list, whatever is next in the queue.
+
+    EOS events will be filtered and will not be returned. EOS handling
+    can be done the ususal way, same as with _pull_sample().
+
+-   appsrc: allow configuration of internal queue limits in time and
+    buffers and add leaky mode.
+
+    There is internal queuing inside appsrc so the application thread
+    can push data into the element which will then be picked up by the
+    source element’s streaming thread and pushed into the pipeline from
+    that streaming thread. This queue is unlimited by default and until
+    now it was only possible to set a maximum size limit in bytes. When
+    that byte limit is reached, the pushing thread (application thread)
+    would be blocked until more space becomes available.
+
+    A limit in bytes is not particularly useful for many use cases, so
+    now it is possible to also configure limits in time and buffers
+    using the new "max-time" and "max-buffers" properties. Of course
+    there are also matching new read-only"current-level-buffers" and
+    "current-level-time properties" properties to query the current fill
+    level of the internal queue in time and buffers.
+
+    And as if that wasn’t enough the internal queue can also be
+    configured as leaky using the new "leaky-type" property. That way
+    when the queue is full the application thread won’t be blocked when
+    it tries to push in more data, but instead either the new buffer
+    will be dropped or the oldest data in the queue will be dropped.
+
+Better string serialization of nested GstCaps and GstStructures
+
+-   New string serialisation format for structs and caps that can handle
+    nested structs and caps properly by using brackets to delimit nested
+    items (e.g. some-struct, some-field=[nested-struct, nested=true]).
+    Unlike the default format the new variant can also support more than
+    one level of nesting. For backwards-compatibility reasons the old
+    format is still output by default when serialising caps and structs
+    using the existing API. The new functions gst_caps_serialize() and
+    gst_structure_serialize() can be used to output strings in the new
+    format.
+
+Convenience API for custom GstMetas
+
+-   New convenience API to register and create custom GstMetas:
+    gst_meta_register_custom() and gst_buffer_add_custom_meta(). Such
+    custom meta is backed by a GstStructure and does not require that
+    users of the API expose their GstMeta implementation as public API
+    for other components to make use of it. In addition, it provides a
+    simpler interface by ignoring the impl vs. api distinction that the
+    regular API exposes. This new API is meant to be the meta
+    counterpart to custom events and messages, and to be more convenient
+    than the lower-level API when the absolute best performance isn’t a
+    requirement. The reason it’s less performant than a “proper” meta is
+    that a proper meta is just a C struct in the end whereas this goes
+    through the GstStructure API which has a bit more overhead, which
+    for most scenarios is negligible however. This new API is useful for
+    experimentation or proprietary metas, but also has some limitations:
+    it can only be used if there’s a single producer of these metas;
+    it’s not allowed to register the same custom meta multiple times or
+    from multiple places.
+
+Additional Element Properties on Encoding Profiles
+
+-   GstEncodingProfile: The new "element-properties" and
+    gst_encoding_profile_set_element_properties() API allows
+    applications to set additional element properties on encoding
+    profiles to configure muxers and encoders. So far the encoding
+    profile template was the only place where this could be specified,
+    but often what applications want to do is take a ready-made encoding
+    profile shipped by GStreamer or the application and then tweak the
+    settings on top of that, which is now possible with this API. Since
+    applications can’t always know in advance what encoder element will
+    be used in the end, it’s even possible to specify properties on a
+    per-element basis.
+
+    Encoding Profiles are used in the encodebin, transcodebin and
+    camerabin elements and APIs to configure output formats (containers
+    and elementary streams).
+
+Audio Level Indication Meta for RFC 6464
+
+-   New GstAudioLevelMeta containing Audio Level Indication as per RFC
+    6464
+
+-   The level element has been updated to add GstAudioLevelMeta on
+    buffers if the "audio-level-meta" property is set to TRUE. This can
+    then in turn be picked up by RTP payloaders to signal the audio
+    level to receivers through RTP header extensions (see above).
+
+-   New Client-to-Mixer Audio Level Indication (RFC6464) RTP Header
+    Extension which should be automatically created and used by RTP
+    payloaders and depayloaders if their "auto-header-extension"
+    property is enabled and if the extension is part of the RTP caps.
+
+Automatic packet loss, data corruption and keyframe request handling for video decoders
+
+-   The GstVideoDecoder base class has gained various new APIs to
+    automatically handle packet loss and data corruption better by
+    default, especially in RTP, RTSP and WebRTC streaming scenarios, and
+    to give subclasses more control about how they want to handle
+    missing data:
+
+    -   Video decoder subclasses can mark output frames as corrupted via
+        the new GST_VIDEO_CODEC_FRAME_FLAG_CORRUPTED flag
+
+    -   A new "discard-corrupted-frames" property allows applications to
+        configure decoders so that corrupted frames are directly
+        discarded instead of being forwarded inside the pipeline. This
+        is a replacement for the "output-corrupt" property of the FFmpeg
+        decoders.
+
+    -   RTP depayloaders can now signal to decoders that data is missing
+        when sending GAP events for lost packets. GAP events can be sent
+        for various reason in a GStreamer pipeline. Often they are just
+        used to let downstream elements know that there isn’t a buffer
+        available at the moment, so downstream elements can move on
+        instead of waiting for one. They are also sent by RTP
+        depayloaders in the case that packets are missing, however, and
+        so far a decoder was not able to differentiate the two cases.
+        This has been remedied now: GAP events can be decorated with
+        gst_event_set_gap_flags() and GST_GAP_FLAG_MISSING_DATA to let
+        decoders now what happened, and decoders can then use that in
+        some cases to handle missing data better.
+
+    -   The GstVideoDecoder::handle_missing_data vfunc was added to
+        inform subclasses about packet loss or missing data and let them
+        handle it in their own way if they like.
+
+    -   gst_video_decoder_set_needs_sync_point() lets subclasses signal
+        that they need the stream to start with a sync point. If
+        enabled, the base class will discard all non-sync point frames
+        in the beginning and after a flush and does not pass them to the
+        subclass. Furthermore, if the first frame is not a sync point,
+        the base class will try and request a sync frame from upstream
+        by sending a force-key-unit event (see next items).
+
+    -   New "automatic-request-sync-points" and
+        "automatic-request-sync-point-flags" properties to automatically
+        request sync points when needed, e.g. on packet loss or if the
+        first frame is not a keyframe. Applications may want to enable
+        this on decoders operating in e.g. RTP/WebRTC/RTSP receiver
+        pipelines.
+
+    -   The new "min-force-key-unit-interval" property can be used to
+        ensure there’s a minimal interval between keyframe requests to
+        upstream (and/or the sender) and we’re not flooding the sender
+        with key unit requests.
+
+    -   gst_video_decoder_request_sync_point() allows subclasses to
+        request a new sync point (e.g. if they choose to do their own
+        missing data handling). This will still honour the
+        "min-force-key-unit-interval" property if set.
+
+Improved support for custom minimal GStreamer builds
+
+-   Element registration and registration of other plugin features
+    inside plugin init functions has been improved in order to
+    facilitate minimal custom GStreamer builds.
+
+-   A number of new macros have been added to declare and create
+    per-element and per-pluginfeature register functions in all plugins,
+    and then call those from the per-plugin plugin_init functions:
+
+    -   GST_ELEMENT_REGISTER_DEFINE,
+        GST_DEVICE_PROVIDER_REGISTER_DEFINE,
+        GST_DYNAMIC_TYPE_REGISTER_DEFINE, GST_TYPE_FIND_REGISTER_DEFINE
+        for the actual registration call with GStreamer
+    -   GST_ELEMENT_REGISTER, GST_DEVICE_PROVIDER_REGISTER,
+        GST_DYNAMIC_TYPE_REGISTER, GST_PLUGIN_STATIC_REGISTER,
+        GST_TYPE_FIND_REGISTER to call the registration function defined
+        by the REGISTER_DEFINE macro
+    -   GST_ELEMENT_REGISTER_DECLARE,
+        GST_DEVICE_PROVIDER_REGISTER_DECLARE,
+        GST_DYNAMIC_TYPE_REGISTER_DECLARE,
+        GST_TYPE_FIND_REGISTER_DECLARE to declare the registration
+        function defined by the REGISTER_DEFINE macro
+    -   and various variants for advanced use cases.
+
+-   This means that applications can call the per-element and
+    per-pluginfeature registration functions for only the elements they
+    need instead of registering plugins as a whole with all kinds of
+    elements that may not be required (e.g. encoder and decoder instead
+    of just decoder). In case of static linking all unused functions and
+    their dependencies would be removed in this case by the linker,
+    which helps minimise binary size for custom builds.
+
+-   gst_init() will automatically call a gst_init_static_plugins()
+    function if one exists.
+
+-   See the GStreamer static build documentation and Stéphane’s blog
+    post Generate a minimal GStreamer build, tailored to your needs for
+    more details.
 
 New elements
 
--   this section will be filled in in due course
+-   New aesdec and aesenc elements for AES encryption and decryption in
+    a custom format.
+
+-   New encodebin2 element with dynamic/sometimes source pads in order
+    to support the option of doing the muxing outside of encodebin,
+    e.g. in combination with a splitmuxsink.
+
+-   New fakeaudiosink and videocodectestsink elements for testing and
+    debugging (see below for more details)
+
+-   rtpisacpay, rtpisacdepay: new RTP payloader and depayloader for iSAC
+    audio codec
+
+-   rtpst2022-1-fecdec, rtpst2022-1-fecenc: new elements providing SMPTE
+    2022-1 2-D Forward Error Correction. More details in Mathieu’s blog
+    post.
+
+-   isac: new plugin wrapping the Internet Speech Audio Codec reference
+    encoder and decoder from the WebRTC project.
+
+-   asio: plugin for Steinberg ASIO (Audio Streaming Input/Output) API
+
+-   gssrc, gssink: add source and sink for Google Cloud Storage
+
+-   onnx: new plugin to apply ONNX neural network models to video
+
+-   openaptx: aptX and aptX-HD codecs using libopenaptx (v0.2.0)
+
+-   qroverlay, debugqroverlay: new elements that allows overlaying data
+    on top of video in form of a QR code
+
+-   cvtracker: new OpenCV-based tracker element
+
+-   av1parse, vp9parse: new parsers for AV1 and VP9 video
+
+-   va: work on the new VA-API plugin implementation for
+    hardware-accelerated video decoding and encoding has continued at
+    pace, with various new decoders and filters having joined the
+    initial vah264dec:
+
+    -   vah265dec: VA-API H.265 decoder
+    -   vavp8dec: VA-API VP8 decoder
+    -   vavp9dec: VA-API VP9 decoder
+    -   vaav1dec: VA-API AV1 decoder
+    -   vampeg2dec: VA-API MPEG-2 decoder
+    -   vadeinterlace: : VA-API deinterlace filter
+    -   vapostproc: : VA-API postproc filter (color conversion,
+        resizing, cropping, color balance, video rotation, skin tone
+        enhancement, denoise, sharpen)
+
+    See Víctor’s blog post “GstVA in GStreamer 1.20” for more details
+    and what’s coming up next.
+
+-   vaapiav1dec: new AV1 decoder element (in gstreamer-vaapi)
+
+-   msdkav1dec: hardware-accelerated AV1 decoder using the Intel Media
+    SDK / oneVPL
+
+-   nvcodec plugin for NVIDIA NVCODEC API for hardware-accelerated video
+    encoding and decoding:
+
+    -   cudaconvert, cudascale: new CUDA based video color space convert
+        and rescale elements
+    -   cudaupload, cudadownload: new helper elements for memory
+        transfer between CUDA and system memory spaces
+    -   nvvp8sldec, nvvp9sldec: new GstCodecs-based VP8/VP9 decoders
+
+-   Various new hardware-accelerated elements for Windows:
+
+    -   d3d11screencapturesrc: new desktop capture element, including a
+        GstDeviceProvider implementation to enumerate/select target
+        monitors for capture.
+    -   d3d11av1dec and d3d11mpeg2dec: AV1 and MPEG-2 decoders
+    -   d3d11deinterlace: deinterlacing filter
+    -   d3d11compositor: video composing element
+    -   see Windows section below for more details
+
+-   new Rust plugins:
+
+    -   audiornnoise: Removes noise from an audio stream
+    -   awstranscribeparse: Parses AWS audio transcripts into timed text
+        buffers
+    -   ccdetect: Detects if valid closed captions are present in a
+        closed captions stream
+    -   cea608tojson: Converts CEA-608 Closed Captions to a JSON
+        representation
+    -   cmafmux: CMAF fragmented MP4 muxer
+    -   dashmp4mux: DASH fragmented MP4 muxer
+    -   isofmp4mux: ISO fragmented MP4 muxer
+    -   ebur128level: EBU R128 Loudness Level Measurement
+    -   ffv1dec: FFV1 video decoder
+    -   gtk4paintablesink: GTK4 video sink, which provides a
+        GdkPaintable that can be rendered in various widgets
+    -   hlssink3: HTTP Live Streaming sink
+    -   hrtfrender: Head-Related Transfer Function (HRTF) renderer
+    -   hsvdetector: HSV colorspace detector
+    -   hsvfilter: HSV colorspace filter
+    -   jsongstenc: Wraps buffers containing any valid top-level JSON
+        structures into higher level JSON objects, and outputs those as
+        ndjson
+    -   jsongstparse: Parses ndjson as output by jsongstenc
+    -   jsontovtt: converts JSON to WebVTT subtitles
+    -   regex: Applies regular expression operations on text
+    -   roundedcorners: Adds rounded corners to video
+    -   spotifyaudiosrc: Spotify source
+    -   textahead: Display upcoming text buffers ahead (e.g. for
+        Karaoke)
+    -   transcriberbin: passthrough bin that transcribes raw audio to
+        closed captions using awstranscriber and puts the captions as
+        metas onto the video
+    -   tttojson: Converts timed text to a JSON representation
+    -   uriplaylistbin: Playlist source bin
+    -   webpdec-rs: WebP image decoder with animation support
+
+-   New plugin codecalpha with elements to assist with WebM Alpha
+    decoding
+
+    -   codecalphademux: Split stream with GstVideoCodecAlphaMeta into
+        two streams
+    -   alphacombine: Combine two raw video stream (I420 or NV12) as one
+        stream with alpha channel (A420 or AV12)
+    -   vp8alphadecodebin: A bin to handle software decoding of VP8 with
+        alpha
+    -   vp9alphadecodebin: A bin to handle software decoding of VP9 with
+        alpha
+
+-   New hardware accelerated elements for Linux:
+
+    -   v4l2slmpeg2dec: Support for Linux Stateless MPEG2 decoders
+    -   v4l2slvp9dec: Support for Linux Stateless VP9 decoders
+    -   v4l2slvp8alphadecodebin: Support HW accelerated VP8 with alpha
+        layer decoding
+    -   v4l2slvp9alphadecodebin: Support HW accelerated VP9 with alpha
+        layer decoding
 
 New element features and additions
 
--   this section will be filled in in due course
+-   assrender: handle more font mime types; better interaction with
+    matroskademux for embedded fonts
+
+-   audiobuffersplit: Add support for specifying output buffer size in
+    bytes (not just duration)
+
+-   audiolatency: new "samplesperbuffer" property so users can configure
+    the number of samples per buffer. The default value is 240 samples
+    which is equivalent to 5ms latency with a sample rate of 48000,
+    which might be larger than actual buffer size of audio capture
+    device.
+
+-   audiomixer, audiointerleave, GstAudioAggregator: now keep a count of
+    samples that are dropped or processed as statistic and can be made
+    to post QoS messages on the bus whenever samples are dropped by
+    setting the "qos-messages" property on input pads.
+
+-   audiomixer, compositor: improved handling of new inputs added at
+    runtime. New API was added to the GstAggregator base class to allow
+    subclasses to opt into an aggregation mode where inactive pads are
+    ignored when processing input buffers
+    (gst_aggregator_set_ignore_inactive_pads(),
+    gst_aggregator_pad_is_inactive()). An “inactive pad” in this context
+    is a pad which, in live mode, hasn’t yet received a first buffer,
+    but has been waited on at least once. What would happen usually in
+    this case is that the aggregator would wait for data on this pad
+    every time, up to the maximum configured latency. This would
+    inadvertently push mixer elements in live mode to the configured
+    latency envelope and delay processing when new inputs are added at
+    runtime until these inputs have actually produced data. This is
+    usually undesirable. With this new API, new inputs can be added
+    (requested) and configured and they won’t delay the data processing.
+    Applications can opt into this new behaviour by setting the
+    "ignore-inactive-pads" property on compositor, audiomixer or other
+    GstAudioAggregator-based elements.
+
+-   cccombiner: implement “scheduling” of captions. So far cccombiner’s
+    behaviour was essentially that of a funnel: it strictly looked at
+    input timestamps to associate together video and caption buffers.
+    Now it will try to smoothly schedule caption buffers in order to
+    have exactly one per output video buffer. This might involve
+    rewriting input captions, for example when the input is CDP then
+    sequence counters are rewritten, time codes are dropped and
+    potentially re-injected if the input video frame had a time code
+    meta. This can also lead to the input drifting from synchronization,
+    when there isn’t enough padding in the input stream to catch up. In
+    that case the element will start dropping old caption buffers once
+    the number of buffers in its internal queue reaches a certain limit
+    (configurable via the "max-scheduled" property). The new original
+    funnel-like behaviour can be restored by setting the "scheduling"
+    property to FALSE.
+
+-   ccconverter: new "cdp-mode" property to specify which sections to
+    include in CDP packets (timecode, CC data, service info). Various
+    software, including ffmpeg’s Decklink support, fails parsing CDP
+    packets that contain anything but CC data in the CDP packets.
+
+-   clocksync: new "sync-to-first" property for automatic timestamp
+    offset setup: if set clocksync will set up the "ts-offset" value
+    based on the first buffer and the pipeline’s running time when the
+    first buffer arrived. The newly configured "ts-offset" in this case
+    would be the value that allows outputting the first buffer without
+    waiting on the clock. This is useful for example to feed a non-live
+    input into an already-running pipeline.
+
+-   compositor:
+
+    -   multi-threaded input conversion and compositing. Set the
+        "max-threads" property to activate this.
+    -   new "sizing-policy" property to support display aspect ratio
+        (DAR)-aware scaling. By default the image is scaled to fill the
+        configured destination rectangle without padding and without
+        keeping the aspect ratio. With sizing-policy=keep-aspect-ratio
+        the input image is scaled to fit the destination rectangle
+        specified by GstCompositorPad:{xpos, ypos, width, height}
+        properties preserving the aspect ratio. As a result, the image
+        will be centered in the destination rectangle with padding if
+        necessary.
+    -   new "zero-size-is-unscaled" property on input pads. By default
+        pad width=0 or pad height=0 mean that the stream should not be
+        scaled in that dimension. But if the "zero-size-is-unscaled"
+        property is set to FALSE a width or height of 0 is instead
+        interpreted to mean that the input image on that pad should not
+        be composited, which is useful when creating animations where an
+        input image is made smaller and smaller until it disappears.
+    -   improved handling of new inputs at runtime via
+        "ignore-inactive-pads"property (see above for details)
+    -   allow output format with alpha even if none of the inputs have
+        alpha (also glvideomixer and other GstVideoAggregator
+        subclasses)
+
+-   dashsink: add h265 codec support and signals for allowing custom
+    playlist/fragment output
+
+-   decodebin3:
+
+    -   improved decoder selection, especially for hardware decoders
+    -   make input activation “atomic” when adding inputs dynamically
+    -   better interleave handling: take into account decoder latency
+        for interleave size
+
+-   decklink:
+
+    -   Updated DeckLink SDK to 11.2 to support DeckLink 8K Pro
+    -   decklinkvideosrc:
+        -   More accurate and stable capture timestamps: use the
+            hardware reference clock time when the frame was finished
+            being captured instead of a clock time much further down the
+            road.
+        -   Automatically detect widescreen vs. normal NTSC/PAL
+
+-   encodebin:
+
+    -   add “smart encoding” support for H.265, VP8 and VP9 (i.e. only
+        re-encode where needed and otherwise pass through encoded video
+        as-is).
+    -   H264/H265 smart encoding improvements: respect user-specified
+        stream-format, but if not specified default to avc3/hvc1 with
+        in-band SPS/PPS/VPS signalling for more flexibility.
+    -   new encodebin2 element with dynamic/sometimes source pads in
+        order to support the option of doing the muxing outside of
+        encodebin, e.g. in combination with splitmuxsink.
+    -   add APIs to set element properties on encoding profiles (see
+        below)
+
+-   errorignore: new "ignore-eos" property to also ignore FLOW_EOS from
+    downstream elements
+
+-   giosrc: add support for growing source files: applications can
+    specify that the underlying file being read is growing by setting
+    the "is-growing" property. If set, the source won’t EOS when it
+    reaches the end of the file, but will instead start monitoring it
+    and will start reading data again whenever a change is detected. The
+    new "waiting-data" and "done-waiting-data" signals keep the
+    application informed about the current state.
+
+-   gtksink, gtkglsink:
+
+    -   scroll event support: forwarded as navigation events into the
+        pipeline
+    -   "video-aspect-ratio-override" property to force a specific
+        aspect ratio
+    -   "rotate-method" property and support automatic rotation based on
+        image tags
+
+-   identity: new "stats" property allows applications to retrieve the
+    number of bytes and buffers that have passed through so far.
+
+-   interlace: add support for more formats, esp 10-bit, 12-bit and
+    16-bit ones
+
+-   jack: new "low-latency" property for automatic latency-optimized
+    setting and "port-names" property to select ports explicitly
+
+-   jpegdec: support output conversion to RGB using libjpeg-turbo (for
+    certain input files)
+
+-   line21dec:
+
+    -   "mode" property to control whether and how detected closed
+        captions should be inserted in the list of existing close
+        caption metas on the input frame (if any): add, drop, or
+        replace.
+    -   "ntsc-only" property to only look for captions if video has NTSC
+        resolution
+
+-   line21enc: new "remove-caption-meta" to remove metas from output
+    buffers after encoding the captions into the video data; support for
+    CDP closed captions
+
+-   matroskademux, matroskamux: Add support for ffv1, a lossless
+    intra-frame video coding format.
+
+-   matroskamux: accept in-band SPS/PPS/VPS for H264 and H265
+    (i.e. stream-format avc3 and hev1) which allows on-the-fly
+    profile/level/resolution changes.
+
+-   matroskamux: new "cluster-timestamp-offset" property, useful for use
+    cases where the container timestamps should map to some absolute
+    wall clock time, for example.
+
+-   rtpsrc: add "caps" property to allow explicit setting of the caps
+    where needed
+
+-   mpegts: support SCTE-35 passthrough via new "send-scte35-events"
+    property on MPEG-TS demuxer tsdemux. When enabled, SCTE 35 sections
+    (eg ad placement opportunities) are forwarded as events donwstream
+    where they can be picked up again by mpegtsmux. This required a
+    semantic change in the SCTE-35 section API: timestamps are now in
+    running time instead of muxer pts.
+
+-   tsdemux: Handle PCR-less MPEG-TS streams; more robust timestamp
+    handling in certain corner cases and for poorly muxed streams.
+
+-   mpegtsmux:
+
+    -   More conformance improvements to make MPEG-TS analyzers happy:
+        -   PCR timing accuracy: Improvements to the way mpegtsmux
+            outputs PCR observations in CBR mode, so that a PCR
+            observation is always inserted when needed, so that we never
+            miss the configured pcr-interval, as that triggers various
+            MPEG-TS analyser errors.
+        -   Improved PCR/SI scheduling
+    -   Don’t write PCR until PAT/PMT are output to make sure streams
+        start cleanly with a PAT/PMT.
+    -   Allow overriding the automatic PMT PID selection via
+        application-supplied PMT_%d fields in the prog-map
+        structure/property.
+
+-   mp4mux:
+
+    -   new "first-moov-then-finalise" mode for fragmented output where
+        the output will start with a self-contained moov atom for the
+        first fragment, and then produce regular fragments. Then at the
+        end when the file is finalised, the initial moov is invalidated
+        and a new moov is written covering the entire file. This way the
+        file is a “fragmented mp4” file while it is still being written
+        out, and remains playable at all times, but at the end it is
+        turned into a regular mp4 file (with former fragment headers
+        remaining as unused junk data in the file).
+    -   support H.264 avc3 and H.265 hvc1 stream formats as input where
+        the codec data is signalled in-band inside the bitstream instead
+        of caps/file headers.
+    -   support profile/level/resolution changes for H264/H265 input
+        streams (i.e. codec data changing on the fly). Each codec_data
+        is put into its own SampleTableEntry inside the stsd, unless the
+        input is in avc3 stream format in which case it’s written
+        in-band an not in the headers.
+
+-   multifilesink: new ""min-keyframe-distance"" property to make
+    minimum distance between keyframes in next-file=key-frame mode
+    configurable instead of hard-coding it to 10 seconds.
+
+-   mxfdemux has seen a big refactoring to support non-frame wrappings
+    and more accurate timestamp/seek handling for some formats
+
+-   msdk plugin for hardware-accelerated video encoding and decoding
+    using the Intel Media SDK:
+
+    -   oneVPL support (Intel oneAPI Video Processing Library)
+    -   AV1 decoding support
+    -   H.264 decoder now supports constrained-high and progressive-high
+        profiles
+    -   H.264 encoder:
+        -   more configuration options (properties):
+            "intra-refresh-type", "min-qp" , "max-qp", "p-pyramid",
+            "dblk-idc"
+    -   H.265 encoder:
+        -   can output main-still-picture profile
+        -   now inserts HDR SEIs (mastering display colour volume and
+            content light level)
+        -   more configuration options (properties):
+            "intra-refresh-type", "min-qp" , "max-qp", "p-pyramid",
+            "b-pyramid", "dblk-idc", "transform-skip"
+        -   support for RGB 10bit format
+    -   External bitrate control in encoders
+    -   Video post proc element msdkvpp gained support for 12-bit pixel
+        formats P012_LE, Y212_LE and Y412_LE
+
+-   nvh264sldec: interlaced stream support
+
+-   openh264enc: support main, high, constrained-high and
+    progressive-high profiles
+
+-   openjpeg: support for multithreaded decoding and encoding
+
+-   rtspsrc: now supports IPv6 also for tunneled mode (RTSP-over-HTTP);
+    new "ignore-x-server-reply" property to ignore the
+    x-server-ip-address server header reply in case of HTTP tunneling,
+    as it is often broken.
+
+-   souphttpsrc: Runtime compatibility support for libsoup2 and
+    libsoup3. libsoup3 is the latest major version of libsoup, but
+    libsoup2 and libsoup3 can’t co-exist in the same process because
+    there is no namespacing or versioning for GObject types. As a
+    result, it would be awkward if the GStreamer souphttpsrc plugin
+    linked to a specific version of libsoup, because it would only work
+    with applications that use the same version of libsoup. To make this
+    work, the soup plugin now tries to determine the libsoup version
+    used by the application (and its other dependencies) at runtime on
+    systems where GStreamer is linked dynamically. libsoup3 support is
+    still considered somewhat experimental at this point.
+
+-   srtsrc, srtsink: add signals for the application to accept/reject
+    incoming connections
+
+-   timeoverlay: new elapsed-running-time time mode which shows the
+    running time since the first running time (and each flush-stop).
+
+-   udpsrc: new timestamping mode to retrieve packet receive timestamps
+    from the kernel via socket control messages (SO_TIMESTAMPNS) on
+    supported platforms
+
+-   uritranscodebin: new setup-source and element-setup signals for
+    applications to configure elements used
+
+-   v4l2codecs plugin gained support for 4x4 and 32x32 tile formats
+    enabling some platforms or direct renders. Important memory usage
+    improvement.
+
+-   v4l2slh264dec now implements the final Linux uAPI as shipped on
+    Linux 5.11 and later.
+
+-   valve: add "drop-mode" property and provide two new modes of
+    operation: in drop-mode=forward-sticky-events sticky events
+    (stream-start, segment, tags, caps, etc.) are forwarded downstream
+    even when dropping is enabled; drop-mode=transform-to-gap will in
+    addition also convert buffers into gap events when dropping is
+    enabled, which lets downstream elements know that time is advancing
+    and might allow for preroll in many scenarios. By default all events
+    and all buffers are dropped when dropping is enabled, which can
+    cause problems with caps negotiation not progressing or branches not
+    prerolling when dropping is enabled.
+
+-   videocrop: support for many more pixel formats, e.g. planar YUV
+    formats with > 8bits and GBR* video formats; can now also accept
+    video not backed by system memory as long as downstream supports the
+    GstCropMeta
+
+-   videotestsrc: new smpte-rp-219 pattern for SMPTE75 RP-219 conformant
+    color bars
+
+-   vp8enc: finish support for temporal scalability: two new properties
+    ("temporal-scalability-layer-flags",
+    "temporal-scalability-layer-sync-flags") and a unit change on the
+    "temporal-scalability-target-bitrate" property (now expects bps);
+    also make temporal scalability details available to RTP payloaders
+    as buffer metadata.
+
+-   vp9enc: new properties to tweak encoder performance:
+
+    -   "aq-mode" to configure adaptive quantization modes
+    -   "frame-parallel-decoding" to configure whether to create a
+        bitstream that reduces decoding dependencies between frames
+        which allows staged parallel processing of more than one video
+        frames in the decoder. (Defaults to TRUE)
+    -   "row-mt", "tile-columns" and "tile-rows" so multithreading can
+        be enabled on a per-tile basis, instead of on a per tile-column
+        basis. In combination with the new "tile-rows" property, this
+        allows the encoder to make much better use of the available CPU
+        power.
+
+-   vp9dec, vp9enc: add support for 10-bit 4:2:0 and 4:2:2 YUV, as well
+    as 8-bit 4:4:4
+
+-   vp8enc, vp9enc now default to “good quality” for the deadline
+    property rather then “best quality”. Having the deadline set to best
+    quality causes the encoder to be absurdly slow, most real-life users
+    will prefer good-enough quality with better performance instead.
+
+-   wpesrc:
+
+    -   implement audio support: a new sometimes source pad will be
+        created for each audio stream created by the web engine.
+    -   move wpesrc to wpevideosrc and add a wrapper bin wpesrc to also
+        support audio
+    -   also handles web:// URIs now (same as cefsrc)
+    -   post messages with the estimated load progress on the bus
 
-Plugin and library moves
+-   x265enc: add negative DTS support, which means timestamps are now
+    offset by 1h same as with x264enc
+
+RTP Payloaders and Depayloaders
+
+-   rtpisacpay, rtpisacdepay: new RTP payloader and depayloader for iSAC
+    audio codec
+
+-   rtph264depay:
+
+    -   new "request-keyframe" property to make the depayloader
+        automatically request a new keyframe from the sender on packet
+        loss, consistent with the new property on rtpvp8depay.
+    -   new "wait-for-keyframe" property to make depayloader wait for a
+        new keyframe at the beginning and after packet loss (only
+        effective if the depayloader outputs AUs), consistent with the
+        existing property on rtpvp8depay.
+
+-   rtpopuspay, rtpopusdepay: support libwebrtc-compatible multichannel
+    audio in addition to the previously supported multichannel audio
+    modes
+
+-   rtpopuspay: add DTX (Discontinuous Transmission) support
+
+-   rtpvp8depay: new "request-keyframe" property to make the depayloader
+    automatically request a new keyframe from the sender on packet loss.
+
+-   rtpvp8pay: temporal scaling support
 
--   this section will be filled in in due course
+-   rtpvp9depay: Improved SVC handling (aggregate all layers)
+
+RTP Infrastructure
+
+-   rtpst2022-1-fecdec, rtpst2022-1-fecenc: new elements providing SMPTE
+    2022-1 2-D Forward Error Correction. More details in Mathieu’s blog
+    post.
+
+-   rtpreddec: BUNDLE support
+
+-   rtpredenc, rtpulpfecenc: add support for Transport-wide Congestion
+    Control (TWCC)
+
+-   rtpsession: new "twcc-feedback-interval" property to allow RTCP TWCC
+    reports to be scheduled on a timer instead of per marker-bit.
+
+Plugin and library moves
 
 -   There were no plugin moves or library moves in this cycle.
 
@@ -56,8 +998,6 @@ Plugin removals
 
 The following elements or plugins have been removed:
 
--   this section will be filled in in due course
-
 -   The ofa audio fingerprinting plugin has been removed. The MusicIP
     database has been defunct for years so this plugin is likely neither
     useful nor used by anyone.
@@ -71,135 +1011,575 @@ The following elements or plugins have been removed:
 
 Miscellaneous API additions
 
--   this section will be filled in in due course
+Core
+
+-   gst_buffer_new_memdup() is a convenience function for the
+    widely-used gst_buffer_new_wrapped(g_memdup(data,size),size)
+    pattern.
+
+-   gst_caps_features_new_single() creates a new single GstCapsFeatures,
+    avoiding the need to use the vararg function with NULL terminator
+    for simple cases.
+
+-   gst_element_type_set_skip_documentation() can be used by plugins to
+    signal that certain elements should not be included in the GStreamer
+    plugin documentation. This is useful for plugins where elements are
+    registered dynamically based on hardware capabilities and/or where
+    the available plugins and properties vary from system to system.
+    This is used in the d3d11 plugin for example to ensure that only the
+    list of default elements is advertised in the documentation.
+
+-   gst_type_find_suggest_empty_simple() is a new convenience function
+    for typefinders for cases where there’s only a media type and no
+    other fields.
+
+-   New API to create elements and set properties at construction time,
+    which is not only convenient, but also allows GStreamer elements to
+    have construct-only properties: gst_element_factory_make_full(),
+    gst_element_factory_make_valist(),
+    gst_element_factory_make_with_properties(),
+    gst_element_factory_create_full(),
+    gst_element_factory_create_valist(),
+    gst_element_factory_create_with_properties().
+
+-   GstSharedTaskPool: new “shared” task pool subclass with slightly
+    different default behaviour than the existing GstTaskPool which
+    would create unlimited number of threads for new tasks. The shared
+    taskpool creates up to N threads (default: 1) and then distributes
+    pending tasks to those threads round-robin style, and blocks if no
+    thread is available. It is possible to join tasks. This can be used
+    by plugins to implement simple multi-threaded processing and is used
+    for the new multi-threaded video conversion and compositing done in
+    GstVideoAggregator, videoconverter and compositor.
+
+Plugins Base Utils library
+
+-   GstDiscoverer:
+
+    -   gst_discoverer_container_info_get_tags() was added to retrieve
+        global/container tags (vs. per-stream tags). Per-Stream tags can
+        be retrieved via the existing
+        gst_discoverer_stream_info_get_tags().
+        gst_discoverer_info_get_tags(), which for many files returns a
+        confusing mix of stream and container tags, has been deprecated
+        in favour of the container/stream-specific functions.
+    -   gst_discoverer_stream_info_get_stream_number() returns a unique
+        integer identifier for a given stream within the given
+        GstDiscoverer context. (If this matches the stream number inside
+        the container bitstream that’s by coincidence and not by
+        design.)
+
+-   gst_pb_utils_get_caps_description_flags() can be used to query
+    whether certain caps represent a container, audio, video, image,
+    subtitles, tags, or something else. This only works for formats
+    known to GStreamer.
+
+-   gst_pb_utils_get_file_extension_from_caps() returns a possible file
+    extension for given caps.
+
+-   gst_codec_utils_h264_get_profile_flags_level(): Parses profile,
+    flags, and level from H264 AvcC codec_data. The format of H264 AVCC
+    extradata/sequence_header is documented in the ITU-T H.264
+    specification section 7.3.2.1.1 as well as in ISO/IEC 14496-15
+    section 5.3.3.1.2.
+
+-   gst_codec_utils_caps_get_mime_codec() to convert caps to a RFC 6381
+    compatible MIME codec string codec. Useful for providing the codecs
+    field inside the Content-Type HTTP header for containerized formats,
+    such as mp4 or matroska.
+
+GStreamer OpenGL integration library and plugins
+
+-   glcolorconvert: added suppport for converting the video formats
+    A420, AV12, BGR, BGRA, RGBP and BGRP.
+
+-   Added support to GstGLBuffer for persistent buffer mappings where a
+    Pixel Buffer Object (PBO) can be mapped by both the CPU and the GPU.
+    This removes a memcpy() when uploading textures or vertices
+    particularly when software decoders (e.g. libav) are direct
+    rendering into our memory. Improves transfer performance
+    significantly. Requires OpenGL 4.4, GL_ARB_buffer_storage or
+    GL_EXT_buffer_storage
+
+-   Added various helper functions for handling 4x4 matrices of affine
+    transformations as used by GstVideoAffineTransformationMeta.
+
+-   Add support to GstGLContext for allowing the application to control
+    the config (EGLConfig, GLXConfig, etc) used when creating the OpenGL
+    context. This allows the ability to choose between RGB16 or RGB10A2
+    or RGBA8 back/front buffer configurations that were previously
+    hardcoded. GstGLContext also supports retrieving the configuration
+    it was created with or from an externally provide OpenGL context
+    handle. This infrastructure is also used to create a compatible
+    config from an application/externally provided OpenGL context in
+    order to improve compatibility with other OpenGL frameworks and GUI
+    toolkits. A new environment variable GST_GL_CONFIG was also added to
+    be able to request a specific configuration from the command line.
+    Note: different platforms will have different functionality
+    available.
+
+-   Add support for choosing between EGL and WGL at runtime when running
+    on Windows. Previously this was a build-time switch. Allows use in
+    e.g. Gtk applications on Windows that target EGL/ANGLE without
+    recompiling GStreamer. gst_gl_display_new_with_type() can be used by
+    applications to choose a specific display type to use.
+
+-   Build fixes to explicitly check for Broadcom-specific libraries on
+    older versions of the Raspberry Pi platform. The Broadcom OpenGL ES
+    and EGL libraries have different filenames. Using the vc4 Mesa
+    driver on the Raspberry Pi is not affected.
+
+-   Added support to glupload and gldownload for transferring RGBA
+    buffers using the memory:NVMM available on the Nvidia Tegra family
+    of embedded devices.
+
+-   Added support for choosing libOpenGL and libGLX as used in a GLVND
+    environment on unix-based platforms. This allows using desktop
+    OpenGL and EGL without pulling in any GLX symbols as would be
+    required with libGL.
+
+Video library
+
+-   New raw video formats:
+
+    -   AV12 (NV12 with alpha plane)
+    -   RGBP and BGRP (planar RGB formats)
+    -   ARGB64 variants with specified endianness instead of host
+        endianness:
+        -   ARGB64_LE, ARGB64_BE
+        -   RGBA64_BE, RGBA64_LE
+        -   BGRA64_BE, BGRA64_LE
+        -   ABGR64_BE, ABGR64_LE
+
+-   gst_video_orientation_from_tag() is new convenience API to parse the
+    image orientation from a GstTagList.
+
+-   GstVideoDecoder subframe support (see below)
+
+-   GstVideoCodecState now also carries some HDR metadata
+
+-   Ancillary video data: implement transform functions for AFD/Bar
+    metas, so they will be forwarded in more cases
+
+MPEG-TS library
+
+This library only handles section parsing and such, see above for
+changes to the actual mpegtsmux and mpegtsdemux elements.
+
+-   many additions and improvements to SCTE-35 section parsing
+-   new API for fetching extended descriptors:
+    gst_mpegts_find_descriptor_with_extension()
+-   add support for SIT sections (Selection Information Tables)
+-   expose event-from-section constructor gst_event_new_mpegts_section()
+-   parse Audio Preselection Descriptor needed for Dolby AC-4
+
+GstWebRTC library + webrtcbin
+
+-   Change the way in which sink pads and transceivers are matched
+    together to support easier usage. If a pad is created without a
+    specific index (i.e. using sink_%u as the pad template), then an
+    available compatible transceiver will be searched for. If a specific
+    index is requested (i.e. sink_1) then if a transceiver for that
+    m-line already exists, that transceiver must match the new sink pad
+    request. If there is no transceiver available in either scenario, a
+    new transceiver is created. If a mixture of both sink_1 and sink_%u
+    requests result in an impossible situation, an error will be
+    produced at pad request time or from create offer/answer.
+
+-   webrtcbin now uses regular ICE nomination instead of libnice’s
+    default of aggressive ICE nomination. Regular ICE nomination is the
+    default recommended by various relevant standards and improves
+    connectivity in specific network scenarios.
+
+-   Add support for limiting the port range used for RTP with the
+    addition of the min-rtp-port and max-rtp-port properties on the ICE
+    object.
+
+-   Expose the SCTP transport as a property on webrtcbin to more closely
+    match the WebRTC specification.
+
+-   Added support for taking into account the data channel transport
+    state when determining the value of the "connection-state" property.
+    Previous versions of the WebRTC spec did not include the data
+    channel state when computing this value.
+
+-   Add configuration for choosing the size of the underlying sockets
+    used for transporting media data
+
+-   Always advertise support for the transport-cc RTCP feedback protocol
+    as rtpbin supports it. For full support, the configured caps (input
+    or through codec-preferences) need to include the relevant RTP
+    header extension.
+
+-   Numerous fixes to caps and media handling to fail-fast when an
+    incompatible situation is detected.
+
+-   Improved support for attaching the required media after a remote
+    offer has been set.
+
+-   Add support for dynamically changing the amount of FEC used for a
+    particular stream.
+
+-   webrtcbin now stops further SDP processing at the first error it
+    encounters.
+
+-   Completed support for either local or the remote closing a data
+    channel.
+
+-   Various fixes when performing BUNDLEing of the media streams in
+    relation to RTX and FEC usage.
+
+-   Add support for writing out QoS DSCP marking on outgoing packets to
+    improve reliability in some network scenarios.
+
+-   Improvements to the statistics returned by the get-stats signal
+    including the addition of the raw statistics from the internal
+    RTPSource, the TWCC stats when available.
+
+-   The webrtc library does not expose any objects anymore with public
+    fields. Instead properties have been added to replace that
+    functionality. If you are accessing such fields in your application,
+    switch to the corresponding properties.
+
+GstCodecs and Video Parsers
+
+-   Support for render delays to improve throughput across all CODECs
+    (used with NVDEC and V4L2).
+-   lots of improvements to parsers and the codec parsing decoder base
+    classes (H264, H265, VP8, VP9, AV1, MPEG-2) used for various
+    hardware-accelerate decoder APIs.
+
+Bindings support
+
+-   gst_allocation_params_new() allocates a GstAllocationParams struct
+    on the heap. This should only be used by bindings (and freed via
+    gst_allocation_params_free() then). In C code you would allocate
+    this on the stack and only init it in place.
+
+-   gst_debug_log_literal() can be used to log a string to the debug log
+    without going through any printf format expansion and associated
+    overhead. This is mostly useful for bindings such as the Rust
+    bindings which may have done their own formatting already .
+
+-   Provide non-inlined versions of refcounting APIs for various
+    GStreamer mini objects, so that they can be consumed by bindings
+    (e.g. gstreamer-sharp): gst_buffer_ref, gst_buffer_unref,
+    gst_clear_buffer, gst_buffer_copy, gst_buffer_replace,
+    gst_buffer_list_ref, gst_buffer_list_unref, gst_clear_buffer_list,
+    gst_buffer_list_copy, gst_buffer_list_replace, gst_buffer_list_take,
+    gst_caps_ref, gst_caps_unref, gst_clear_caps, gst_caps_replace,
+    gst_caps_take, gst_context_ref, gst_context_unref, gst_context_copy,
+    gst_context_replace, gst_event_replace, gst_event_steal,
+    gst_event_take, gst_event_ref, gst_event_unref, gst_clear_event,
+    gst_event_copy, gst_memory_ref, gst_memory_unref, gst_message_ref,
+    gst_message_unref, gst_clear_message, gst_message_copy,
+    gst_message_replace, gst_message_take, gst_promise_ref,
+    gst_promise_unref, gst_query_ref, gst_query_unref, gst_clear_query,
+    gst_query_copy, gst_query_replace, gst_query_take, gst_sample_ref,
+    gst_sample_unref, gst_sample_copy, gst_tag_list_ref,
+    gst_tag_list_unref, gst_clear_tag_list, gst_tag_list_replace,
+    gst_tag_list_take, gst_uri_copy, gst_uri_ref, gst_uri_unref,
+    gst_clear_uri.
+
+-   expose a GType for GstMiniObject
+
+-   gst_device_provider_probe() now returns non-floating device object
+
+API Deprecations
+
+-   gst_element_get_request_pad() has been deprecated in favour of the
+    newly-added gst_element_request_pad_simple() which does the exact
+    same thing but has a less confusing name that hopefully makes clear
+    that the function request a new pad rather than just retrieves an
+    already-existing request pad.
+
+-   gst_discoverer_info_get_tags(), which for many files returns a
+    confusing mix of stream and container tags, has been deprecated in
+    favour of the container-specific and stream-specific functions,
+    gst_discoverer_container_info_get_tags() and
+    gst_discoverer_stream_info_get_tags().
+
+-   gst_video_sink_center_rect() was deprecated in favour of the more
+    generic newly-added gst_video_center_rect().
+
+-   The GST_MEMORY_FLAG_NO_SHARE flag has been deprecated, as it tends
+    to cause problems and prevents sub-buffering. If pooling or lifetime
+    tracking is required, memories should be allocated through a custom
+    GstAllocator instead of relying on the lifetime of the buffers the
+    memories were originally attached to, which is fragile anyway.
+
+-   The GstPlayer high-level playback library is being replaced with the
+    new GstPlay library (see above). GstPlayer should be considered
+    deprecated at this point and will be marked as such in the next
+    development cycle. Applications should be ported to GstPlay.
+
+-   Gstreamer Editing Services: ges_video_transition_set_border(),
+    ges_video_transition_get_border()
+    ges_video_transition_set_inverted()
+    ges_video_transition_is_inverted() have been deprecated, use
+    ges_timeline_element_set_children_properties() instead.
 
 Miscellaneous performance, latency and memory optimisations
 
--   this section will be filled in in due course
+More video conversion fast paths
 
-Miscellaneous other changes and enhancements
+-   v210 ↔ I420, YV12, Y42B, UYVY and YUY2
+-   A420 → RGB
+
+Less jitter when waiting on the system clock
+
+-   Better system clock wait accuracy, less jitter: where available,
+    clock_nanosleep is used for higher accuracy for waits below 500
+    usecs, and waits below 2ms will first use the regular waiting system
+    and then clock_nanosleep for the remainder. The various wait
+    implementation have a latency ranging from 50 to 500+ microseconds.
+    While this is not a major issue when dealing with a low number of
+    waits per second (for ex: video), it does introduce a non-negligible
+    jitter for synchronisation of higher packet rate systems.
+
+Video decoder subframe support
 
--   this section will be filled in in due course
+-   The GstVideoDecoder base class gained API to process input at the
+    sub-frame level. That way video decoders can start decoding slices
+    before they have received the full input frame in its entirety (to
+    the extent this is supported by the codec, of course). This helps
+    with CPU utilisation and reduces latency.
 
-Tracing framework and debugging improvements
+-   This functionality is now being used in the OpenJPEG JPEG 2000
+    decoder, the FFmpeg H.264 decoder (in case of NAL-aligned input) and
+    the OpenMAX H.264/H.265 decoders (in case of NAL-aligned input).
 
--   this section will be filled in in due course
+Miscellaneous other changes and enhancements
+
+-   GstDeviceMonitor no longer fails to start just because one of the
+    device providers failed to start. That could happen for example on
+    systems where the pulseaudio device provider is installed, but
+    pulseaudio isn’t actually running but ALSA is used for audio
+    instead. In the same vein the device monitor now keeps track of
+    which providers have been started (via the new
+    gst_device_provider_is_started()) and only stops actually running
+    device providers when stopping the device monitor.
+
+-   On embedded systems it can be useful to create a registry that can
+    be shared and read by multiple processes running as different users.
+    It is now possible to set the new GST_REGISTRY_MODE environment
+    variable to specify the file mode for the registry file, which by
+    default is set to be only user readable/writable.
+
+-   GstNetClientClock will signal lost sync in case the remote time
+    resets (e.g. because device power cycles), by emitting the “synced”
+    signal with synced=FALSE parameter, so applications can take action.
+
+-   gst_value_deserialize_with_pspec() allows deserialization with a
+    hint for what the target GType should be. This allows for example
+    passing arrays of flags through the command line or
+    gst_util_set_object_arg(), eg: foo="<bar,bar+baz>".
+
+-   It’s now allowed to create an empty GstVideoOverlayComposition
+    without any rectangles by passing a NULL rectangle to
+    gst_video_overlay_composition_new(). This is useful for bindings and
+    simplifies application code in some places.
+
+Tracing framework, debugging and testing improvements
+
+-   New factories tracer to list loaded elements (and other plugin
+    features). This can be useful to collect a list of elements needed
+    for an application, which then in turn can be used to create a
+    tailored minimal GStreamer build that contains just the elements
+    needed and nothing else.
+-   New plugin-feature-loaded tracing hook for use by tracers like the
+    new factories tracer
+
+-   GstHarness: Add gst_harness_set_live() so that harnesses can be set
+    to non-live and return is-live=false in latency queries if needed.
+    Default behaviour is to always return is-live=true in latency
+    queries.
+
+-   navseek: new "hold-eos" property. When enabled, the element will
+    hold back an EOS event until the next keystroke (via navigation
+    events). This can be used to keep a video sink showing the last
+    frame of a video pipeline until a key is pressed instead of tearing
+    it down immediately on EOS.
+
+-   New fakeaudiosink element: mimics an audio sink and can be used for
+    testing and CI pipelines on systems where no audio system is
+    installed or running. It differs from fakesink in that it only
+    support audio caps and syncs to the clock by default like a normal
+    audio sink. It also implements the GstStreamVolume interface like
+    most audio sinks do.
+
+-   New videocodectestsink element for video codec conformance testing:
+    Calculates MD5 checksums for video frames and skips any padding
+    whilst doing so. Can optionally also write back the video data with
+    padding removed into a file for easy byte-by-byte comparison with
+    reference data.
 
 Tools
 
--   this section will be filled in in due course
+gst-inspect-1.0
+
+-   Can sort the list of plugins by passing --sort=name as command line
+    option
+
+gst-launch-1.0
+
+-   will now error out on top-level properties that don’t exist and
+    which were silently ignored before
+-   On Windows the high-resolution clock is enabled now, which provides
+    better clock and timer performance on Windows (see Windows section
+    below for more details).
+
+gst-play-1.0
+
+-   New --start-position command line argument to start playback from
+    the specified position
+-   Audio can be muted/unmuted in interactive mode by pressing the m
+    key.
+-   On Windows the high-resolution clock is enabled now (see Windows
+    section below for more details)
+
+gst-device-monitor-1.0
+
+-   New --include-hidden command line argument to also show “hidden”
+    device providers
+
+ges-launch-1.0
+
+-   New interactive mode that allows seeking and such. Can be disabled
+    by passing the --no-interactive argument on the command line.
+-   Option to forward tags
+-   Allow using an existing clip to determine the rendering format (both
+    topology and profile) via new --profile-from command line argument.
 
 GStreamer RTSP server
 
--   this section will be filled in in due course
+-   GstRTSPMediaFactory gained API to disable RTCP
+    (gst_rtsp_media_factory_set_enable_rtcp(), "enable-rtcp" property).
+    Previously RTCP was always allowed for all RTSP medias. With this
+    change it is possible to disable RTCP completely, no matter if the
+    client wants to do RTCP or not.
+
+-   Make a mount point of / work correctly. While not allowed by the
+    RTSP 2 spec, the RTSP 1 spec is silent on this and it is used in the
+    wild. It is now possible to use / as a mount path in
+    gst-rtsp-server, e.g. rtsp://example.com/ would work with this now.
+    Note that query/fragment parts of the URI are not necessarily
+    correctly handled, and behaviour will differ between various
+    client/server implementations; so use it if you must but don’t bug
+    us if it doesn’t work with third party clients as you’d hoped.
+
+-   multithreading fixes (races, refcounting issues, deadlocks)
+
+-   ONVIF audio backchannel fixes
+
+-   ONVIF trick mode optimisations
+
+-   rtspclientsink: new "update-sdp" signal that allows updating the SDP
+    before sending it to the server via ANNOUNCE. This can be used to
+    add additional metadata to the SDP, for example. The order and
+    number of medias must not be changed, however.
 
 GStreamer VAAPI
 
--   this section will be filled in in due course
+-   new AV1 decoder element (vaapiav1dec)
+
+-   H264 decoder: handle stereoscopic 3D video with frame packing
+    arrangement SEI messages
+
+-   H265 encoder: added Screen Content Coding extensions support
+
+-   H265 decoder: gained MAIN_444_12 profile support (decoded to
+    Y412_LE), and 4:2:2 12-bits support (decoded to Y212_LE)
+
+-   vaapipostproc: gained BT2020 color standard support
+
+-   vaapidecode: now generates caps templates dynamically at runtime in
+    order to advertise actually supported caps instead of all
+    theoretically supported caps.
+
+-   GST_VAAPI_DRM_DEVICE environment variable to force a specified DRM
+    device when a DRM display is used. It is ignored when other types of
+    displays are used. By default /dev/dri/renderD128 is used for DRM
+    display.
 
 GStreamer OMX
 
--   this section will be filled in in due course
+-   subframe support in H.264/H.265 decoders
 
 GStreamer Editing Services and NLE
 
--   this section will be filled in in due course
+-   framepositioner: new "operator" property to access blending modes in
+    the compositor
+-   timeline: Implement snapping to markers
+-   smart-mixer: Add support for d3d11compositor and glvideomixer
+-   titleclip: add "draw-shadow" child property
+-   ges:// URI support to define a timeline from a description.
+-   command-line-formatter
+    -   Add track management to timeline description
+    -   Add keyframe support
+-   ges-launch-1.0:
+    -   Add an interactive mode where we can seek etc…
+    -   Add option to forward tags
+    -   Allow using an existing clip to determine the rendering format
+        (both topology and profile) via new --profile-from command line
+        argument.
+-   Fix static build
 
 GStreamer validate
 
--   this section will be filled in in due course
+-   report: Add a way to force backtraces on reports even if not a
+    critical issue (GST_VALIDATE_ISSUE_FLAGS_FORCE_BACKTRACE)
+-   Add a flag to gst_validate_replace_variables_in_string() allow
+    defining how to resolve variables in structs
+-   Add gst_validate_bin_monitor_get_scenario() to get the bin monitor
+    scenario, which is useful for applications that use Validate
+    directly.
+-   Add an expected-values parameter to wait, message-type=XX allowing
+    more precise filtering of the message we are waiting for.
+-   Add config file support: each test can now use a config file for the
+    given media file used to test.
+-   Add support to check properties of object properties
+-   scenario: Add an "action-done" signal to signal when an action is
+    done
+-   scenario: Add a "run-command" action type
+-   scenario: Allow forcing running action on idle from scenario file
+-   scenario: Allow iterating over arrays in foreach
+-   scenario: Rename ‘interlaced’ action to ‘non-blocking’
+-   scenario: Add a non-blocking flag to the wait signal
 
 GStreamer Python Bindings
 
--   this section will be filled in in due course
+-   Fixes for Python 3.10
+-   Various build fixes
+-   at least one known breaking change caused by g-i annotation changes
+    (see below)
 
 GStreamer C# Bindings
 
--   this section will be filled in in due course
+-   Fix GstDebugGraphDetails enum
+-   Updated to latests GtkSharp
+-   Updated to include GStreamer 1.20 API
 
 GStreamer Rust Bindings and Rust Plugins
 
-The GStreamer Rust bindings are released separately with a different
-release cadence that’s tied to gtk-rs, but the latest release has
-already been updated for the upcoming new GStreamer 1.20 API.
-
-gst-plugins-rs, the module containing GStreamer plugins written in Rust,
-has also seen lots of activity with many new elements and plugins.
-
-What follows is a list of elements and plugins available in
-gst-plugins-rs, so people don’t miss out on all those potentially useful
-elements that have no C equivalent.
-
--   FIXME: add new elements
-
-Rust audio plugins
-
--   audiornnoise: New element for audio denoising which implements the
-    noise removal algorithm of the Xiph RNNoise library, in Rust
--   rsaudioecho: Port of the audioecho element from gst-plugins-good
-    rsaudioloudnorm: Live audio loudness normalization element based on
-    the FFmpeg af_loudnorm filter
--   claxondec: FLAC lossless audio codec decoder element based on the
-    pure-Rust claxon implementation
--   csoundfilter: Audio filter that can use any filter defined via the
-    Csound audio programming language
--   lewtondec: Vorbis audio decoder element based on the pure-Rust
-    lewton implementation
-
-Rust video plugins
-
--   cdgdec/cdgparse: Decoder and parser for the CD+G video codec based
-    on a pure-Rust CD+G implementation, used for example by karaoke CDs
--   cea608overlay: CEA-608 Closed Captions overlay element
--   cea608tott: CEA-608 Closed Captions to timed-text (e.g. VTT or SRT
-    subtitles) converter
--   tttocea608: CEA-608 Closed Captions from timed-text converter
--   mccenc/mccparse: MacCaption Closed Caption format encoder and parser
--   sccenc/sccparse: Scenarist Closed Caption format encoder and parser
--   dav1dec: AV1 video decoder based on the dav1d decoder implementation
-    by the VLC project
--   rav1enc: AV1 video encoder based on the fast and pure-Rust rav1e
-    encoder implementation
--   rsflvdemux: Alternative to the flvdemux FLV demuxer element from
-    gst-plugins-good, not feature-equivalent yet
--   rsgifenc/rspngenc: GIF/PNG encoder elements based on the pure-Rust
-    implementations by the image-rs project
-
-Rust text plugins
-
--   textwrap: Element for line-wrapping timed text (e.g. subtitles) for
-    better screen-fitting, including hyphenation support for some
-    languages
-
-Rust network plugins
-
--   reqwesthttpsrc: HTTP(S) source element based on the Rust
-    reqwest/hyper HTTP implementations and almost feature-equivalent
-    with the main GStreamer HTTP source souphttpsrc
--   s3src/s3sink: Source/sink element for the Amazon S3 cloud storage
--   awstranscriber: Live audio to timed text transcription element using
-    the Amazon AWS Transcribe API
-
-Generic Rust plugins
-
--   sodiumencrypter/sodiumdecrypter: Encryption/decryption element based
-    on libsodium/NaCl
--   togglerecord: Recording element that allows to pause/resume
-    recordings easily and considers keyframe boundaries
--   fallbackswitch/fallbacksrc: Elements for handling potentially
-    failing (network) sources, restarting them on errors/timeout and
-    showing a fallback stream instead
--   threadshare: Set of elements that provide alternatives for various
-    existing GStreamer elements but allow to share the streaming threads
-    between each other to reduce the number of threads
--   rsfilesrc/rsfilesink: File source/sink elements as replacements for
-    the existing filesrc/filesink elements
+-   The GStreamer Rust bindings are released separately with a different
+    release cadence that’s tied to gtk-rs, but the latest release has
+    already been updated for the upcoming new GStreamer 1.20 API (v1_20
+    feature).
+
+-   gst-plugins-rs, the module containing GStreamer plugins written in
+    Rust, has also seen lots of activity with many new elements and
+    plugins. See the New Elements section above for a list of new Rust
+    elements.
 
 Build and Dependencies
 
 -   Meson 0.59 or newer is required to build GStreamer now.
 
--   FIXME: this section will be filled in in due course
+-   The GLib requirement has been bumped to GLib 2.56 or newer (from
+    March 2018).
+
+-   The wpe plugin now requires wpe >= 2.28 and wpebackend-fdo >= 1.8
 
 Explicit opt-in required for build of certain plugins with (A)GPL dependencies
 
@@ -211,11 +1591,9 @@ even if the required dependencies are available.
 See Building plugins with (A)GPL-licensed dependencies for more details
 and a non-exhaustive list of plugins affected.
 
-gst-build: replaced by Monorepo
+gst-build: replaced by mono repository
 
--   this section will be filled in in due course
-
--   FIXME: describe + link to Monorepo FAQ
+See mono repository section above and the GStreamer mono repository FAQ.
 
 Cerbero
 
@@ -223,76 +1601,254 @@ Cerbero is a meta build system used to build GStreamer plus dependencies
 on platforms where dependencies are not readily available, such as
 Windows, Android, iOS and macOS.
 
-General improvements
+General Cerbero improvements
 
--   this section will be filled in in due course
+-   Plugin removed: libvisual
+-   New plugins: rtpmanagerbad and rist
 
-macOS / iOS
+macOS / iOS specific Cerbero improvements
 
--   this section will be filled in in due course
+-   XCode 12 support
+-   macOS OS release support is now future-proof, similar to iOS
+-   macOS Apple Silicon (ARM64) cross-compile support has been added
+-   macOS Apple Silicon (ARM64) native support is currently experimental
 
-Windows
+Windows specific Cerbero improvements
 
--   this section will be filled in in due course
+-   Visual Studio 2022 support has been added
+-   bootstrap is faster since it requires building fewer build-tools
+    recipes on Windows
+-   package is faster due to better scheduling of recipe stages and
+    elimination of unnecessary autotools regeneration
+-   The following plugins are no longer built on Windows:
+    -   a52dec (another decoder is still available in libav)
+    -   dvdread
+    -   resindvd
 
 Windows MSI installer
 
--   this section will be filled in in due course
+-   no major changes
 
-Linux
+Linux specific Cerbero improvements
 
--   this section will be filled in in due course
+-   Fedora, Debian OS release support is now more future-proof
+-   Amazon Linux 2 support has been added
 
-Android
+Android specific Cerbero improvements
 
--   this section will be filled in in due course
+-   no major changes
 
 Platform-specific changes and improvements
 
 Android
 
--   this section will be filled in in due course
+-   No major changes
 
 macOS and iOS
 
--   this section will be filled in in due course
+-   applemedia: add ProRes support to vtenc and vtdec
 
 Windows
 
--   this section will be filled in in due course
+-   On Windows the high-resolution clock is enabled now in the
+    gst-launch-1.0 and gst-play-1.0 command line tools, which provides
+    better clock and timer performance on Windows, at the cost of higher
+    power consumption. By default, without the high-resolution clock
+    enabled, the timer precision on Windows is system-dependent and may
+    be as bad as 15ms which is not good enough for many multimedia
+    applications. Developers may want to do the same in their Windows
+    applications if they think it’s a good idea for their application
+    use case, and depending on the Windows version they target. This is
+    not done automatically by GStreamer because on older Windows
+    versions (pre-Windows 10) this affects a global Windows setting and
+    also there’s a power consumption vs. performance trade-off that may
+    differ from application to application.
+
+-   dxgiscreencapsrc now supports resolution changes
+
+-   The wasapi2 audio plugin was rewritten and now has a higher rank
+    than the old wasapi plugin since it has a number of additional
+    features such as automatic stream routing, and no
+    known-but-hard-to-fix issues. The plugin is always built if the
+    Windows 10 SDK is available now.
+
+-   The wasapi device providers now detect and notify dynamic device
+    additions/removals
+
+-   d3d11screencapturesrc: new desktop capture element, including
+    GstDeviceProvider implementation to enumerate/select target monitors
+    for capture.
+
+-   Direct3D11/DXVA decoder now supports AV1 and MPEG2 codecs
+    (d3d11av1dec, d3d11mpeg2dec)
+
+-   VP9 decoding got more reliable and stable thanks to a newly written
+    codec parser
+
+-   Support for decoding interlaced H.264/AVC streams
+
+-   Hardware-accelerated video deinterlacing (d3d11deinterlace) and
+    video mixing (d3d11compositor)
+
+-   Video mixing with the Direct3D11 API (d3d11compositor)
+
+-   MediaFoundation API based hardware encoders gained the ability to
+    receive Direct3D11 textures as an input
+
+-   Seungha’s blog post “GStreamer ❤ Windows: A primer on the cool stuff
+    you’ll find in the 1.20 release” describes many of the
+    Windows-related improvements in more detail
 
 Linux
 
--   this section will be filled in in due course
+-   bluez: LDAC Bluetooth audio codec support in a2dpsink and avdtpsink,
+    as well as an LDAC RTP payloader (rtpldacpay) and an LDAC audio
+    encoder (ldacenc)
+
+-   kmssink: gained support for NV24, NV61, RGB16/BGR16 formats;
+    auto-detect NVIDIA Tegra driver
 
 Documentation improvements
 
--   this section will be filled in in due course
+-   hardware-accelerated GPU plugins will now no longer always list all
+    the element variants for all available GPUs, since those are
+    system-dependent and it’s confusing for users to see those in the
+    documentation just because the GStreamer developer who generated the
+    docs had multiple GPUs to play with at the time. Instead just show
+    the default elements.
 
-Possibly Breaking Changes
+Possibly Breaking and Other Noteworthy Behavioural Changes
+
+-   gst_parse_launch(), gst_parse_bin_from_description() and friends
+    will now error out when setting properties that don’t exist on
+    top-level bins. They were silently ignored before.
+
+-   The GstWebRTC library does not expose any objects anymore with
+    public fields. Instead properties have been added to replace that
+    functionality. If you are accessing such fields in your application,
+    switch to the corresponding properties.
 
--   FIXME: this section will be filled in in due course
--   MPEG-TS SCTE-35 API changes (FIXME: flesh out)
--   gst_parse_launch() and friends now error out on non-existing
-    properties on top-level bins where they would silently fail and
-    ignore those before.
 -   playbin and uridecodebin now emit the source-setup signal before the
     element is added to the bin and linked so that the source element is
     already configured before any scheduling query comes in, which is
-    useful for elements such as appsrc or giostreamsrc. (Merge Request)
+    useful for elements such as appsrc or giostreamsrc.
+
+-   The source element inside urisourcebin (used inside uridecodebin3
+    which is used inside playbin3) is no longer called "source". This
+    shouldn’t affect anyone hopefully, because there’s a "setup-source"
+    signal to configure the source element and no one should rely on
+    names of internal elements anyway.
+
+-   The vp8enc element now expects bps (bits per second) for the
+    "temporal-scalability-target-bitrate" property, which is consistent
+    with the "target-bitrate" property. Since additional configuration
+    is required with modern libvpx to make temporal scaling work anyway,
+    chances are that very few people will have been using this property
+
+-   vp8enc and vp9enc now default to “good quality” for the "deadline"
+    property rather then “best quality”. Having the deadline set to best
+    quality causes the encoder to be absurdly slow, most real-life users
+    will want the good quality tradeoff instead.
+
+-   The experimental GstTranscoder library API in gst-plugins-bad was
+    changed from a GObject signal-based notification mechanism to a
+    GstBus/message-based mechanism akin to GstPlayer/GstPlay.
+
+-   MPEG-TS SCTE-35 API: semantic change for SCTE-35 splice commands:
+    timestamps passed by the application should be in running time now,
+    since users of the API can’t really be expected to predict the local
+    PTS of the muxer.
+
+-   The GstContext used by souphttpsrc to share the session between
+    multiple element instances has changed. Previously it provided
+    direct access to the internal SoupSession object, now it only
+    provides access to an opaque, internal type. This change is
+    necessary because SoupSession is not thread-safe at all and can’t be
+    shared safely between arbitrary external code and souphttpsrc.
+
+-   Python bindings: GObject-introspection related Annotation fixes have
+    led to a case of a GstVideo.VideoInfo-related function signature
+    changing in the Python bindings (possibly one or two other cases
+    too). This is for a function that should never have been exposed in
+    the first place though, so the bindings are being updated to throw
+    an exception in that case, and the correct replacement API has been
+    added in form of an override.
 
 Known Issues
 
--   this section will be filled in in due course
-
--   There are a couple of known WebRTC-related regressions/blockers:
-
-    -   webrtc: DTLS setup with Chrome is broken
-    -   webrtcbin: First keyframe is usually lost
+-   nothing in particular at this point (but also see possibly breaking
+    changes section above)
 
 Contributors
 
--   this section will be filled in in due course
+Aaron Boxer, Adam Leppky, Adam Williamson, Alba Mendez, Alejandro
+González, Aleksandr Slobodeniuk, Alexander Vandenbulcke, Alex Ashley,
+Alicia Boya García, Andika Triwidada, Andoni Morales Alastruey, Andrew
+Wesie, Andrey Moiseev, Antonio Ospite, Antonio Rojas, Arthur Crippa
+Búrigo, Arun Raghavan, Ashley Brighthope, Axel Kellermann, Baek, Bastien
+Nocera, Bastien Reboulet, Benjamin Gaignard, Bing Song, Binh Truong,
+Biswapriyo Nath, Brad Hards, Brad Smith, Brady J. Garvin, Branko
+Subasic, Camilo Celis Guzman, Chris Bass, ChrisDuncanAnyvision, Chris
+White, Corentin Damman, Daniel Almeida, Daniel Knobe, Daniel Stone,
+david, David Fernandez, David Keijser, David Phung, Devarsh Thakkar,
+Dinesh Manajipet, Dmitry Samoylov, Dmitry Shusharin, Dominique Martinet,
+Doug Nazar, Ederson de Souza, Edward Hervey, Emmanuel Gil Peyrot,
+Enrique Ocaña González, Ezequiel Garcia, Fabian Orccon, Fabrice
+Fontaine, Fernando Jimenez Moreno, Florian Karydes, Francisco Javier
+Velázquez-García, François Laignel, Frederich Munch, Fredrik Pålsson,
+George Kiagiadakis, Georg Lippitsch, Göran Jönsson, Guido Günther,
+Guillaume Desmottes, Guiqin Zou, Haakon Sporsheim, Haelwenn (lanodan)
+Monnier, Haihao Xiang, Haihua Hu, Havard Graff, He Junyan, Helmut
+Januschka, Henry Wilkes, Hosang Lee, Hou Qi, Ignacio Casal Quinteiro,
+Igor Kovalenko, Ilya Kreymer, Imanol Fernandez, Jacek Tomaszewski, Jade
+Macho, Jakub Adam, Jakub Janků, Jan Alexander Steffens (heftig), Jan
+Schmidt, Jason Carrete, Jason Pereira, Jay Douglass, Jeongki Kim, Jérôme
+Laheurte, Jimmi Holst Christensen, Johan Sternerup, John Hassell, John
+Lindgren, John-Mark Bell, Jonathan Matthew, Jordan Petridis, Jose
+Quaresma, Julian Bouzas, Julien, Kai Uwe Broulik, Kasper Steensig
+Jensen, Kellermann Axel, Kevin Song, Khem Raj, Knut Inge Hvidsten, Knut
+Saastad, Kristofer Björkström, Lars Lundqvist, Lawrence Troup, Lim Siew
+Hoon, Lucas Stach, Ludvig Rappe, Luis Paulo Fernandes de Barros, Luke
+Yelavich, Mads Buvik Sandvei, Marc Leeman, Marco Felsch, Marek Vasut,
+Marian Cichy, Marijn Suijten, Marius Vlad, Markus Ebner, Mart Raudsepp,
+Matej Knopp, Mathieu Duponchelle, Matthew Waters, Matthieu De Beule,
+Mengkejiergeli Ba, Michael de Gans, Michael Olbrich, Michael Tretter,
+Michal Dzik, Miguel Paris, Mikhail Fludkov, mkba, Nazar Mokrynskyi,
+Nicholas Jackson, Nicola Murino, Nicolas Dufresne, Niklas Hambüchen,
+Nikolay Sivov, Nirbheek Chauhan, Olivier Blin, Olivier Crete, Olivier
+Crête, Paul Goulpié, Per Förlin, Peter Boba, P H, Philippe Normand,
+Philipp Zabel, Pieter Willem Jordaan, Piotrek Brzeziński, Rafał
+Dzięgiel, Rafostar, raghavendra, Raghavendra, Raju Babannavar, Raleigh
+Littles III, Randy Li, Randy Li (ayaka), Ratchanan Srirattanamet, Raul
+Tambre, reed.lawrence, Ricky Tang, Robert Rosengren, Robert Swain, Robin
+Burchell, Roman Sivriver, R S Nikhil Krishna, Ruben Gonzalez, Ruslan
+Khamidullin, Sanchayan Maity, Scott Moreau, Sebastian Dröge, Sergei
+Kovalev, Seungha Yang, Sid Sethupathi, sohwan.park, Sonny Piers, Staz M,
+Stefan Brüns, Stéphane Cerveau, Stephan Hesse, Stian Selnes, Stirling
+Westrup, Théo MAILLART, Thibault Saunier, Tim, Timo Wischer, Tim-Philipp
+Müller, Tim Schneider, Tobias Ronge, Tom Schoonjans, Tulio Beloqui,
+tyler-aicradle, U. Artie Eoff, Ung, Val Doroshchuk, VaL Doroshchuk,
+Víctor Manuel Jáquez Leal, Vivek R, Vivia Nikolaidou, Vivienne
+Watermeier, Vladimir Menshakov, Will Miller, Wim Taymans, Xabier
+Rodriguez Calvar, Xavier Claessens, Xℹ Ruoyao, Yacine Bandou, Yinhang
+Liu, youngh.lee, youngsoo.lee, yychao, Zebediah Figura, Zhang yuankun,
+Zhang Yuankun, Zhao, Zhao Zhili, , Aleksandar Topic, Antonio Ospite,
+Bastien Nocera, Benjamin Gaignard, Brad Hards, Carlos Falgueras García,
+Célestin Marot, Corentin Damman, Corentin Noël, Daniel Almeida, Daniel
+Knobe, Danny Smith, Dave Piché, Dmitry Osipenko, Fabrice Fontaine,
+fjmax, Florian Zwoch, Guillaume Desmottes, Haihua Hu, Heinrich Kruger,
+He Junyan, Jakub Adam, James Cowgill, Jan Alexander Steffens (heftig),
+Jean Felder, Jeongki Kim, Jiri Uncovsky, Joe Todd, Jordan Petridis,
+Krystian Wojtas, Marc-André Lureau, Marcin Kolny, Marc Leeman, Mark
+Nauwelaerts, Martin Reboredo, Mathieu Duponchelle, Matthew Waters,
+Mengkejiergeli Ba, Michael Gruner, Nicolas Dufresne, Nirbheek Chauhan,
+Olivier Crête, Philippe Normand, Rafał Dzięgiel, Ralf Sippl, Robert
+Mader, Sanchayan Maity, Sangchul Lee, Sebastian Dröge, Seungha Yang,
+Stéphane Cerveau, Teh Yule Kim, Thibault Saunier, Thomas Klausner, Timo
+Wischer, Tim-Philipp Müller, Tobias Reineke, Tomasz Andrzejak, Trung Do,
+Tyler Compton, Ung, Víctor Manuel Jáquez Leal, Vivia Nikolaidou, Wim
+Taymans, wngecn, Wonchul Lee, wuchang li, Xavier Claessens, Xi Ruoyao,
+Yoshiharu Hirose, Zhao,
 
 … and many others who have contributed bug reports, translations, sent
 suggestions or helped testing.
@@ -307,7 +1863,7 @@ the git 1.20 branch, which will be a stable branch.
 
 1.20.0
 
-1.20.0 is scheduled to be released around October/November 2021.
+1.20.0 is scheduled to be released around early February 2022.
 
 Schedule for 1.22
 
@@ -315,7 +1871,9 @@ Our next major feature release will be 1.22, and 1.21 will be the
 unstable development version leading up to the stable 1.22 release. The
 development of 1.21/1.22 will happen in the git main branch.
 
-The plan for the 1.22 development cycle is yet to be confirmed.
+The plan for the 1.22 development cycle is yet to be confirmed. Assuming
+no major project-wide reorganisations in the 1.22 cycle we might try and
+aim for a release around August 2022.
 
 1.22 will be backwards-compatible to the stable 1.20, 1.18, 1.16, 1.14,
 1.12, 1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series.
@@ -323,6 +1881,7 @@ The plan for the 1.22 development cycle is yet to be confirmed.
 ------------------------------------------------------------------------
 
 These release notes have been prepared by Tim-Philipp Müller with
-contributions from …
+contributions from Matthew Waters, Nicolas Dufresne, Nirbheek Chauhan,
+Sebastian Dröge and Seungha Yang.
 
 License: CC BY-SA 4.0
diff --git a/subprojects/gst-devtools/RELEASE b/subprojects/gst-devtools/RELEASE
index 997085726e..0ea325b2ef 100644
--- a/subprojects/gst-devtools/RELEASE
+++ b/subprojects/gst-devtools/RELEASE
@@ -1,4 +1,4 @@
-This is GStreamer gst-devtools 1.19.3.
+This is GStreamer gst-devtools 1.19.90.
 
 GStreamer 1.19 is the development branch leading up to the next major
 stable version which will be 1.20.
diff --git a/subprojects/gst-devtools/gst-devtools.doap b/subprojects/gst-devtools/gst-devtools.doap
index 1146614d25..77a35da09a 100644
--- a/subprojects/gst-devtools/gst-devtools.doap
+++ b/subprojects/gst-devtools/gst-devtools.doap
@@ -55,6 +55,16 @@
 
  <release>
   <Version>
+   <revision>1.19.90</revision>
+   <branch>main</branch>
+   <name></name>
+   <created>2022-01-28</created>
+   <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-devtools/gst-devtools-1.19.90.tar.xz" />
+  </Version>
+ </release>
+
+ <release>
+  <Version>
    <revision>1.19.3</revision>
    <branch>main</branch>
    <name></name>
diff --git a/subprojects/gst-devtools/meson.build b/subprojects/gst-devtools/meson.build
index 7d50af5d10..3146fa30e3 100644
--- a/subprojects/gst-devtools/meson.build
+++ b/subprojects/gst-devtools/meson.build
@@ -1,5 +1,5 @@
 project('gst-devtools', 'c',
-  version : '1.19.3.1',
+  version : '1.19.90',
   meson_version : '>= 0.59',
   default_options : [ 'warning_level=1',
                       'c_std=gnu99',
diff --git a/subprojects/gst-docs/meson.build b/subprojects/gst-docs/meson.build
index 95efff66bd..8e0584a2e5 100644
--- a/subprojects/gst-docs/meson.build
+++ b/subprojects/gst-docs/meson.build
@@ -1,5 +1,5 @@
 project('GStreamer manuals and tutorials', 'c',
-  version: '1.19.3.1',
+  version: '1.19.90',
   meson_version : '>= 0.59')
 
 hotdoc_p = find_program('hotdoc')
diff --git a/subprojects/gst-editing-services/ChangeLog b/subprojects/gst-editing-services/ChangeLog
index cc6448a729..ac62a0a6d9 100644
--- a/subprojects/gst-editing-services/ChangeLog
+++ b/subprojects/gst-editing-services/ChangeLog
@@ -1,3 +1,18 @@
+=== release 1.19.90 ===
+
+2022-01-28 14:28:35 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* NEWS:
+	* RELEASE:
+	* gst-editing-services.doap:
+	* meson.build:
+	  Release 1.19.90
+
+2022-01-28 14:28:28 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* ChangeLog:
+	  Update ChangeLogs for 1.19.90
+
 2022-01-24 19:44:32 +0200  Sebastian Dröge <sebastian@centricular.com>
 
 	* ges/meson.build:
diff --git a/subprojects/gst-editing-services/NEWS b/subprojects/gst-editing-services/NEWS
index e9f5227c15..1a512c1848 100644
--- a/subprojects/gst-editing-services/NEWS
+++ b/subprojects/gst-editing-services/NEWS
@@ -1,23 +1,19 @@
 GStreamer 1.20 Release Notes
 
 GStreamer 1.20 has not been released yet. It is scheduled for release in
-November 2021.
+late January / early February 2022.
 
 1.19.x is the unstable development version that is being developed in
-the git main branch and which will eventually result in 1.20, and 1.19.3
-is the current development release in that series
-
-It is expected that feature freeze will be in early October 2021,
-followed by one or two 1.19.9x pre-releases and the new 1.20 stable
-release around October/November 2021.
+the git main branch and which will eventually result in 1.20, and
+1.19.90 is the first release candidate in that series (1.20rc1).
 
 1.20 will be backwards-compatible to the stable 1.18, 1.16, 1.14, 1.12,
-1.10, 1.8, 1.6,, 1.4, 1.2 and 1.0 release series.
+1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series.
 
 See https://gstreamer.freedesktop.org/releases/1.20/ for the latest
 version of this document.
 
-Last updated: Monday 1 November 2021, 01:00 UTC (log)
+Last updated: Wednesday 26 January 2022, 01:00 UTC (log)
 
 Introduction
 
@@ -30,25 +26,971 @@ fixes and other improvements.
 
 Highlights
 
--   this section will be completed in due course
+-   Development in GitLab was switched to a single git repository
+    containing all the modules
+-   GstPlay: new high-level playback library, replaces GstPlayer
+-   WebM Alpha decoding support
+-   Encoding profiles can now be tweaked with additional
+    application-specified element properties
+-   Compositor: multi-threaded video conversion and mixing
+-   RTP header extensions: unified support in RTP depayloader and
+    payloader base classes
+-   SMPTE 2022-1 2-D Forward Error Correction support
+-   Smart encoding (passthrough) support for VP8, VP9, H.265 in
+    encodebin and transcodebin
+-   Runtime compatibility support for libsoup2 and libsoup3 (libsoup3
+    support experimental)
+-   Video decoder subframe support
+-   Video decoder automatic packet-loss, data corruption, and keyframe
+    request handling for RTP / WebRTC / RTSP
+-   MP4 and Matroska muxers now support profile/level/resolution changes
+    for H264/H265 input streams (i.e. codec data changing on the fly)
+-   MP4 muxing mode that initially creates a fragmented mp4 which is
+    converted to a regular mp4 on EOS
+-   Audio support for the WebKit Port for Embedded (WPE) web page source
+    element
+-   CUDA based video color space convert and rescale elements and
+    upload/download elements
+-   NVIDIA memory:NVMM support for OpenGL glupload and gldownload
+    elements
+-   Many WebRTC improvements
+-   The new VA-API plugin implemention fleshed out with more decoders
+    and new postproc elements
+-   AppSink API to retrieve events in addition to buffers and buffer
+    lists
+-   AppSrc gained more configuration options for the internal queue
+    (leakiness, limits in buffers and time, getters to read current
+    levels)
+-   Updated Rust bindings and many new Rust plugins
+-   Improved support for custom minimal GStreamer builds
+-   Support build against FFmpeg 5.0
+-   Linux Stateless CODEC support gained MPEG2 and VP9
+-   Windows Direct3D11/DXVA decoder gained AV1 and MPEG2 support
+-   Lots of new plugins, features, performance improvements and bug
+    fixes
 
 Major new features and changes
 
 Noteworthy new features and API
 
--   this section will be filled in in due course
+-   gst_element_get_request_pad() has been deprecated in favour of the
+    newly-added gst_element_request_pad_simple() which does the exact
+    same thing but has a less confusing name that hopefully makes clear
+    that the function request a new pad rather than just retrieves an
+    already-existing request pad.
+
+Development in GitLab was switched to a single git repository containing all the modules
+
+The GStreamer multimedia framework is a set of libraries and plugins
+split into a number of distinct modules which are released independently
+and which have so far been developed in separate git repositories in
+freedesktop.org GitLab.
+
+In addition to these separate git repositories there was a gst-build
+module that would use the Meson build systems’s subproject feature to
+download each individual module and then build everything in one go. It
+would also provide an uninstalled development environment that made it
+easy to work on GStreamer and use or test versions other than the
+system-installed GStreamer version.
+
+All of these modules have now (as of 28 September 2021) been merged into
+a single git repository (“Mono repository” or “monorepo”) which should
+simplify development workflows and continuous integration, especially
+where changes need to be made to multiple modules at once.
+
+This mono repository merge will primarily affect GStreamer developers
+and contributors and anyone who has workflows based on the GStreamer git
+repositories.
+
+The Rust bindings and Rust plugins modules have not been merged into the
+mono repository at this time because they follow a different release
+cycle.
+
+The mono repository lives in the existing GStreamer core git repository
+in GitLab in the new main branch and all future development will happen
+on this branch.
+
+Modules will continue to be released as separate tarballs.
+
+For more details, please see the GStreamer mono repository FAQ.
+
+GstPlay: new high-level playback library replacing GstPlayer
+
+-   GstPlay is a new high-level playback library that replaces the older
+    GstPlayer API. It is basically the same API as GstPlayer but
+    refactored to use bus messages for application notifications instead
+    of GObject signals. There is still a signal adapter object for those
+    who prefer signals. Since the existing GstPlayer API is already in
+    use in various applications, it didn’t seem like a good idea to
+    break it entirely. Instead a new API was added, and it is expected
+    that this new GstPlay API will be moved to gst-plugins-base in
+    future.
+
+-   The existing GstPlayer API is scheduled for deprecation and will be
+    removed at some point in the future (e.g. in GStreamer 1.24), so
+    application developers are urged to migrate to the new GstPlay API
+    at their earliest convenience.
+
+WebM alpha decoding
+
+-   Implement WebM alpha decoding (VP8/VP9 with alpha), which required
+    support and additions in various places. This is supported both with
+    software decoders and hardware-accelerated decoders.
+
+-   VP8/VP9 don’t support alpha components natively in the codec, so the
+    way this is implemented in WebM is by encoding the alpha plane with
+    transparency data as a separate VP8/VP9 stream. Inside the WebM
+    container (a variant of Matroska) this is coded as a single video
+    track with the “normal” VP8/VP9 video data making up the main video
+    data and each frame of video having an encoded alpha frame attached
+    to it as extra data ("BlockAdditional").
+
+-   matroskademux has been extended extract this per-frame alpha side
+    data and attach it in form of a GstVideoCodecAlphaMeta to the
+    regular video buffers. Note that this new meta is specific to this
+    VP8/VP9 alpha support and can’t be used to just add alpha support to
+    other codecs that don’t support it. Lastly, matroskademux also
+    advertises the fact that the streams contain alpha in the caps.
+
+-   The new codecalpha plugin contains various bits of infrastructure to
+    support autoplugging and debugging:
+
+    -   codecalphademux splits out the alpha stream from the metas on
+        the regular VP8/VP9 buffers
+    -   alphacombine takes two decoded raw video streams (one alpha, one
+        the regular video) and combines it into a video stream with
+        alpha
+    -   vp8alphadecodebin + vp9alphadecodebin are wrapper bins that use
+        the regular vp8dec and vp9dec software decoders to decode
+        regular and alpha streams and combine them again. To decodebin
+        these look like regular decoders which ju
+    -   The V4L2 CODEC plugin has stateless VP8/VP9 decoders that can
+        decode both alpha and non-alpha stream with a single decoder
+        instance
+
+-   A new AV12 video format was added which is basically NV12 with an
+    alpha plane, which is more convenient for many hardware-accelerated
+    decoders.
+
+-   Watch Nicolas Dufresne’s LCA 2022 talk “Bringing WebM Alpha support
+    to GStreamer” for all the details and a demo.
+
+RTP Header Extensions Base Class and Automatic Header Extension Handling in RTP Payloaders and Depayloaders
+
+-   RTP Header Extensions are specified in RFC 5285 and provide a way to
+    add small pieces of data to RTP packets in between the RTP header
+    and the RTP payload. This is often used for per-frame metadata,
+    extended timestamps or other application-specific extra data. There
+    are several commonly-used extensions specified in various RFCs, but
+    senders are free to put any kind of data in there, as long as sender
+    and receiver both know what that data is. Receivers that don’t know
+    about the header extensions will just skip the extra data without
+    ever looking at it. These header extensions can often be combined
+    with any kind of payload format, so may need to be supported by many
+    RTP payloader and depayloader elements.
+
+-   Inserting and extracting RTP header extension data has so far been a
+    bit inconvenient in GStreamer: There are functions to add and
+    retrieve RTP header extension data from RTP packets, but nothing
+    works automatically, even for common extensions. People would have
+    to do the insertion/extraction either in custom elements
+    before/after the RTP payloader/depayloader, or inside pad probes,
+    which isn’t very nice.
+
+-   This release adds various pieces of new infrastructure for generic
+    RTP header extension handling, as well as some implementations for
+    common extensions:
+
+    -   GstRTPHeaderExtension is a new helper base class for reading and
+        writing RTP header extensions. Nominally this subclasses
+        GstElement, but only so these extensions are stored in the
+        registry where they can be looked up by URI or name. They don’t
+        have pads and don’t get added to the pipeline graph as an
+        element.
+
+    -   "add-extension" and "clear-extension" action signals on RTP
+        payloaders and depayloaders for manual extension management
+
+    -   The "request-extension" signal will be emitted if an extension
+        is encountered that requires explicit mapping by the application
+
+    -   new "auto-header-extension" property on RTP payloaders and
+        depayloaders for automatic handling of known header extensions.
+        This is enabled by default. The extensions must be signalled via
+        caps / SDP.
+
+    -   RTP header extension implementations:
+
+        -   rtphdrextclientaudiolevel: Client-to-Mixer Audio Level
+            Indication (RFC 6464) (also see below)
+        -   rtphdrextcolorspace: Color Space extension, extends RTP
+            packets with color space and high dynamic range (HDR)
+            information
+        -   rtphdrexttwcc: Transport Wide Congestion Control support
+
+-   gst_rtp_buffer_remove_extension_data() is a new helper function to
+    remove an RTP header extension from an RTP buffer
+
+-   The existing gst_rtp_buffer_set_extension_data() now also supports
+    shrinking the extension data in size
+
+AppSink and AppSrc improvements
+
+-   appsink: new API to pull events out of appsink in addition to
+    buffers and buffer lists.
+
+    There was previously no way for users to receive incoming events
+    from appsink properly serialised with the data flow, even if they
+    are serialised events. The reason for that is that the only way to
+    intercept events was via a pad probe on the appsink sink pad, but
+    there is also internal queuing inside of appsink, so it’s difficult
+    to ascertain the right order of everything in all cases.
+
+    There is now a new "new-serialized-event" signal which will be
+    emitted when there’s a new event pending (just like the existing
+    "new-sample" signal). The "emit-signals" property must be set to
+    TRUE in order to activate this (but it’s also fine to just pull from
+    the application thread without using the signals).
+
+    gst_app_sink_pull_object() and gst_app_sink_try_pull_object() can be
+    used to pull out either an event or a new sample carrying a buffer
+    or buffer list, whatever is next in the queue.
+
+    EOS events will be filtered and will not be returned. EOS handling
+    can be done the ususal way, same as with _pull_sample().
+
+-   appsrc: allow configuration of internal queue limits in time and
+    buffers and add leaky mode.
+
+    There is internal queuing inside appsrc so the application thread
+    can push data into the element which will then be picked up by the
+    source element’s streaming thread and pushed into the pipeline from
+    that streaming thread. This queue is unlimited by default and until
+    now it was only possible to set a maximum size limit in bytes. When
+    that byte limit is reached, the pushing thread (application thread)
+    would be blocked until more space becomes available.
+
+    A limit in bytes is not particularly useful for many use cases, so
+    now it is possible to also configure limits in time and buffers
+    using the new "max-time" and "max-buffers" properties. Of course
+    there are also matching new read-only"current-level-buffers" and
+    "current-level-time properties" properties to query the current fill
+    level of the internal queue in time and buffers.
+
+    And as if that wasn’t enough the internal queue can also be
+    configured as leaky using the new "leaky-type" property. That way
+    when the queue is full the application thread won’t be blocked when
+    it tries to push in more data, but instead either the new buffer
+    will be dropped or the oldest data in the queue will be dropped.
+
+Better string serialization of nested GstCaps and GstStructures
+
+-   New string serialisation format for structs and caps that can handle
+    nested structs and caps properly by using brackets to delimit nested
+    items (e.g. some-struct, some-field=[nested-struct, nested=true]).
+    Unlike the default format the new variant can also support more than
+    one level of nesting. For backwards-compatibility reasons the old
+    format is still output by default when serialising caps and structs
+    using the existing API. The new functions gst_caps_serialize() and
+    gst_structure_serialize() can be used to output strings in the new
+    format.
+
+Convenience API for custom GstMetas
+
+-   New convenience API to register and create custom GstMetas:
+    gst_meta_register_custom() and gst_buffer_add_custom_meta(). Such
+    custom meta is backed by a GstStructure and does not require that
+    users of the API expose their GstMeta implementation as public API
+    for other components to make use of it. In addition, it provides a
+    simpler interface by ignoring the impl vs. api distinction that the
+    regular API exposes. This new API is meant to be the meta
+    counterpart to custom events and messages, and to be more convenient
+    than the lower-level API when the absolute best performance isn’t a
+    requirement. The reason it’s less performant than a “proper” meta is
+    that a proper meta is just a C struct in the end whereas this goes
+    through the GstStructure API which has a bit more overhead, which
+    for most scenarios is negligible however. This new API is useful for
+    experimentation or proprietary metas, but also has some limitations:
+    it can only be used if there’s a single producer of these metas;
+    it’s not allowed to register the same custom meta multiple times or
+    from multiple places.
+
+Additional Element Properties on Encoding Profiles
+
+-   GstEncodingProfile: The new "element-properties" and
+    gst_encoding_profile_set_element_properties() API allows
+    applications to set additional element properties on encoding
+    profiles to configure muxers and encoders. So far the encoding
+    profile template was the only place where this could be specified,
+    but often what applications want to do is take a ready-made encoding
+    profile shipped by GStreamer or the application and then tweak the
+    settings on top of that, which is now possible with this API. Since
+    applications can’t always know in advance what encoder element will
+    be used in the end, it’s even possible to specify properties on a
+    per-element basis.
+
+    Encoding Profiles are used in the encodebin, transcodebin and
+    camerabin elements and APIs to configure output formats (containers
+    and elementary streams).
+
+Audio Level Indication Meta for RFC 6464
+
+-   New GstAudioLevelMeta containing Audio Level Indication as per RFC
+    6464
+
+-   The level element has been updated to add GstAudioLevelMeta on
+    buffers if the "audio-level-meta" property is set to TRUE. This can
+    then in turn be picked up by RTP payloaders to signal the audio
+    level to receivers through RTP header extensions (see above).
+
+-   New Client-to-Mixer Audio Level Indication (RFC6464) RTP Header
+    Extension which should be automatically created and used by RTP
+    payloaders and depayloaders if their "auto-header-extension"
+    property is enabled and if the extension is part of the RTP caps.
+
+Automatic packet loss, data corruption and keyframe request handling for video decoders
+
+-   The GstVideoDecoder base class has gained various new APIs to
+    automatically handle packet loss and data corruption better by
+    default, especially in RTP, RTSP and WebRTC streaming scenarios, and
+    to give subclasses more control about how they want to handle
+    missing data:
+
+    -   Video decoder subclasses can mark output frames as corrupted via
+        the new GST_VIDEO_CODEC_FRAME_FLAG_CORRUPTED flag
+
+    -   A new "discard-corrupted-frames" property allows applications to
+        configure decoders so that corrupted frames are directly
+        discarded instead of being forwarded inside the pipeline. This
+        is a replacement for the "output-corrupt" property of the FFmpeg
+        decoders.
+
+    -   RTP depayloaders can now signal to decoders that data is missing
+        when sending GAP events for lost packets. GAP events can be sent
+        for various reason in a GStreamer pipeline. Often they are just
+        used to let downstream elements know that there isn’t a buffer
+        available at the moment, so downstream elements can move on
+        instead of waiting for one. They are also sent by RTP
+        depayloaders in the case that packets are missing, however, and
+        so far a decoder was not able to differentiate the two cases.
+        This has been remedied now: GAP events can be decorated with
+        gst_event_set_gap_flags() and GST_GAP_FLAG_MISSING_DATA to let
+        decoders now what happened, and decoders can then use that in
+        some cases to handle missing data better.
+
+    -   The GstVideoDecoder::handle_missing_data vfunc was added to
+        inform subclasses about packet loss or missing data and let them
+        handle it in their own way if they like.
+
+    -   gst_video_decoder_set_needs_sync_point() lets subclasses signal
+        that they need the stream to start with a sync point. If
+        enabled, the base class will discard all non-sync point frames
+        in the beginning and after a flush and does not pass them to the
+        subclass. Furthermore, if the first frame is not a sync point,
+        the base class will try and request a sync frame from upstream
+        by sending a force-key-unit event (see next items).
+
+    -   New "automatic-request-sync-points" and
+        "automatic-request-sync-point-flags" properties to automatically
+        request sync points when needed, e.g. on packet loss or if the
+        first frame is not a keyframe. Applications may want to enable
+        this on decoders operating in e.g. RTP/WebRTC/RTSP receiver
+        pipelines.
+
+    -   The new "min-force-key-unit-interval" property can be used to
+        ensure there’s a minimal interval between keyframe requests to
+        upstream (and/or the sender) and we’re not flooding the sender
+        with key unit requests.
+
+    -   gst_video_decoder_request_sync_point() allows subclasses to
+        request a new sync point (e.g. if they choose to do their own
+        missing data handling). This will still honour the
+        "min-force-key-unit-interval" property if set.
+
+Improved support for custom minimal GStreamer builds
+
+-   Element registration and registration of other plugin features
+    inside plugin init functions has been improved in order to
+    facilitate minimal custom GStreamer builds.
+
+-   A number of new macros have been added to declare and create
+    per-element and per-pluginfeature register functions in all plugins,
+    and then call those from the per-plugin plugin_init functions:
+
+    -   GST_ELEMENT_REGISTER_DEFINE,
+        GST_DEVICE_PROVIDER_REGISTER_DEFINE,
+        GST_DYNAMIC_TYPE_REGISTER_DEFINE, GST_TYPE_FIND_REGISTER_DEFINE
+        for the actual registration call with GStreamer
+    -   GST_ELEMENT_REGISTER, GST_DEVICE_PROVIDER_REGISTER,
+        GST_DYNAMIC_TYPE_REGISTER, GST_PLUGIN_STATIC_REGISTER,
+        GST_TYPE_FIND_REGISTER to call the registration function defined
+        by the REGISTER_DEFINE macro
+    -   GST_ELEMENT_REGISTER_DECLARE,
+        GST_DEVICE_PROVIDER_REGISTER_DECLARE,
+        GST_DYNAMIC_TYPE_REGISTER_DECLARE,
+        GST_TYPE_FIND_REGISTER_DECLARE to declare the registration
+        function defined by the REGISTER_DEFINE macro
+    -   and various variants for advanced use cases.
+
+-   This means that applications can call the per-element and
+    per-pluginfeature registration functions for only the elements they
+    need instead of registering plugins as a whole with all kinds of
+    elements that may not be required (e.g. encoder and decoder instead
+    of just decoder). In case of static linking all unused functions and
+    their dependencies would be removed in this case by the linker,
+    which helps minimise binary size for custom builds.
+
+-   gst_init() will automatically call a gst_init_static_plugins()
+    function if one exists.
+
+-   See the GStreamer static build documentation and Stéphane’s blog
+    post Generate a minimal GStreamer build, tailored to your needs for
+    more details.
 
 New elements
 
--   this section will be filled in in due course
+-   New aesdec and aesenc elements for AES encryption and decryption in
+    a custom format.
+
+-   New encodebin2 element with dynamic/sometimes source pads in order
+    to support the option of doing the muxing outside of encodebin,
+    e.g. in combination with a splitmuxsink.
+
+-   New fakeaudiosink and videocodectestsink elements for testing and
+    debugging (see below for more details)
+
+-   rtpisacpay, rtpisacdepay: new RTP payloader and depayloader for iSAC
+    audio codec
+
+-   rtpst2022-1-fecdec, rtpst2022-1-fecenc: new elements providing SMPTE
+    2022-1 2-D Forward Error Correction. More details in Mathieu’s blog
+    post.
+
+-   isac: new plugin wrapping the Internet Speech Audio Codec reference
+    encoder and decoder from the WebRTC project.
+
+-   asio: plugin for Steinberg ASIO (Audio Streaming Input/Output) API
+
+-   gssrc, gssink: add source and sink for Google Cloud Storage
+
+-   onnx: new plugin to apply ONNX neural network models to video
+
+-   openaptx: aptX and aptX-HD codecs using libopenaptx (v0.2.0)
+
+-   qroverlay, debugqroverlay: new elements that allows overlaying data
+    on top of video in form of a QR code
+
+-   cvtracker: new OpenCV-based tracker element
+
+-   av1parse, vp9parse: new parsers for AV1 and VP9 video
+
+-   va: work on the new VA-API plugin implementation for
+    hardware-accelerated video decoding and encoding has continued at
+    pace, with various new decoders and filters having joined the
+    initial vah264dec:
+
+    -   vah265dec: VA-API H.265 decoder
+    -   vavp8dec: VA-API VP8 decoder
+    -   vavp9dec: VA-API VP9 decoder
+    -   vaav1dec: VA-API AV1 decoder
+    -   vampeg2dec: VA-API MPEG-2 decoder
+    -   vadeinterlace: : VA-API deinterlace filter
+    -   vapostproc: : VA-API postproc filter (color conversion,
+        resizing, cropping, color balance, video rotation, skin tone
+        enhancement, denoise, sharpen)
+
+    See Víctor’s blog post “GstVA in GStreamer 1.20” for more details
+    and what’s coming up next.
+
+-   vaapiav1dec: new AV1 decoder element (in gstreamer-vaapi)
+
+-   msdkav1dec: hardware-accelerated AV1 decoder using the Intel Media
+    SDK / oneVPL
+
+-   nvcodec plugin for NVIDIA NVCODEC API for hardware-accelerated video
+    encoding and decoding:
+
+    -   cudaconvert, cudascale: new CUDA based video color space convert
+        and rescale elements
+    -   cudaupload, cudadownload: new helper elements for memory
+        transfer between CUDA and system memory spaces
+    -   nvvp8sldec, nvvp9sldec: new GstCodecs-based VP8/VP9 decoders
+
+-   Various new hardware-accelerated elements for Windows:
+
+    -   d3d11screencapturesrc: new desktop capture element, including a
+        GstDeviceProvider implementation to enumerate/select target
+        monitors for capture.
+    -   d3d11av1dec and d3d11mpeg2dec: AV1 and MPEG-2 decoders
+    -   d3d11deinterlace: deinterlacing filter
+    -   d3d11compositor: video composing element
+    -   see Windows section below for more details
+
+-   new Rust plugins:
+
+    -   audiornnoise: Removes noise from an audio stream
+    -   awstranscribeparse: Parses AWS audio transcripts into timed text
+        buffers
+    -   ccdetect: Detects if valid closed captions are present in a
+        closed captions stream
+    -   cea608tojson: Converts CEA-608 Closed Captions to a JSON
+        representation
+    -   cmafmux: CMAF fragmented MP4 muxer
+    -   dashmp4mux: DASH fragmented MP4 muxer
+    -   isofmp4mux: ISO fragmented MP4 muxer
+    -   ebur128level: EBU R128 Loudness Level Measurement
+    -   ffv1dec: FFV1 video decoder
+    -   gtk4paintablesink: GTK4 video sink, which provides a
+        GdkPaintable that can be rendered in various widgets
+    -   hlssink3: HTTP Live Streaming sink
+    -   hrtfrender: Head-Related Transfer Function (HRTF) renderer
+    -   hsvdetector: HSV colorspace detector
+    -   hsvfilter: HSV colorspace filter
+    -   jsongstenc: Wraps buffers containing any valid top-level JSON
+        structures into higher level JSON objects, and outputs those as
+        ndjson
+    -   jsongstparse: Parses ndjson as output by jsongstenc
+    -   jsontovtt: converts JSON to WebVTT subtitles
+    -   regex: Applies regular expression operations on text
+    -   roundedcorners: Adds rounded corners to video
+    -   spotifyaudiosrc: Spotify source
+    -   textahead: Display upcoming text buffers ahead (e.g. for
+        Karaoke)
+    -   transcriberbin: passthrough bin that transcribes raw audio to
+        closed captions using awstranscriber and puts the captions as
+        metas onto the video
+    -   tttojson: Converts timed text to a JSON representation
+    -   uriplaylistbin: Playlist source bin
+    -   webpdec-rs: WebP image decoder with animation support
+
+-   New plugin codecalpha with elements to assist with WebM Alpha
+    decoding
+
+    -   codecalphademux: Split stream with GstVideoCodecAlphaMeta into
+        two streams
+    -   alphacombine: Combine two raw video stream (I420 or NV12) as one
+        stream with alpha channel (A420 or AV12)
+    -   vp8alphadecodebin: A bin to handle software decoding of VP8 with
+        alpha
+    -   vp9alphadecodebin: A bin to handle software decoding of VP9 with
+        alpha
+
+-   New hardware accelerated elements for Linux:
+
+    -   v4l2slmpeg2dec: Support for Linux Stateless MPEG2 decoders
+    -   v4l2slvp9dec: Support for Linux Stateless VP9 decoders
+    -   v4l2slvp8alphadecodebin: Support HW accelerated VP8 with alpha
+        layer decoding
+    -   v4l2slvp9alphadecodebin: Support HW accelerated VP9 with alpha
+        layer decoding
 
 New element features and additions
 
--   this section will be filled in in due course
+-   assrender: handle more font mime types; better interaction with
+    matroskademux for embedded fonts
+
+-   audiobuffersplit: Add support for specifying output buffer size in
+    bytes (not just duration)
+
+-   audiolatency: new "samplesperbuffer" property so users can configure
+    the number of samples per buffer. The default value is 240 samples
+    which is equivalent to 5ms latency with a sample rate of 48000,
+    which might be larger than actual buffer size of audio capture
+    device.
+
+-   audiomixer, audiointerleave, GstAudioAggregator: now keep a count of
+    samples that are dropped or processed as statistic and can be made
+    to post QoS messages on the bus whenever samples are dropped by
+    setting the "qos-messages" property on input pads.
+
+-   audiomixer, compositor: improved handling of new inputs added at
+    runtime. New API was added to the GstAggregator base class to allow
+    subclasses to opt into an aggregation mode where inactive pads are
+    ignored when processing input buffers
+    (gst_aggregator_set_ignore_inactive_pads(),
+    gst_aggregator_pad_is_inactive()). An “inactive pad” in this context
+    is a pad which, in live mode, hasn’t yet received a first buffer,
+    but has been waited on at least once. What would happen usually in
+    this case is that the aggregator would wait for data on this pad
+    every time, up to the maximum configured latency. This would
+    inadvertently push mixer elements in live mode to the configured
+    latency envelope and delay processing when new inputs are added at
+    runtime until these inputs have actually produced data. This is
+    usually undesirable. With this new API, new inputs can be added
+    (requested) and configured and they won’t delay the data processing.
+    Applications can opt into this new behaviour by setting the
+    "ignore-inactive-pads" property on compositor, audiomixer or other
+    GstAudioAggregator-based elements.
+
+-   cccombiner: implement “scheduling” of captions. So far cccombiner’s
+    behaviour was essentially that of a funnel: it strictly looked at
+    input timestamps to associate together video and caption buffers.
+    Now it will try to smoothly schedule caption buffers in order to
+    have exactly one per output video buffer. This might involve
+    rewriting input captions, for example when the input is CDP then
+    sequence counters are rewritten, time codes are dropped and
+    potentially re-injected if the input video frame had a time code
+    meta. This can also lead to the input drifting from synchronization,
+    when there isn’t enough padding in the input stream to catch up. In
+    that case the element will start dropping old caption buffers once
+    the number of buffers in its internal queue reaches a certain limit
+    (configurable via the "max-scheduled" property). The new original
+    funnel-like behaviour can be restored by setting the "scheduling"
+    property to FALSE.
+
+-   ccconverter: new "cdp-mode" property to specify which sections to
+    include in CDP packets (timecode, CC data, service info). Various
+    software, including ffmpeg’s Decklink support, fails parsing CDP
+    packets that contain anything but CC data in the CDP packets.
+
+-   clocksync: new "sync-to-first" property for automatic timestamp
+    offset setup: if set clocksync will set up the "ts-offset" value
+    based on the first buffer and the pipeline’s running time when the
+    first buffer arrived. The newly configured "ts-offset" in this case
+    would be the value that allows outputting the first buffer without
+    waiting on the clock. This is useful for example to feed a non-live
+    input into an already-running pipeline.
+
+-   compositor:
+
+    -   multi-threaded input conversion and compositing. Set the
+        "max-threads" property to activate this.
+    -   new "sizing-policy" property to support display aspect ratio
+        (DAR)-aware scaling. By default the image is scaled to fill the
+        configured destination rectangle without padding and without
+        keeping the aspect ratio. With sizing-policy=keep-aspect-ratio
+        the input image is scaled to fit the destination rectangle
+        specified by GstCompositorPad:{xpos, ypos, width, height}
+        properties preserving the aspect ratio. As a result, the image
+        will be centered in the destination rectangle with padding if
+        necessary.
+    -   new "zero-size-is-unscaled" property on input pads. By default
+        pad width=0 or pad height=0 mean that the stream should not be
+        scaled in that dimension. But if the "zero-size-is-unscaled"
+        property is set to FALSE a width or height of 0 is instead
+        interpreted to mean that the input image on that pad should not
+        be composited, which is useful when creating animations where an
+        input image is made smaller and smaller until it disappears.
+    -   improved handling of new inputs at runtime via
+        "ignore-inactive-pads"property (see above for details)
+    -   allow output format with alpha even if none of the inputs have
+        alpha (also glvideomixer and other GstVideoAggregator
+        subclasses)
+
+-   dashsink: add h265 codec support and signals for allowing custom
+    playlist/fragment output
+
+-   decodebin3:
+
+    -   improved decoder selection, especially for hardware decoders
+    -   make input activation “atomic” when adding inputs dynamically
+    -   better interleave handling: take into account decoder latency
+        for interleave size
+
+-   decklink:
+
+    -   Updated DeckLink SDK to 11.2 to support DeckLink 8K Pro
+    -   decklinkvideosrc:
+        -   More accurate and stable capture timestamps: use the
+            hardware reference clock time when the frame was finished
+            being captured instead of a clock time much further down the
+            road.
+        -   Automatically detect widescreen vs. normal NTSC/PAL
+
+-   encodebin:
+
+    -   add “smart encoding” support for H.265, VP8 and VP9 (i.e. only
+        re-encode where needed and otherwise pass through encoded video
+        as-is).
+    -   H264/H265 smart encoding improvements: respect user-specified
+        stream-format, but if not specified default to avc3/hvc1 with
+        in-band SPS/PPS/VPS signalling for more flexibility.
+    -   new encodebin2 element with dynamic/sometimes source pads in
+        order to support the option of doing the muxing outside of
+        encodebin, e.g. in combination with splitmuxsink.
+    -   add APIs to set element properties on encoding profiles (see
+        below)
+
+-   errorignore: new "ignore-eos" property to also ignore FLOW_EOS from
+    downstream elements
+
+-   giosrc: add support for growing source files: applications can
+    specify that the underlying file being read is growing by setting
+    the "is-growing" property. If set, the source won’t EOS when it
+    reaches the end of the file, but will instead start monitoring it
+    and will start reading data again whenever a change is detected. The
+    new "waiting-data" and "done-waiting-data" signals keep the
+    application informed about the current state.
+
+-   gtksink, gtkglsink:
+
+    -   scroll event support: forwarded as navigation events into the
+        pipeline
+    -   "video-aspect-ratio-override" property to force a specific
+        aspect ratio
+    -   "rotate-method" property and support automatic rotation based on
+        image tags
+
+-   identity: new "stats" property allows applications to retrieve the
+    number of bytes and buffers that have passed through so far.
+
+-   interlace: add support for more formats, esp 10-bit, 12-bit and
+    16-bit ones
+
+-   jack: new "low-latency" property for automatic latency-optimized
+    setting and "port-names" property to select ports explicitly
+
+-   jpegdec: support output conversion to RGB using libjpeg-turbo (for
+    certain input files)
+
+-   line21dec:
+
+    -   "mode" property to control whether and how detected closed
+        captions should be inserted in the list of existing close
+        caption metas on the input frame (if any): add, drop, or
+        replace.
+    -   "ntsc-only" property to only look for captions if video has NTSC
+        resolution
+
+-   line21enc: new "remove-caption-meta" to remove metas from output
+    buffers after encoding the captions into the video data; support for
+    CDP closed captions
+
+-   matroskademux, matroskamux: Add support for ffv1, a lossless
+    intra-frame video coding format.
+
+-   matroskamux: accept in-band SPS/PPS/VPS for H264 and H265
+    (i.e. stream-format avc3 and hev1) which allows on-the-fly
+    profile/level/resolution changes.
+
+-   matroskamux: new "cluster-timestamp-offset" property, useful for use
+    cases where the container timestamps should map to some absolute
+    wall clock time, for example.
+
+-   rtpsrc: add "caps" property to allow explicit setting of the caps
+    where needed
+
+-   mpegts: support SCTE-35 passthrough via new "send-scte35-events"
+    property on MPEG-TS demuxer tsdemux. When enabled, SCTE 35 sections
+    (eg ad placement opportunities) are forwarded as events donwstream
+    where they can be picked up again by mpegtsmux. This required a
+    semantic change in the SCTE-35 section API: timestamps are now in
+    running time instead of muxer pts.
+
+-   tsdemux: Handle PCR-less MPEG-TS streams; more robust timestamp
+    handling in certain corner cases and for poorly muxed streams.
+
+-   mpegtsmux:
+
+    -   More conformance improvements to make MPEG-TS analyzers happy:
+        -   PCR timing accuracy: Improvements to the way mpegtsmux
+            outputs PCR observations in CBR mode, so that a PCR
+            observation is always inserted when needed, so that we never
+            miss the configured pcr-interval, as that triggers various
+            MPEG-TS analyser errors.
+        -   Improved PCR/SI scheduling
+    -   Don’t write PCR until PAT/PMT are output to make sure streams
+        start cleanly with a PAT/PMT.
+    -   Allow overriding the automatic PMT PID selection via
+        application-supplied PMT_%d fields in the prog-map
+        structure/property.
+
+-   mp4mux:
+
+    -   new "first-moov-then-finalise" mode for fragmented output where
+        the output will start with a self-contained moov atom for the
+        first fragment, and then produce regular fragments. Then at the
+        end when the file is finalised, the initial moov is invalidated
+        and a new moov is written covering the entire file. This way the
+        file is a “fragmented mp4” file while it is still being written
+        out, and remains playable at all times, but at the end it is
+        turned into a regular mp4 file (with former fragment headers
+        remaining as unused junk data in the file).
+    -   support H.264 avc3 and H.265 hvc1 stream formats as input where
+        the codec data is signalled in-band inside the bitstream instead
+        of caps/file headers.
+    -   support profile/level/resolution changes for H264/H265 input
+        streams (i.e. codec data changing on the fly). Each codec_data
+        is put into its own SampleTableEntry inside the stsd, unless the
+        input is in avc3 stream format in which case it’s written
+        in-band an not in the headers.
+
+-   multifilesink: new ""min-keyframe-distance"" property to make
+    minimum distance between keyframes in next-file=key-frame mode
+    configurable instead of hard-coding it to 10 seconds.
+
+-   mxfdemux has seen a big refactoring to support non-frame wrappings
+    and more accurate timestamp/seek handling for some formats
+
+-   msdk plugin for hardware-accelerated video encoding and decoding
+    using the Intel Media SDK:
+
+    -   oneVPL support (Intel oneAPI Video Processing Library)
+    -   AV1 decoding support
+    -   H.264 decoder now supports constrained-high and progressive-high
+        profiles
+    -   H.264 encoder:
+        -   more configuration options (properties):
+            "intra-refresh-type", "min-qp" , "max-qp", "p-pyramid",
+            "dblk-idc"
+    -   H.265 encoder:
+        -   can output main-still-picture profile
+        -   now inserts HDR SEIs (mastering display colour volume and
+            content light level)
+        -   more configuration options (properties):
+            "intra-refresh-type", "min-qp" , "max-qp", "p-pyramid",
+            "b-pyramid", "dblk-idc", "transform-skip"
+        -   support for RGB 10bit format
+    -   External bitrate control in encoders
+    -   Video post proc element msdkvpp gained support for 12-bit pixel
+        formats P012_LE, Y212_LE and Y412_LE
+
+-   nvh264sldec: interlaced stream support
+
+-   openh264enc: support main, high, constrained-high and
+    progressive-high profiles
+
+-   openjpeg: support for multithreaded decoding and encoding
+
+-   rtspsrc: now supports IPv6 also for tunneled mode (RTSP-over-HTTP);
+    new "ignore-x-server-reply" property to ignore the
+    x-server-ip-address server header reply in case of HTTP tunneling,
+    as it is often broken.
+
+-   souphttpsrc: Runtime compatibility support for libsoup2 and
+    libsoup3. libsoup3 is the latest major version of libsoup, but
+    libsoup2 and libsoup3 can’t co-exist in the same process because
+    there is no namespacing or versioning for GObject types. As a
+    result, it would be awkward if the GStreamer souphttpsrc plugin
+    linked to a specific version of libsoup, because it would only work
+    with applications that use the same version of libsoup. To make this
+    work, the soup plugin now tries to determine the libsoup version
+    used by the application (and its other dependencies) at runtime on
+    systems where GStreamer is linked dynamically. libsoup3 support is
+    still considered somewhat experimental at this point.
+
+-   srtsrc, srtsink: add signals for the application to accept/reject
+    incoming connections
+
+-   timeoverlay: new elapsed-running-time time mode which shows the
+    running time since the first running time (and each flush-stop).
+
+-   udpsrc: new timestamping mode to retrieve packet receive timestamps
+    from the kernel via socket control messages (SO_TIMESTAMPNS) on
+    supported platforms
+
+-   uritranscodebin: new setup-source and element-setup signals for
+    applications to configure elements used
+
+-   v4l2codecs plugin gained support for 4x4 and 32x32 tile formats
+    enabling some platforms or direct renders. Important memory usage
+    improvement.
+
+-   v4l2slh264dec now implements the final Linux uAPI as shipped on
+    Linux 5.11 and later.
+
+-   valve: add "drop-mode" property and provide two new modes of
+    operation: in drop-mode=forward-sticky-events sticky events
+    (stream-start, segment, tags, caps, etc.) are forwarded downstream
+    even when dropping is enabled; drop-mode=transform-to-gap will in
+    addition also convert buffers into gap events when dropping is
+    enabled, which lets downstream elements know that time is advancing
+    and might allow for preroll in many scenarios. By default all events
+    and all buffers are dropped when dropping is enabled, which can
+    cause problems with caps negotiation not progressing or branches not
+    prerolling when dropping is enabled.
+
+-   videocrop: support for many more pixel formats, e.g. planar YUV
+    formats with > 8bits and GBR* video formats; can now also accept
+    video not backed by system memory as long as downstream supports the
+    GstCropMeta
+
+-   videotestsrc: new smpte-rp-219 pattern for SMPTE75 RP-219 conformant
+    color bars
+
+-   vp8enc: finish support for temporal scalability: two new properties
+    ("temporal-scalability-layer-flags",
+    "temporal-scalability-layer-sync-flags") and a unit change on the
+    "temporal-scalability-target-bitrate" property (now expects bps);
+    also make temporal scalability details available to RTP payloaders
+    as buffer metadata.
+
+-   vp9enc: new properties to tweak encoder performance:
+
+    -   "aq-mode" to configure adaptive quantization modes
+    -   "frame-parallel-decoding" to configure whether to create a
+        bitstream that reduces decoding dependencies between frames
+        which allows staged parallel processing of more than one video
+        frames in the decoder. (Defaults to TRUE)
+    -   "row-mt", "tile-columns" and "tile-rows" so multithreading can
+        be enabled on a per-tile basis, instead of on a per tile-column
+        basis. In combination with the new "tile-rows" property, this
+        allows the encoder to make much better use of the available CPU
+        power.
+
+-   vp9dec, vp9enc: add support for 10-bit 4:2:0 and 4:2:2 YUV, as well
+    as 8-bit 4:4:4
+
+-   vp8enc, vp9enc now default to “good quality” for the deadline
+    property rather then “best quality”. Having the deadline set to best
+    quality causes the encoder to be absurdly slow, most real-life users
+    will prefer good-enough quality with better performance instead.
+
+-   wpesrc:
+
+    -   implement audio support: a new sometimes source pad will be
+        created for each audio stream created by the web engine.
+    -   move wpesrc to wpevideosrc and add a wrapper bin wpesrc to also
+        support audio
+    -   also handles web:// URIs now (same as cefsrc)
+    -   post messages with the estimated load progress on the bus
 
-Plugin and library moves
+-   x265enc: add negative DTS support, which means timestamps are now
+    offset by 1h same as with x264enc
+
+RTP Payloaders and Depayloaders
+
+-   rtpisacpay, rtpisacdepay: new RTP payloader and depayloader for iSAC
+    audio codec
+
+-   rtph264depay:
+
+    -   new "request-keyframe" property to make the depayloader
+        automatically request a new keyframe from the sender on packet
+        loss, consistent with the new property on rtpvp8depay.
+    -   new "wait-for-keyframe" property to make depayloader wait for a
+        new keyframe at the beginning and after packet loss (only
+        effective if the depayloader outputs AUs), consistent with the
+        existing property on rtpvp8depay.
+
+-   rtpopuspay, rtpopusdepay: support libwebrtc-compatible multichannel
+    audio in addition to the previously supported multichannel audio
+    modes
+
+-   rtpopuspay: add DTX (Discontinuous Transmission) support
+
+-   rtpvp8depay: new "request-keyframe" property to make the depayloader
+    automatically request a new keyframe from the sender on packet loss.
+
+-   rtpvp8pay: temporal scaling support
 
--   this section will be filled in in due course
+-   rtpvp9depay: Improved SVC handling (aggregate all layers)
+
+RTP Infrastructure
+
+-   rtpst2022-1-fecdec, rtpst2022-1-fecenc: new elements providing SMPTE
+    2022-1 2-D Forward Error Correction. More details in Mathieu’s blog
+    post.
+
+-   rtpreddec: BUNDLE support
+
+-   rtpredenc, rtpulpfecenc: add support for Transport-wide Congestion
+    Control (TWCC)
+
+-   rtpsession: new "twcc-feedback-interval" property to allow RTCP TWCC
+    reports to be scheduled on a timer instead of per marker-bit.
+
+Plugin and library moves
 
 -   There were no plugin moves or library moves in this cycle.
 
@@ -56,8 +998,6 @@ Plugin removals
 
 The following elements or plugins have been removed:
 
--   this section will be filled in in due course
-
 -   The ofa audio fingerprinting plugin has been removed. The MusicIP
     database has been defunct for years so this plugin is likely neither
     useful nor used by anyone.
@@ -71,135 +1011,575 @@ The following elements or plugins have been removed:
 
 Miscellaneous API additions
 
--   this section will be filled in in due course
+Core
+
+-   gst_buffer_new_memdup() is a convenience function for the
+    widely-used gst_buffer_new_wrapped(g_memdup(data,size),size)
+    pattern.
+
+-   gst_caps_features_new_single() creates a new single GstCapsFeatures,
+    avoiding the need to use the vararg function with NULL terminator
+    for simple cases.
+
+-   gst_element_type_set_skip_documentation() can be used by plugins to
+    signal that certain elements should not be included in the GStreamer
+    plugin documentation. This is useful for plugins where elements are
+    registered dynamically based on hardware capabilities and/or where
+    the available plugins and properties vary from system to system.
+    This is used in the d3d11 plugin for example to ensure that only the
+    list of default elements is advertised in the documentation.
+
+-   gst_type_find_suggest_empty_simple() is a new convenience function
+    for typefinders for cases where there’s only a media type and no
+    other fields.
+
+-   New API to create elements and set properties at construction time,
+    which is not only convenient, but also allows GStreamer elements to
+    have construct-only properties: gst_element_factory_make_full(),
+    gst_element_factory_make_valist(),
+    gst_element_factory_make_with_properties(),
+    gst_element_factory_create_full(),
+    gst_element_factory_create_valist(),
+    gst_element_factory_create_with_properties().
+
+-   GstSharedTaskPool: new “shared” task pool subclass with slightly
+    different default behaviour than the existing GstTaskPool which
+    would create unlimited number of threads for new tasks. The shared
+    taskpool creates up to N threads (default: 1) and then distributes
+    pending tasks to those threads round-robin style, and blocks if no
+    thread is available. It is possible to join tasks. This can be used
+    by plugins to implement simple multi-threaded processing and is used
+    for the new multi-threaded video conversion and compositing done in
+    GstVideoAggregator, videoconverter and compositor.
+
+Plugins Base Utils library
+
+-   GstDiscoverer:
+
+    -   gst_discoverer_container_info_get_tags() was added to retrieve
+        global/container tags (vs. per-stream tags). Per-Stream tags can
+        be retrieved via the existing
+        gst_discoverer_stream_info_get_tags().
+        gst_discoverer_info_get_tags(), which for many files returns a
+        confusing mix of stream and container tags, has been deprecated
+        in favour of the container/stream-specific functions.
+    -   gst_discoverer_stream_info_get_stream_number() returns a unique
+        integer identifier for a given stream within the given
+        GstDiscoverer context. (If this matches the stream number inside
+        the container bitstream that’s by coincidence and not by
+        design.)
+
+-   gst_pb_utils_get_caps_description_flags() can be used to query
+    whether certain caps represent a container, audio, video, image,
+    subtitles, tags, or something else. This only works for formats
+    known to GStreamer.
+
+-   gst_pb_utils_get_file_extension_from_caps() returns a possible file
+    extension for given caps.
+
+-   gst_codec_utils_h264_get_profile_flags_level(): Parses profile,
+    flags, and level from H264 AvcC codec_data. The format of H264 AVCC
+    extradata/sequence_header is documented in the ITU-T H.264
+    specification section 7.3.2.1.1 as well as in ISO/IEC 14496-15
+    section 5.3.3.1.2.
+
+-   gst_codec_utils_caps_get_mime_codec() to convert caps to a RFC 6381
+    compatible MIME codec string codec. Useful for providing the codecs
+    field inside the Content-Type HTTP header for containerized formats,
+    such as mp4 or matroska.
+
+GStreamer OpenGL integration library and plugins
+
+-   glcolorconvert: added suppport for converting the video formats
+    A420, AV12, BGR, BGRA, RGBP and BGRP.
+
+-   Added support to GstGLBuffer for persistent buffer mappings where a
+    Pixel Buffer Object (PBO) can be mapped by both the CPU and the GPU.
+    This removes a memcpy() when uploading textures or vertices
+    particularly when software decoders (e.g. libav) are direct
+    rendering into our memory. Improves transfer performance
+    significantly. Requires OpenGL 4.4, GL_ARB_buffer_storage or
+    GL_EXT_buffer_storage
+
+-   Added various helper functions for handling 4x4 matrices of affine
+    transformations as used by GstVideoAffineTransformationMeta.
+
+-   Add support to GstGLContext for allowing the application to control
+    the config (EGLConfig, GLXConfig, etc) used when creating the OpenGL
+    context. This allows the ability to choose between RGB16 or RGB10A2
+    or RGBA8 back/front buffer configurations that were previously
+    hardcoded. GstGLContext also supports retrieving the configuration
+    it was created with or from an externally provide OpenGL context
+    handle. This infrastructure is also used to create a compatible
+    config from an application/externally provided OpenGL context in
+    order to improve compatibility with other OpenGL frameworks and GUI
+    toolkits. A new environment variable GST_GL_CONFIG was also added to
+    be able to request a specific configuration from the command line.
+    Note: different platforms will have different functionality
+    available.
+
+-   Add support for choosing between EGL and WGL at runtime when running
+    on Windows. Previously this was a build-time switch. Allows use in
+    e.g. Gtk applications on Windows that target EGL/ANGLE without
+    recompiling GStreamer. gst_gl_display_new_with_type() can be used by
+    applications to choose a specific display type to use.
+
+-   Build fixes to explicitly check for Broadcom-specific libraries on
+    older versions of the Raspberry Pi platform. The Broadcom OpenGL ES
+    and EGL libraries have different filenames. Using the vc4 Mesa
+    driver on the Raspberry Pi is not affected.
+
+-   Added support to glupload and gldownload for transferring RGBA
+    buffers using the memory:NVMM available on the Nvidia Tegra family
+    of embedded devices.
+
+-   Added support for choosing libOpenGL and libGLX as used in a GLVND
+    environment on unix-based platforms. This allows using desktop
+    OpenGL and EGL without pulling in any GLX symbols as would be
+    required with libGL.
+
+Video library
+
+-   New raw video formats:
+
+    -   AV12 (NV12 with alpha plane)
+    -   RGBP and BGRP (planar RGB formats)
+    -   ARGB64 variants with specified endianness instead of host
+        endianness:
+        -   ARGB64_LE, ARGB64_BE
+        -   RGBA64_BE, RGBA64_LE
+        -   BGRA64_BE, BGRA64_LE
+        -   ABGR64_BE, ABGR64_LE
+
+-   gst_video_orientation_from_tag() is new convenience API to parse the
+    image orientation from a GstTagList.
+
+-   GstVideoDecoder subframe support (see below)
+
+-   GstVideoCodecState now also carries some HDR metadata
+
+-   Ancillary video data: implement transform functions for AFD/Bar
+    metas, so they will be forwarded in more cases
+
+MPEG-TS library
+
+This library only handles section parsing and such, see above for
+changes to the actual mpegtsmux and mpegtsdemux elements.
+
+-   many additions and improvements to SCTE-35 section parsing
+-   new API for fetching extended descriptors:
+    gst_mpegts_find_descriptor_with_extension()
+-   add support for SIT sections (Selection Information Tables)
+-   expose event-from-section constructor gst_event_new_mpegts_section()
+-   parse Audio Preselection Descriptor needed for Dolby AC-4
+
+GstWebRTC library + webrtcbin
+
+-   Change the way in which sink pads and transceivers are matched
+    together to support easier usage. If a pad is created without a
+    specific index (i.e. using sink_%u as the pad template), then an
+    available compatible transceiver will be searched for. If a specific
+    index is requested (i.e. sink_1) then if a transceiver for that
+    m-line already exists, that transceiver must match the new sink pad
+    request. If there is no transceiver available in either scenario, a
+    new transceiver is created. If a mixture of both sink_1 and sink_%u
+    requests result in an impossible situation, an error will be
+    produced at pad request time or from create offer/answer.
+
+-   webrtcbin now uses regular ICE nomination instead of libnice’s
+    default of aggressive ICE nomination. Regular ICE nomination is the
+    default recommended by various relevant standards and improves
+    connectivity in specific network scenarios.
+
+-   Add support for limiting the port range used for RTP with the
+    addition of the min-rtp-port and max-rtp-port properties on the ICE
+    object.
+
+-   Expose the SCTP transport as a property on webrtcbin to more closely
+    match the WebRTC specification.
+
+-   Added support for taking into account the data channel transport
+    state when determining the value of the "connection-state" property.
+    Previous versions of the WebRTC spec did not include the data
+    channel state when computing this value.
+
+-   Add configuration for choosing the size of the underlying sockets
+    used for transporting media data
+
+-   Always advertise support for the transport-cc RTCP feedback protocol
+    as rtpbin supports it. For full support, the configured caps (input
+    or through codec-preferences) need to include the relevant RTP
+    header extension.
+
+-   Numerous fixes to caps and media handling to fail-fast when an
+    incompatible situation is detected.
+
+-   Improved support for attaching the required media after a remote
+    offer has been set.
+
+-   Add support for dynamically changing the amount of FEC used for a
+    particular stream.
+
+-   webrtcbin now stops further SDP processing at the first error it
+    encounters.
+
+-   Completed support for either local or the remote closing a data
+    channel.
+
+-   Various fixes when performing BUNDLEing of the media streams in
+    relation to RTX and FEC usage.
+
+-   Add support for writing out QoS DSCP marking on outgoing packets to
+    improve reliability in some network scenarios.
+
+-   Improvements to the statistics returned by the get-stats signal
+    including the addition of the raw statistics from the internal
+    RTPSource, the TWCC stats when available.
+
+-   The webrtc library does not expose any objects anymore with public
+    fields. Instead properties have been added to replace that
+    functionality. If you are accessing such fields in your application,
+    switch to the corresponding properties.
+
+GstCodecs and Video Parsers
+
+-   Support for render delays to improve throughput across all CODECs
+    (used with NVDEC and V4L2).
+-   lots of improvements to parsers and the codec parsing decoder base
+    classes (H264, H265, VP8, VP9, AV1, MPEG-2) used for various
+    hardware-accelerate decoder APIs.
+
+Bindings support
+
+-   gst_allocation_params_new() allocates a GstAllocationParams struct
+    on the heap. This should only be used by bindings (and freed via
+    gst_allocation_params_free() then). In C code you would allocate
+    this on the stack and only init it in place.
+
+-   gst_debug_log_literal() can be used to log a string to the debug log
+    without going through any printf format expansion and associated
+    overhead. This is mostly useful for bindings such as the Rust
+    bindings which may have done their own formatting already .
+
+-   Provide non-inlined versions of refcounting APIs for various
+    GStreamer mini objects, so that they can be consumed by bindings
+    (e.g. gstreamer-sharp): gst_buffer_ref, gst_buffer_unref,
+    gst_clear_buffer, gst_buffer_copy, gst_buffer_replace,
+    gst_buffer_list_ref, gst_buffer_list_unref, gst_clear_buffer_list,
+    gst_buffer_list_copy, gst_buffer_list_replace, gst_buffer_list_take,
+    gst_caps_ref, gst_caps_unref, gst_clear_caps, gst_caps_replace,
+    gst_caps_take, gst_context_ref, gst_context_unref, gst_context_copy,
+    gst_context_replace, gst_event_replace, gst_event_steal,
+    gst_event_take, gst_event_ref, gst_event_unref, gst_clear_event,
+    gst_event_copy, gst_memory_ref, gst_memory_unref, gst_message_ref,
+    gst_message_unref, gst_clear_message, gst_message_copy,
+    gst_message_replace, gst_message_take, gst_promise_ref,
+    gst_promise_unref, gst_query_ref, gst_query_unref, gst_clear_query,
+    gst_query_copy, gst_query_replace, gst_query_take, gst_sample_ref,
+    gst_sample_unref, gst_sample_copy, gst_tag_list_ref,
+    gst_tag_list_unref, gst_clear_tag_list, gst_tag_list_replace,
+    gst_tag_list_take, gst_uri_copy, gst_uri_ref, gst_uri_unref,
+    gst_clear_uri.
+
+-   expose a GType for GstMiniObject
+
+-   gst_device_provider_probe() now returns non-floating device object
+
+API Deprecations
+
+-   gst_element_get_request_pad() has been deprecated in favour of the
+    newly-added gst_element_request_pad_simple() which does the exact
+    same thing but has a less confusing name that hopefully makes clear
+    that the function request a new pad rather than just retrieves an
+    already-existing request pad.
+
+-   gst_discoverer_info_get_tags(), which for many files returns a
+    confusing mix of stream and container tags, has been deprecated in
+    favour of the container-specific and stream-specific functions,
+    gst_discoverer_container_info_get_tags() and
+    gst_discoverer_stream_info_get_tags().
+
+-   gst_video_sink_center_rect() was deprecated in favour of the more
+    generic newly-added gst_video_center_rect().
+
+-   The GST_MEMORY_FLAG_NO_SHARE flag has been deprecated, as it tends
+    to cause problems and prevents sub-buffering. If pooling or lifetime
+    tracking is required, memories should be allocated through a custom
+    GstAllocator instead of relying on the lifetime of the buffers the
+    memories were originally attached to, which is fragile anyway.
+
+-   The GstPlayer high-level playback library is being replaced with the
+    new GstPlay library (see above). GstPlayer should be considered
+    deprecated at this point and will be marked as such in the next
+    development cycle. Applications should be ported to GstPlay.
+
+-   Gstreamer Editing Services: ges_video_transition_set_border(),
+    ges_video_transition_get_border()
+    ges_video_transition_set_inverted()
+    ges_video_transition_is_inverted() have been deprecated, use
+    ges_timeline_element_set_children_properties() instead.
 
 Miscellaneous performance, latency and memory optimisations
 
--   this section will be filled in in due course
+More video conversion fast paths
 
-Miscellaneous other changes and enhancements
+-   v210 ↔ I420, YV12, Y42B, UYVY and YUY2
+-   A420 → RGB
+
+Less jitter when waiting on the system clock
+
+-   Better system clock wait accuracy, less jitter: where available,
+    clock_nanosleep is used for higher accuracy for waits below 500
+    usecs, and waits below 2ms will first use the regular waiting system
+    and then clock_nanosleep for the remainder. The various wait
+    implementation have a latency ranging from 50 to 500+ microseconds.
+    While this is not a major issue when dealing with a low number of
+    waits per second (for ex: video), it does introduce a non-negligible
+    jitter for synchronisation of higher packet rate systems.
+
+Video decoder subframe support
 
--   this section will be filled in in due course
+-   The GstVideoDecoder base class gained API to process input at the
+    sub-frame level. That way video decoders can start decoding slices
+    before they have received the full input frame in its entirety (to
+    the extent this is supported by the codec, of course). This helps
+    with CPU utilisation and reduces latency.
 
-Tracing framework and debugging improvements
+-   This functionality is now being used in the OpenJPEG JPEG 2000
+    decoder, the FFmpeg H.264 decoder (in case of NAL-aligned input) and
+    the OpenMAX H.264/H.265 decoders (in case of NAL-aligned input).
 
--   this section will be filled in in due course
+Miscellaneous other changes and enhancements
+
+-   GstDeviceMonitor no longer fails to start just because one of the
+    device providers failed to start. That could happen for example on
+    systems where the pulseaudio device provider is installed, but
+    pulseaudio isn’t actually running but ALSA is used for audio
+    instead. In the same vein the device monitor now keeps track of
+    which providers have been started (via the new
+    gst_device_provider_is_started()) and only stops actually running
+    device providers when stopping the device monitor.
+
+-   On embedded systems it can be useful to create a registry that can
+    be shared and read by multiple processes running as different users.
+    It is now possible to set the new GST_REGISTRY_MODE environment
+    variable to specify the file mode for the registry file, which by
+    default is set to be only user readable/writable.
+
+-   GstNetClientClock will signal lost sync in case the remote time
+    resets (e.g. because device power cycles), by emitting the “synced”
+    signal with synced=FALSE parameter, so applications can take action.
+
+-   gst_value_deserialize_with_pspec() allows deserialization with a
+    hint for what the target GType should be. This allows for example
+    passing arrays of flags through the command line or
+    gst_util_set_object_arg(), eg: foo="<bar,bar+baz>".
+
+-   It’s now allowed to create an empty GstVideoOverlayComposition
+    without any rectangles by passing a NULL rectangle to
+    gst_video_overlay_composition_new(). This is useful for bindings and
+    simplifies application code in some places.
+
+Tracing framework, debugging and testing improvements
+
+-   New factories tracer to list loaded elements (and other plugin
+    features). This can be useful to collect a list of elements needed
+    for an application, which then in turn can be used to create a
+    tailored minimal GStreamer build that contains just the elements
+    needed and nothing else.
+-   New plugin-feature-loaded tracing hook for use by tracers like the
+    new factories tracer
+
+-   GstHarness: Add gst_harness_set_live() so that harnesses can be set
+    to non-live and return is-live=false in latency queries if needed.
+    Default behaviour is to always return is-live=true in latency
+    queries.
+
+-   navseek: new "hold-eos" property. When enabled, the element will
+    hold back an EOS event until the next keystroke (via navigation
+    events). This can be used to keep a video sink showing the last
+    frame of a video pipeline until a key is pressed instead of tearing
+    it down immediately on EOS.
+
+-   New fakeaudiosink element: mimics an audio sink and can be used for
+    testing and CI pipelines on systems where no audio system is
+    installed or running. It differs from fakesink in that it only
+    support audio caps and syncs to the clock by default like a normal
+    audio sink. It also implements the GstStreamVolume interface like
+    most audio sinks do.
+
+-   New videocodectestsink element for video codec conformance testing:
+    Calculates MD5 checksums for video frames and skips any padding
+    whilst doing so. Can optionally also write back the video data with
+    padding removed into a file for easy byte-by-byte comparison with
+    reference data.
 
 Tools
 
--   this section will be filled in in due course
+gst-inspect-1.0
+
+-   Can sort the list of plugins by passing --sort=name as command line
+    option
+
+gst-launch-1.0
+
+-   will now error out on top-level properties that don’t exist and
+    which were silently ignored before
+-   On Windows the high-resolution clock is enabled now, which provides
+    better clock and timer performance on Windows (see Windows section
+    below for more details).
+
+gst-play-1.0
+
+-   New --start-position command line argument to start playback from
+    the specified position
+-   Audio can be muted/unmuted in interactive mode by pressing the m
+    key.
+-   On Windows the high-resolution clock is enabled now (see Windows
+    section below for more details)
+
+gst-device-monitor-1.0
+
+-   New --include-hidden command line argument to also show “hidden”
+    device providers
+
+ges-launch-1.0
+
+-   New interactive mode that allows seeking and such. Can be disabled
+    by passing the --no-interactive argument on the command line.
+-   Option to forward tags
+-   Allow using an existing clip to determine the rendering format (both
+    topology and profile) via new --profile-from command line argument.
 
 GStreamer RTSP server
 
--   this section will be filled in in due course
+-   GstRTSPMediaFactory gained API to disable RTCP
+    (gst_rtsp_media_factory_set_enable_rtcp(), "enable-rtcp" property).
+    Previously RTCP was always allowed for all RTSP medias. With this
+    change it is possible to disable RTCP completely, no matter if the
+    client wants to do RTCP or not.
+
+-   Make a mount point of / work correctly. While not allowed by the
+    RTSP 2 spec, the RTSP 1 spec is silent on this and it is used in the
+    wild. It is now possible to use / as a mount path in
+    gst-rtsp-server, e.g. rtsp://example.com/ would work with this now.
+    Note that query/fragment parts of the URI are not necessarily
+    correctly handled, and behaviour will differ between various
+    client/server implementations; so use it if you must but don’t bug
+    us if it doesn’t work with third party clients as you’d hoped.
+
+-   multithreading fixes (races, refcounting issues, deadlocks)
+
+-   ONVIF audio backchannel fixes
+
+-   ONVIF trick mode optimisations
+
+-   rtspclientsink: new "update-sdp" signal that allows updating the SDP
+    before sending it to the server via ANNOUNCE. This can be used to
+    add additional metadata to the SDP, for example. The order and
+    number of medias must not be changed, however.
 
 GStreamer VAAPI
 
--   this section will be filled in in due course
+-   new AV1 decoder element (vaapiav1dec)
+
+-   H264 decoder: handle stereoscopic 3D video with frame packing
+    arrangement SEI messages
+
+-   H265 encoder: added Screen Content Coding extensions support
+
+-   H265 decoder: gained MAIN_444_12 profile support (decoded to
+    Y412_LE), and 4:2:2 12-bits support (decoded to Y212_LE)
+
+-   vaapipostproc: gained BT2020 color standard support
+
+-   vaapidecode: now generates caps templates dynamically at runtime in
+    order to advertise actually supported caps instead of all
+    theoretically supported caps.
+
+-   GST_VAAPI_DRM_DEVICE environment variable to force a specified DRM
+    device when a DRM display is used. It is ignored when other types of
+    displays are used. By default /dev/dri/renderD128 is used for DRM
+    display.
 
 GStreamer OMX
 
--   this section will be filled in in due course
+-   subframe support in H.264/H.265 decoders
 
 GStreamer Editing Services and NLE
 
--   this section will be filled in in due course
+-   framepositioner: new "operator" property to access blending modes in
+    the compositor
+-   timeline: Implement snapping to markers
+-   smart-mixer: Add support for d3d11compositor and glvideomixer
+-   titleclip: add "draw-shadow" child property
+-   ges:// URI support to define a timeline from a description.
+-   command-line-formatter
+    -   Add track management to timeline description
+    -   Add keyframe support
+-   ges-launch-1.0:
+    -   Add an interactive mode where we can seek etc…
+    -   Add option to forward tags
+    -   Allow using an existing clip to determine the rendering format
+        (both topology and profile) via new --profile-from command line
+        argument.
+-   Fix static build
 
 GStreamer validate
 
--   this section will be filled in in due course
+-   report: Add a way to force backtraces on reports even if not a
+    critical issue (GST_VALIDATE_ISSUE_FLAGS_FORCE_BACKTRACE)
+-   Add a flag to gst_validate_replace_variables_in_string() allow
+    defining how to resolve variables in structs
+-   Add gst_validate_bin_monitor_get_scenario() to get the bin monitor
+    scenario, which is useful for applications that use Validate
+    directly.
+-   Add an expected-values parameter to wait, message-type=XX allowing
+    more precise filtering of the message we are waiting for.
+-   Add config file support: each test can now use a config file for the
+    given media file used to test.
+-   Add support to check properties of object properties
+-   scenario: Add an "action-done" signal to signal when an action is
+    done
+-   scenario: Add a "run-command" action type
+-   scenario: Allow forcing running action on idle from scenario file
+-   scenario: Allow iterating over arrays in foreach
+-   scenario: Rename ‘interlaced’ action to ‘non-blocking’
+-   scenario: Add a non-blocking flag to the wait signal
 
 GStreamer Python Bindings
 
--   this section will be filled in in due course
+-   Fixes for Python 3.10
+-   Various build fixes
+-   at least one known breaking change caused by g-i annotation changes
+    (see below)
 
 GStreamer C# Bindings
 
--   this section will be filled in in due course
+-   Fix GstDebugGraphDetails enum
+-   Updated to latests GtkSharp
+-   Updated to include GStreamer 1.20 API
 
 GStreamer Rust Bindings and Rust Plugins
 
-The GStreamer Rust bindings are released separately with a different
-release cadence that’s tied to gtk-rs, but the latest release has
-already been updated for the upcoming new GStreamer 1.20 API.
-
-gst-plugins-rs, the module containing GStreamer plugins written in Rust,
-has also seen lots of activity with many new elements and plugins.
-
-What follows is a list of elements and plugins available in
-gst-plugins-rs, so people don’t miss out on all those potentially useful
-elements that have no C equivalent.
-
--   FIXME: add new elements
-
-Rust audio plugins
-
--   audiornnoise: New element for audio denoising which implements the
-    noise removal algorithm of the Xiph RNNoise library, in Rust
--   rsaudioecho: Port of the audioecho element from gst-plugins-good
-    rsaudioloudnorm: Live audio loudness normalization element based on
-    the FFmpeg af_loudnorm filter
--   claxondec: FLAC lossless audio codec decoder element based on the
-    pure-Rust claxon implementation
--   csoundfilter: Audio filter that can use any filter defined via the
-    Csound audio programming language
--   lewtondec: Vorbis audio decoder element based on the pure-Rust
-    lewton implementation
-
-Rust video plugins
-
--   cdgdec/cdgparse: Decoder and parser for the CD+G video codec based
-    on a pure-Rust CD+G implementation, used for example by karaoke CDs
--   cea608overlay: CEA-608 Closed Captions overlay element
--   cea608tott: CEA-608 Closed Captions to timed-text (e.g. VTT or SRT
-    subtitles) converter
--   tttocea608: CEA-608 Closed Captions from timed-text converter
--   mccenc/mccparse: MacCaption Closed Caption format encoder and parser
--   sccenc/sccparse: Scenarist Closed Caption format encoder and parser
--   dav1dec: AV1 video decoder based on the dav1d decoder implementation
-    by the VLC project
--   rav1enc: AV1 video encoder based on the fast and pure-Rust rav1e
-    encoder implementation
--   rsflvdemux: Alternative to the flvdemux FLV demuxer element from
-    gst-plugins-good, not feature-equivalent yet
--   rsgifenc/rspngenc: GIF/PNG encoder elements based on the pure-Rust
-    implementations by the image-rs project
-
-Rust text plugins
-
--   textwrap: Element for line-wrapping timed text (e.g. subtitles) for
-    better screen-fitting, including hyphenation support for some
-    languages
-
-Rust network plugins
-
--   reqwesthttpsrc: HTTP(S) source element based on the Rust
-    reqwest/hyper HTTP implementations and almost feature-equivalent
-    with the main GStreamer HTTP source souphttpsrc
--   s3src/s3sink: Source/sink element for the Amazon S3 cloud storage
--   awstranscriber: Live audio to timed text transcription element using
-    the Amazon AWS Transcribe API
-
-Generic Rust plugins
-
--   sodiumencrypter/sodiumdecrypter: Encryption/decryption element based
-    on libsodium/NaCl
--   togglerecord: Recording element that allows to pause/resume
-    recordings easily and considers keyframe boundaries
--   fallbackswitch/fallbacksrc: Elements for handling potentially
-    failing (network) sources, restarting them on errors/timeout and
-    showing a fallback stream instead
--   threadshare: Set of elements that provide alternatives for various
-    existing GStreamer elements but allow to share the streaming threads
-    between each other to reduce the number of threads
--   rsfilesrc/rsfilesink: File source/sink elements as replacements for
-    the existing filesrc/filesink elements
+-   The GStreamer Rust bindings are released separately with a different
+    release cadence that’s tied to gtk-rs, but the latest release has
+    already been updated for the upcoming new GStreamer 1.20 API (v1_20
+    feature).
+
+-   gst-plugins-rs, the module containing GStreamer plugins written in
+    Rust, has also seen lots of activity with many new elements and
+    plugins. See the New Elements section above for a list of new Rust
+    elements.
 
 Build and Dependencies
 
 -   Meson 0.59 or newer is required to build GStreamer now.
 
--   FIXME: this section will be filled in in due course
+-   The GLib requirement has been bumped to GLib 2.56 or newer (from
+    March 2018).
+
+-   The wpe plugin now requires wpe >= 2.28 and wpebackend-fdo >= 1.8
 
 Explicit opt-in required for build of certain plugins with (A)GPL dependencies
 
@@ -211,11 +1591,9 @@ even if the required dependencies are available.
 See Building plugins with (A)GPL-licensed dependencies for more details
 and a non-exhaustive list of plugins affected.
 
-gst-build: replaced by Monorepo
+gst-build: replaced by mono repository
 
--   this section will be filled in in due course
-
--   FIXME: describe + link to Monorepo FAQ
+See mono repository section above and the GStreamer mono repository FAQ.
 
 Cerbero
 
@@ -223,76 +1601,254 @@ Cerbero is a meta build system used to build GStreamer plus dependencies
 on platforms where dependencies are not readily available, such as
 Windows, Android, iOS and macOS.
 
-General improvements
+General Cerbero improvements
 
--   this section will be filled in in due course
+-   Plugin removed: libvisual
+-   New plugins: rtpmanagerbad and rist
 
-macOS / iOS
+macOS / iOS specific Cerbero improvements
 
--   this section will be filled in in due course
+-   XCode 12 support
+-   macOS OS release support is now future-proof, similar to iOS
+-   macOS Apple Silicon (ARM64) cross-compile support has been added
+-   macOS Apple Silicon (ARM64) native support is currently experimental
 
-Windows
+Windows specific Cerbero improvements
 
--   this section will be filled in in due course
+-   Visual Studio 2022 support has been added
+-   bootstrap is faster since it requires building fewer build-tools
+    recipes on Windows
+-   package is faster due to better scheduling of recipe stages and
+    elimination of unnecessary autotools regeneration
+-   The following plugins are no longer built on Windows:
+    -   a52dec (another decoder is still available in libav)
+    -   dvdread
+    -   resindvd
 
 Windows MSI installer
 
--   this section will be filled in in due course
+-   no major changes
 
-Linux
+Linux specific Cerbero improvements
 
--   this section will be filled in in due course
+-   Fedora, Debian OS release support is now more future-proof
+-   Amazon Linux 2 support has been added
 
-Android
+Android specific Cerbero improvements
 
--   this section will be filled in in due course
+-   no major changes
 
 Platform-specific changes and improvements
 
 Android
 
--   this section will be filled in in due course
+-   No major changes
 
 macOS and iOS
 
--   this section will be filled in in due course
+-   applemedia: add ProRes support to vtenc and vtdec
 
 Windows
 
--   this section will be filled in in due course
+-   On Windows the high-resolution clock is enabled now in the
+    gst-launch-1.0 and gst-play-1.0 command line tools, which provides
+    better clock and timer performance on Windows, at the cost of higher
+    power consumption. By default, without the high-resolution clock
+    enabled, the timer precision on Windows is system-dependent and may
+    be as bad as 15ms which is not good enough for many multimedia
+    applications. Developers may want to do the same in their Windows
+    applications if they think it’s a good idea for their application
+    use case, and depending on the Windows version they target. This is
+    not done automatically by GStreamer because on older Windows
+    versions (pre-Windows 10) this affects a global Windows setting and
+    also there’s a power consumption vs. performance trade-off that may
+    differ from application to application.
+
+-   dxgiscreencapsrc now supports resolution changes
+
+-   The wasapi2 audio plugin was rewritten and now has a higher rank
+    than the old wasapi plugin since it has a number of additional
+    features such as automatic stream routing, and no
+    known-but-hard-to-fix issues. The plugin is always built if the
+    Windows 10 SDK is available now.
+
+-   The wasapi device providers now detect and notify dynamic device
+    additions/removals
+
+-   d3d11screencapturesrc: new desktop capture element, including
+    GstDeviceProvider implementation to enumerate/select target monitors
+    for capture.
+
+-   Direct3D11/DXVA decoder now supports AV1 and MPEG2 codecs
+    (d3d11av1dec, d3d11mpeg2dec)
+
+-   VP9 decoding got more reliable and stable thanks to a newly written
+    codec parser
+
+-   Support for decoding interlaced H.264/AVC streams
+
+-   Hardware-accelerated video deinterlacing (d3d11deinterlace) and
+    video mixing (d3d11compositor)
+
+-   Video mixing with the Direct3D11 API (d3d11compositor)
+
+-   MediaFoundation API based hardware encoders gained the ability to
+    receive Direct3D11 textures as an input
+
+-   Seungha’s blog post “GStreamer ❤ Windows: A primer on the cool stuff
+    you’ll find in the 1.20 release” describes many of the
+    Windows-related improvements in more detail
 
 Linux
 
--   this section will be filled in in due course
+-   bluez: LDAC Bluetooth audio codec support in a2dpsink and avdtpsink,
+    as well as an LDAC RTP payloader (rtpldacpay) and an LDAC audio
+    encoder (ldacenc)
+
+-   kmssink: gained support for NV24, NV61, RGB16/BGR16 formats;
+    auto-detect NVIDIA Tegra driver
 
 Documentation improvements
 
--   this section will be filled in in due course
+-   hardware-accelerated GPU plugins will now no longer always list all
+    the element variants for all available GPUs, since those are
+    system-dependent and it’s confusing for users to see those in the
+    documentation just because the GStreamer developer who generated the
+    docs had multiple GPUs to play with at the time. Instead just show
+    the default elements.
 
-Possibly Breaking Changes
+Possibly Breaking and Other Noteworthy Behavioural Changes
+
+-   gst_parse_launch(), gst_parse_bin_from_description() and friends
+    will now error out when setting properties that don’t exist on
+    top-level bins. They were silently ignored before.
+
+-   The GstWebRTC library does not expose any objects anymore with
+    public fields. Instead properties have been added to replace that
+    functionality. If you are accessing such fields in your application,
+    switch to the corresponding properties.
 
--   FIXME: this section will be filled in in due course
--   MPEG-TS SCTE-35 API changes (FIXME: flesh out)
--   gst_parse_launch() and friends now error out on non-existing
-    properties on top-level bins where they would silently fail and
-    ignore those before.
 -   playbin and uridecodebin now emit the source-setup signal before the
     element is added to the bin and linked so that the source element is
     already configured before any scheduling query comes in, which is
-    useful for elements such as appsrc or giostreamsrc. (Merge Request)
+    useful for elements such as appsrc or giostreamsrc.
+
+-   The source element inside urisourcebin (used inside uridecodebin3
+    which is used inside playbin3) is no longer called "source". This
+    shouldn’t affect anyone hopefully, because there’s a "setup-source"
+    signal to configure the source element and no one should rely on
+    names of internal elements anyway.
+
+-   The vp8enc element now expects bps (bits per second) for the
+    "temporal-scalability-target-bitrate" property, which is consistent
+    with the "target-bitrate" property. Since additional configuration
+    is required with modern libvpx to make temporal scaling work anyway,
+    chances are that very few people will have been using this property
+
+-   vp8enc and vp9enc now default to “good quality” for the "deadline"
+    property rather then “best quality”. Having the deadline set to best
+    quality causes the encoder to be absurdly slow, most real-life users
+    will want the good quality tradeoff instead.
+
+-   The experimental GstTranscoder library API in gst-plugins-bad was
+    changed from a GObject signal-based notification mechanism to a
+    GstBus/message-based mechanism akin to GstPlayer/GstPlay.
+
+-   MPEG-TS SCTE-35 API: semantic change for SCTE-35 splice commands:
+    timestamps passed by the application should be in running time now,
+    since users of the API can’t really be expected to predict the local
+    PTS of the muxer.
+
+-   The GstContext used by souphttpsrc to share the session between
+    multiple element instances has changed. Previously it provided
+    direct access to the internal SoupSession object, now it only
+    provides access to an opaque, internal type. This change is
+    necessary because SoupSession is not thread-safe at all and can’t be
+    shared safely between arbitrary external code and souphttpsrc.
+
+-   Python bindings: GObject-introspection related Annotation fixes have
+    led to a case of a GstVideo.VideoInfo-related function signature
+    changing in the Python bindings (possibly one or two other cases
+    too). This is for a function that should never have been exposed in
+    the first place though, so the bindings are being updated to throw
+    an exception in that case, and the correct replacement API has been
+    added in form of an override.
 
 Known Issues
 
--   this section will be filled in in due course
-
--   There are a couple of known WebRTC-related regressions/blockers:
-
-    -   webrtc: DTLS setup with Chrome is broken
-    -   webrtcbin: First keyframe is usually lost
+-   nothing in particular at this point (but also see possibly breaking
+    changes section above)
 
 Contributors
 
--   this section will be filled in in due course
+Aaron Boxer, Adam Leppky, Adam Williamson, Alba Mendez, Alejandro
+González, Aleksandr Slobodeniuk, Alexander Vandenbulcke, Alex Ashley,
+Alicia Boya García, Andika Triwidada, Andoni Morales Alastruey, Andrew
+Wesie, Andrey Moiseev, Antonio Ospite, Antonio Rojas, Arthur Crippa
+Búrigo, Arun Raghavan, Ashley Brighthope, Axel Kellermann, Baek, Bastien
+Nocera, Bastien Reboulet, Benjamin Gaignard, Bing Song, Binh Truong,
+Biswapriyo Nath, Brad Hards, Brad Smith, Brady J. Garvin, Branko
+Subasic, Camilo Celis Guzman, Chris Bass, ChrisDuncanAnyvision, Chris
+White, Corentin Damman, Daniel Almeida, Daniel Knobe, Daniel Stone,
+david, David Fernandez, David Keijser, David Phung, Devarsh Thakkar,
+Dinesh Manajipet, Dmitry Samoylov, Dmitry Shusharin, Dominique Martinet,
+Doug Nazar, Ederson de Souza, Edward Hervey, Emmanuel Gil Peyrot,
+Enrique Ocaña González, Ezequiel Garcia, Fabian Orccon, Fabrice
+Fontaine, Fernando Jimenez Moreno, Florian Karydes, Francisco Javier
+Velázquez-García, François Laignel, Frederich Munch, Fredrik Pålsson,
+George Kiagiadakis, Georg Lippitsch, Göran Jönsson, Guido Günther,
+Guillaume Desmottes, Guiqin Zou, Haakon Sporsheim, Haelwenn (lanodan)
+Monnier, Haihao Xiang, Haihua Hu, Havard Graff, He Junyan, Helmut
+Januschka, Henry Wilkes, Hosang Lee, Hou Qi, Ignacio Casal Quinteiro,
+Igor Kovalenko, Ilya Kreymer, Imanol Fernandez, Jacek Tomaszewski, Jade
+Macho, Jakub Adam, Jakub Janků, Jan Alexander Steffens (heftig), Jan
+Schmidt, Jason Carrete, Jason Pereira, Jay Douglass, Jeongki Kim, Jérôme
+Laheurte, Jimmi Holst Christensen, Johan Sternerup, John Hassell, John
+Lindgren, John-Mark Bell, Jonathan Matthew, Jordan Petridis, Jose
+Quaresma, Julian Bouzas, Julien, Kai Uwe Broulik, Kasper Steensig
+Jensen, Kellermann Axel, Kevin Song, Khem Raj, Knut Inge Hvidsten, Knut
+Saastad, Kristofer Björkström, Lars Lundqvist, Lawrence Troup, Lim Siew
+Hoon, Lucas Stach, Ludvig Rappe, Luis Paulo Fernandes de Barros, Luke
+Yelavich, Mads Buvik Sandvei, Marc Leeman, Marco Felsch, Marek Vasut,
+Marian Cichy, Marijn Suijten, Marius Vlad, Markus Ebner, Mart Raudsepp,
+Matej Knopp, Mathieu Duponchelle, Matthew Waters, Matthieu De Beule,
+Mengkejiergeli Ba, Michael de Gans, Michael Olbrich, Michael Tretter,
+Michal Dzik, Miguel Paris, Mikhail Fludkov, mkba, Nazar Mokrynskyi,
+Nicholas Jackson, Nicola Murino, Nicolas Dufresne, Niklas Hambüchen,
+Nikolay Sivov, Nirbheek Chauhan, Olivier Blin, Olivier Crete, Olivier
+Crête, Paul Goulpié, Per Förlin, Peter Boba, P H, Philippe Normand,
+Philipp Zabel, Pieter Willem Jordaan, Piotrek Brzeziński, Rafał
+Dzięgiel, Rafostar, raghavendra, Raghavendra, Raju Babannavar, Raleigh
+Littles III, Randy Li, Randy Li (ayaka), Ratchanan Srirattanamet, Raul
+Tambre, reed.lawrence, Ricky Tang, Robert Rosengren, Robert Swain, Robin
+Burchell, Roman Sivriver, R S Nikhil Krishna, Ruben Gonzalez, Ruslan
+Khamidullin, Sanchayan Maity, Scott Moreau, Sebastian Dröge, Sergei
+Kovalev, Seungha Yang, Sid Sethupathi, sohwan.park, Sonny Piers, Staz M,
+Stefan Brüns, Stéphane Cerveau, Stephan Hesse, Stian Selnes, Stirling
+Westrup, Théo MAILLART, Thibault Saunier, Tim, Timo Wischer, Tim-Philipp
+Müller, Tim Schneider, Tobias Ronge, Tom Schoonjans, Tulio Beloqui,
+tyler-aicradle, U. Artie Eoff, Ung, Val Doroshchuk, VaL Doroshchuk,
+Víctor Manuel Jáquez Leal, Vivek R, Vivia Nikolaidou, Vivienne
+Watermeier, Vladimir Menshakov, Will Miller, Wim Taymans, Xabier
+Rodriguez Calvar, Xavier Claessens, Xℹ Ruoyao, Yacine Bandou, Yinhang
+Liu, youngh.lee, youngsoo.lee, yychao, Zebediah Figura, Zhang yuankun,
+Zhang Yuankun, Zhao, Zhao Zhili, , Aleksandar Topic, Antonio Ospite,
+Bastien Nocera, Benjamin Gaignard, Brad Hards, Carlos Falgueras García,
+Célestin Marot, Corentin Damman, Corentin Noël, Daniel Almeida, Daniel
+Knobe, Danny Smith, Dave Piché, Dmitry Osipenko, Fabrice Fontaine,
+fjmax, Florian Zwoch, Guillaume Desmottes, Haihua Hu, Heinrich Kruger,
+He Junyan, Jakub Adam, James Cowgill, Jan Alexander Steffens (heftig),
+Jean Felder, Jeongki Kim, Jiri Uncovsky, Joe Todd, Jordan Petridis,
+Krystian Wojtas, Marc-André Lureau, Marcin Kolny, Marc Leeman, Mark
+Nauwelaerts, Martin Reboredo, Mathieu Duponchelle, Matthew Waters,
+Mengkejiergeli Ba, Michael Gruner, Nicolas Dufresne, Nirbheek Chauhan,
+Olivier Crête, Philippe Normand, Rafał Dzięgiel, Ralf Sippl, Robert
+Mader, Sanchayan Maity, Sangchul Lee, Sebastian Dröge, Seungha Yang,
+Stéphane Cerveau, Teh Yule Kim, Thibault Saunier, Thomas Klausner, Timo
+Wischer, Tim-Philipp Müller, Tobias Reineke, Tomasz Andrzejak, Trung Do,
+Tyler Compton, Ung, Víctor Manuel Jáquez Leal, Vivia Nikolaidou, Wim
+Taymans, wngecn, Wonchul Lee, wuchang li, Xavier Claessens, Xi Ruoyao,
+Yoshiharu Hirose, Zhao,
 
 … and many others who have contributed bug reports, translations, sent
 suggestions or helped testing.
@@ -307,7 +1863,7 @@ the git 1.20 branch, which will be a stable branch.
 
 1.20.0
 
-1.20.0 is scheduled to be released around October/November 2021.
+1.20.0 is scheduled to be released around early February 2022.
 
 Schedule for 1.22
 
@@ -315,7 +1871,9 @@ Our next major feature release will be 1.22, and 1.21 will be the
 unstable development version leading up to the stable 1.22 release. The
 development of 1.21/1.22 will happen in the git main branch.
 
-The plan for the 1.22 development cycle is yet to be confirmed.
+The plan for the 1.22 development cycle is yet to be confirmed. Assuming
+no major project-wide reorganisations in the 1.22 cycle we might try and
+aim for a release around August 2022.
 
 1.22 will be backwards-compatible to the stable 1.20, 1.18, 1.16, 1.14,
 1.12, 1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series.
@@ -323,6 +1881,7 @@ The plan for the 1.22 development cycle is yet to be confirmed.
 ------------------------------------------------------------------------
 
 These release notes have been prepared by Tim-Philipp Müller with
-contributions from …
+contributions from Matthew Waters, Nicolas Dufresne, Nirbheek Chauhan,
+Sebastian Dröge and Seungha Yang.
 
 License: CC BY-SA 4.0
diff --git a/subprojects/gst-editing-services/RELEASE b/subprojects/gst-editing-services/RELEASE
index 2906b5637b..159e440e24 100644
--- a/subprojects/gst-editing-services/RELEASE
+++ b/subprojects/gst-editing-services/RELEASE
@@ -1,4 +1,4 @@
-This is GStreamer gst-editing-services 1.19.3.
+This is GStreamer gst-editing-services 1.19.90.
 
 GStreamer 1.19 is the development branch leading up to the next major
 stable version which will be 1.20.
diff --git a/subprojects/gst-editing-services/gst-editing-services.doap b/subprojects/gst-editing-services/gst-editing-services.doap
index 6e63cc12a8..d92ca09b92 100644
--- a/subprojects/gst-editing-services/gst-editing-services.doap
+++ b/subprojects/gst-editing-services/gst-editing-services.doap
@@ -32,6 +32,16 @@ GStreamer library for creating audio and video editors
 
  <release>
   <Version>
+   <revision>1.19.90</revision>
+   <branch>main</branch>
+   <name></name>
+   <created>2022-01-28</created>
+   <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-editing-services/gst-editing-services-1.19.90.tar.xz" />
+  </Version>
+ </release>
+
+ <release>
+  <Version>
    <revision>1.19.3</revision>
    <branch>main</branch>
    <name></name>
diff --git a/subprojects/gst-editing-services/meson.build b/subprojects/gst-editing-services/meson.build
index f2172f84bb..36b5a73506 100644
--- a/subprojects/gst-editing-services/meson.build
+++ b/subprojects/gst-editing-services/meson.build
@@ -1,5 +1,5 @@
 project('gst-editing-services', 'c',
-  version : '1.19.3.1',
+  version : '1.19.90',
   meson_version : '>= 0.59',
   default_options : [ 'warning_level=1',
                       'buildtype=debugoptimized' ])
diff --git a/subprojects/gst-examples/meson.build b/subprojects/gst-examples/meson.build
index faf80830af..ed6f3475fb 100644
--- a/subprojects/gst-examples/meson.build
+++ b/subprojects/gst-examples/meson.build
@@ -1,4 +1,4 @@
-project('gst-examples', 'c', version : '1.19.3.1', license : 'LGPL')
+project('gst-examples', 'c', version : '1.19.90', license : 'LGPL')
 
 cc = meson.get_compiler('c')
 m_dep = cc.find_library('m', required : false)
diff --git a/subprojects/gst-integration-testsuites/meson.build b/subprojects/gst-integration-testsuites/meson.build
index e694107f31..e20e21ee1d 100644
--- a/subprojects/gst-integration-testsuites/meson.build
+++ b/subprojects/gst-integration-testsuites/meson.build
@@ -1 +1 @@
-project('gst-integration-testsuites', [], version: '1.19.3.1', meson_version : '>= 0.59', license: 'LGPL')
+project('gst-integration-testsuites', [], version: '1.19.90', meson_version : '>= 0.59', license: 'LGPL')
diff --git a/subprojects/gst-libav/ChangeLog b/subprojects/gst-libav/ChangeLog
index 258c3a77c9..437c29ee8c 100644
--- a/subprojects/gst-libav/ChangeLog
+++ b/subprojects/gst-libav/ChangeLog
@@ -1,3 +1,18 @@
+=== release 1.19.90 ===
+
+2022-01-28 14:28:35 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* NEWS:
+	* RELEASE:
+	* gst-libav.doap:
+	* meson.build:
+	  Release 1.19.90
+
+2022-01-28 14:28:28 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* ChangeLog:
+	  Update ChangeLogs for 1.19.90
+
 2022-01-17 01:33:47 +0800  Xi Ruoyao <xry111@mengyan1223.wang>
 
 	* ext/libav/gstavauddec.c:
diff --git a/subprojects/gst-libav/NEWS b/subprojects/gst-libav/NEWS
index e9f5227c15..1a512c1848 100644
--- a/subprojects/gst-libav/NEWS
+++ b/subprojects/gst-libav/NEWS
@@ -1,23 +1,19 @@
 GStreamer 1.20 Release Notes
 
 GStreamer 1.20 has not been released yet. It is scheduled for release in
-November 2021.
+late January / early February 2022.
 
 1.19.x is the unstable development version that is being developed in
-the git main branch and which will eventually result in 1.20, and 1.19.3
-is the current development release in that series
-
-It is expected that feature freeze will be in early October 2021,
-followed by one or two 1.19.9x pre-releases and the new 1.20 stable
-release around October/November 2021.
+the git main branch and which will eventually result in 1.20, and
+1.19.90 is the first release candidate in that series (1.20rc1).
 
 1.20 will be backwards-compatible to the stable 1.18, 1.16, 1.14, 1.12,
-1.10, 1.8, 1.6,, 1.4, 1.2 and 1.0 release series.
+1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series.
 
 See https://gstreamer.freedesktop.org/releases/1.20/ for the latest
 version of this document.
 
-Last updated: Monday 1 November 2021, 01:00 UTC (log)
+Last updated: Wednesday 26 January 2022, 01:00 UTC (log)
 
 Introduction
 
@@ -30,25 +26,971 @@ fixes and other improvements.
 
 Highlights
 
--   this section will be completed in due course
+-   Development in GitLab was switched to a single git repository
+    containing all the modules
+-   GstPlay: new high-level playback library, replaces GstPlayer
+-   WebM Alpha decoding support
+-   Encoding profiles can now be tweaked with additional
+    application-specified element properties
+-   Compositor: multi-threaded video conversion and mixing
+-   RTP header extensions: unified support in RTP depayloader and
+    payloader base classes
+-   SMPTE 2022-1 2-D Forward Error Correction support
+-   Smart encoding (passthrough) support for VP8, VP9, H.265 in
+    encodebin and transcodebin
+-   Runtime compatibility support for libsoup2 and libsoup3 (libsoup3
+    support experimental)
+-   Video decoder subframe support
+-   Video decoder automatic packet-loss, data corruption, and keyframe
+    request handling for RTP / WebRTC / RTSP
+-   MP4 and Matroska muxers now support profile/level/resolution changes
+    for H264/H265 input streams (i.e. codec data changing on the fly)
+-   MP4 muxing mode that initially creates a fragmented mp4 which is
+    converted to a regular mp4 on EOS
+-   Audio support for the WebKit Port for Embedded (WPE) web page source
+    element
+-   CUDA based video color space convert and rescale elements and
+    upload/download elements
+-   NVIDIA memory:NVMM support for OpenGL glupload and gldownload
+    elements
+-   Many WebRTC improvements
+-   The new VA-API plugin implemention fleshed out with more decoders
+    and new postproc elements
+-   AppSink API to retrieve events in addition to buffers and buffer
+    lists
+-   AppSrc gained more configuration options for the internal queue
+    (leakiness, limits in buffers and time, getters to read current
+    levels)
+-   Updated Rust bindings and many new Rust plugins
+-   Improved support for custom minimal GStreamer builds
+-   Support build against FFmpeg 5.0
+-   Linux Stateless CODEC support gained MPEG2 and VP9
+-   Windows Direct3D11/DXVA decoder gained AV1 and MPEG2 support
+-   Lots of new plugins, features, performance improvements and bug
+    fixes
 
 Major new features and changes
 
 Noteworthy new features and API
 
--   this section will be filled in in due course
+-   gst_element_get_request_pad() has been deprecated in favour of the
+    newly-added gst_element_request_pad_simple() which does the exact
+    same thing but has a less confusing name that hopefully makes clear
+    that the function request a new pad rather than just retrieves an
+    already-existing request pad.
+
+Development in GitLab was switched to a single git repository containing all the modules
+
+The GStreamer multimedia framework is a set of libraries and plugins
+split into a number of distinct modules which are released independently
+and which have so far been developed in separate git repositories in
+freedesktop.org GitLab.
+
+In addition to these separate git repositories there was a gst-build
+module that would use the Meson build systems’s subproject feature to
+download each individual module and then build everything in one go. It
+would also provide an uninstalled development environment that made it
+easy to work on GStreamer and use or test versions other than the
+system-installed GStreamer version.
+
+All of these modules have now (as of 28 September 2021) been merged into
+a single git repository (“Mono repository” or “monorepo”) which should
+simplify development workflows and continuous integration, especially
+where changes need to be made to multiple modules at once.
+
+This mono repository merge will primarily affect GStreamer developers
+and contributors and anyone who has workflows based on the GStreamer git
+repositories.
+
+The Rust bindings and Rust plugins modules have not been merged into the
+mono repository at this time because they follow a different release
+cycle.
+
+The mono repository lives in the existing GStreamer core git repository
+in GitLab in the new main branch and all future development will happen
+on this branch.
+
+Modules will continue to be released as separate tarballs.
+
+For more details, please see the GStreamer mono repository FAQ.
+
+GstPlay: new high-level playback library replacing GstPlayer
+
+-   GstPlay is a new high-level playback library that replaces the older
+    GstPlayer API. It is basically the same API as GstPlayer but
+    refactored to use bus messages for application notifications instead
+    of GObject signals. There is still a signal adapter object for those
+    who prefer signals. Since the existing GstPlayer API is already in
+    use in various applications, it didn’t seem like a good idea to
+    break it entirely. Instead a new API was added, and it is expected
+    that this new GstPlay API will be moved to gst-plugins-base in
+    future.
+
+-   The existing GstPlayer API is scheduled for deprecation and will be
+    removed at some point in the future (e.g. in GStreamer 1.24), so
+    application developers are urged to migrate to the new GstPlay API
+    at their earliest convenience.
+
+WebM alpha decoding
+
+-   Implement WebM alpha decoding (VP8/VP9 with alpha), which required
+    support and additions in various places. This is supported both with
+    software decoders and hardware-accelerated decoders.
+
+-   VP8/VP9 don’t support alpha components natively in the codec, so the
+    way this is implemented in WebM is by encoding the alpha plane with
+    transparency data as a separate VP8/VP9 stream. Inside the WebM
+    container (a variant of Matroska) this is coded as a single video
+    track with the “normal” VP8/VP9 video data making up the main video
+    data and each frame of video having an encoded alpha frame attached
+    to it as extra data ("BlockAdditional").
+
+-   matroskademux has been extended extract this per-frame alpha side
+    data and attach it in form of a GstVideoCodecAlphaMeta to the
+    regular video buffers. Note that this new meta is specific to this
+    VP8/VP9 alpha support and can’t be used to just add alpha support to
+    other codecs that don’t support it. Lastly, matroskademux also
+    advertises the fact that the streams contain alpha in the caps.
+
+-   The new codecalpha plugin contains various bits of infrastructure to
+    support autoplugging and debugging:
+
+    -   codecalphademux splits out the alpha stream from the metas on
+        the regular VP8/VP9 buffers
+    -   alphacombine takes two decoded raw video streams (one alpha, one
+        the regular video) and combines it into a video stream with
+        alpha
+    -   vp8alphadecodebin + vp9alphadecodebin are wrapper bins that use
+        the regular vp8dec and vp9dec software decoders to decode
+        regular and alpha streams and combine them again. To decodebin
+        these look like regular decoders which ju
+    -   The V4L2 CODEC plugin has stateless VP8/VP9 decoders that can
+        decode both alpha and non-alpha stream with a single decoder
+        instance
+
+-   A new AV12 video format was added which is basically NV12 with an
+    alpha plane, which is more convenient for many hardware-accelerated
+    decoders.
+
+-   Watch Nicolas Dufresne’s LCA 2022 talk “Bringing WebM Alpha support
+    to GStreamer” for all the details and a demo.
+
+RTP Header Extensions Base Class and Automatic Header Extension Handling in RTP Payloaders and Depayloaders
+
+-   RTP Header Extensions are specified in RFC 5285 and provide a way to
+    add small pieces of data to RTP packets in between the RTP header
+    and the RTP payload. This is often used for per-frame metadata,
+    extended timestamps or other application-specific extra data. There
+    are several commonly-used extensions specified in various RFCs, but
+    senders are free to put any kind of data in there, as long as sender
+    and receiver both know what that data is. Receivers that don’t know
+    about the header extensions will just skip the extra data without
+    ever looking at it. These header extensions can often be combined
+    with any kind of payload format, so may need to be supported by many
+    RTP payloader and depayloader elements.
+
+-   Inserting and extracting RTP header extension data has so far been a
+    bit inconvenient in GStreamer: There are functions to add and
+    retrieve RTP header extension data from RTP packets, but nothing
+    works automatically, even for common extensions. People would have
+    to do the insertion/extraction either in custom elements
+    before/after the RTP payloader/depayloader, or inside pad probes,
+    which isn’t very nice.
+
+-   This release adds various pieces of new infrastructure for generic
+    RTP header extension handling, as well as some implementations for
+    common extensions:
+
+    -   GstRTPHeaderExtension is a new helper base class for reading and
+        writing RTP header extensions. Nominally this subclasses
+        GstElement, but only so these extensions are stored in the
+        registry where they can be looked up by URI or name. They don’t
+        have pads and don’t get added to the pipeline graph as an
+        element.
+
+    -   "add-extension" and "clear-extension" action signals on RTP
+        payloaders and depayloaders for manual extension management
+
+    -   The "request-extension" signal will be emitted if an extension
+        is encountered that requires explicit mapping by the application
+
+    -   new "auto-header-extension" property on RTP payloaders and
+        depayloaders for automatic handling of known header extensions.
+        This is enabled by default. The extensions must be signalled via
+        caps / SDP.
+
+    -   RTP header extension implementations:
+
+        -   rtphdrextclientaudiolevel: Client-to-Mixer Audio Level
+            Indication (RFC 6464) (also see below)
+        -   rtphdrextcolorspace: Color Space extension, extends RTP
+            packets with color space and high dynamic range (HDR)
+            information
+        -   rtphdrexttwcc: Transport Wide Congestion Control support
+
+-   gst_rtp_buffer_remove_extension_data() is a new helper function to
+    remove an RTP header extension from an RTP buffer
+
+-   The existing gst_rtp_buffer_set_extension_data() now also supports
+    shrinking the extension data in size
+
+AppSink and AppSrc improvements
+
+-   appsink: new API to pull events out of appsink in addition to
+    buffers and buffer lists.
+
+    There was previously no way for users to receive incoming events
+    from appsink properly serialised with the data flow, even if they
+    are serialised events. The reason for that is that the only way to
+    intercept events was via a pad probe on the appsink sink pad, but
+    there is also internal queuing inside of appsink, so it’s difficult
+    to ascertain the right order of everything in all cases.
+
+    There is now a new "new-serialized-event" signal which will be
+    emitted when there’s a new event pending (just like the existing
+    "new-sample" signal). The "emit-signals" property must be set to
+    TRUE in order to activate this (but it’s also fine to just pull from
+    the application thread without using the signals).
+
+    gst_app_sink_pull_object() and gst_app_sink_try_pull_object() can be
+    used to pull out either an event or a new sample carrying a buffer
+    or buffer list, whatever is next in the queue.
+
+    EOS events will be filtered and will not be returned. EOS handling
+    can be done the ususal way, same as with _pull_sample().
+
+-   appsrc: allow configuration of internal queue limits in time and
+    buffers and add leaky mode.
+
+    There is internal queuing inside appsrc so the application thread
+    can push data into the element which will then be picked up by the
+    source element’s streaming thread and pushed into the pipeline from
+    that streaming thread. This queue is unlimited by default and until
+    now it was only possible to set a maximum size limit in bytes. When
+    that byte limit is reached, the pushing thread (application thread)
+    would be blocked until more space becomes available.
+
+    A limit in bytes is not particularly useful for many use cases, so
+    now it is possible to also configure limits in time and buffers
+    using the new "max-time" and "max-buffers" properties. Of course
+    there are also matching new read-only"current-level-buffers" and
+    "current-level-time properties" properties to query the current fill
+    level of the internal queue in time and buffers.
+
+    And as if that wasn’t enough the internal queue can also be
+    configured as leaky using the new "leaky-type" property. That way
+    when the queue is full the application thread won’t be blocked when
+    it tries to push in more data, but instead either the new buffer
+    will be dropped or the oldest data in the queue will be dropped.
+
+Better string serialization of nested GstCaps and GstStructures
+
+-   New string serialisation format for structs and caps that can handle
+    nested structs and caps properly by using brackets to delimit nested
+    items (e.g. some-struct, some-field=[nested-struct, nested=true]).
+    Unlike the default format the new variant can also support more than
+    one level of nesting. For backwards-compatibility reasons the old
+    format is still output by default when serialising caps and structs
+    using the existing API. The new functions gst_caps_serialize() and
+    gst_structure_serialize() can be used to output strings in the new
+    format.
+
+Convenience API for custom GstMetas
+
+-   New convenience API to register and create custom GstMetas:
+    gst_meta_register_custom() and gst_buffer_add_custom_meta(). Such
+    custom meta is backed by a GstStructure and does not require that
+    users of the API expose their GstMeta implementation as public API
+    for other components to make use of it. In addition, it provides a
+    simpler interface by ignoring the impl vs. api distinction that the
+    regular API exposes. This new API is meant to be the meta
+    counterpart to custom events and messages, and to be more convenient
+    than the lower-level API when the absolute best performance isn’t a
+    requirement. The reason it’s less performant than a “proper” meta is
+    that a proper meta is just a C struct in the end whereas this goes
+    through the GstStructure API which has a bit more overhead, which
+    for most scenarios is negligible however. This new API is useful for
+    experimentation or proprietary metas, but also has some limitations:
+    it can only be used if there’s a single producer of these metas;
+    it’s not allowed to register the same custom meta multiple times or
+    from multiple places.
+
+Additional Element Properties on Encoding Profiles
+
+-   GstEncodingProfile: The new "element-properties" and
+    gst_encoding_profile_set_element_properties() API allows
+    applications to set additional element properties on encoding
+    profiles to configure muxers and encoders. So far the encoding
+    profile template was the only place where this could be specified,
+    but often what applications want to do is take a ready-made encoding
+    profile shipped by GStreamer or the application and then tweak the
+    settings on top of that, which is now possible with this API. Since
+    applications can’t always know in advance what encoder element will
+    be used in the end, it’s even possible to specify properties on a
+    per-element basis.
+
+    Encoding Profiles are used in the encodebin, transcodebin and
+    camerabin elements and APIs to configure output formats (containers
+    and elementary streams).
+
+Audio Level Indication Meta for RFC 6464
+
+-   New GstAudioLevelMeta containing Audio Level Indication as per RFC
+    6464
+
+-   The level element has been updated to add GstAudioLevelMeta on
+    buffers if the "audio-level-meta" property is set to TRUE. This can
+    then in turn be picked up by RTP payloaders to signal the audio
+    level to receivers through RTP header extensions (see above).
+
+-   New Client-to-Mixer Audio Level Indication (RFC6464) RTP Header
+    Extension which should be automatically created and used by RTP
+    payloaders and depayloaders if their "auto-header-extension"
+    property is enabled and if the extension is part of the RTP caps.
+
+Automatic packet loss, data corruption and keyframe request handling for video decoders
+
+-   The GstVideoDecoder base class has gained various new APIs to
+    automatically handle packet loss and data corruption better by
+    default, especially in RTP, RTSP and WebRTC streaming scenarios, and
+    to give subclasses more control about how they want to handle
+    missing data:
+
+    -   Video decoder subclasses can mark output frames as corrupted via
+        the new GST_VIDEO_CODEC_FRAME_FLAG_CORRUPTED flag
+
+    -   A new "discard-corrupted-frames" property allows applications to
+        configure decoders so that corrupted frames are directly
+        discarded instead of being forwarded inside the pipeline. This
+        is a replacement for the "output-corrupt" property of the FFmpeg
+        decoders.
+
+    -   RTP depayloaders can now signal to decoders that data is missing
+        when sending GAP events for lost packets. GAP events can be sent
+        for various reason in a GStreamer pipeline. Often they are just
+        used to let downstream elements know that there isn’t a buffer
+        available at the moment, so downstream elements can move on
+        instead of waiting for one. They are also sent by RTP
+        depayloaders in the case that packets are missing, however, and
+        so far a decoder was not able to differentiate the two cases.
+        This has been remedied now: GAP events can be decorated with
+        gst_event_set_gap_flags() and GST_GAP_FLAG_MISSING_DATA to let
+        decoders now what happened, and decoders can then use that in
+        some cases to handle missing data better.
+
+    -   The GstVideoDecoder::handle_missing_data vfunc was added to
+        inform subclasses about packet loss or missing data and let them
+        handle it in their own way if they like.
+
+    -   gst_video_decoder_set_needs_sync_point() lets subclasses signal
+        that they need the stream to start with a sync point. If
+        enabled, the base class will discard all non-sync point frames
+        in the beginning and after a flush and does not pass them to the
+        subclass. Furthermore, if the first frame is not a sync point,
+        the base class will try and request a sync frame from upstream
+        by sending a force-key-unit event (see next items).
+
+    -   New "automatic-request-sync-points" and
+        "automatic-request-sync-point-flags" properties to automatically
+        request sync points when needed, e.g. on packet loss or if the
+        first frame is not a keyframe. Applications may want to enable
+        this on decoders operating in e.g. RTP/WebRTC/RTSP receiver
+        pipelines.
+
+    -   The new "min-force-key-unit-interval" property can be used to
+        ensure there’s a minimal interval between keyframe requests to
+        upstream (and/or the sender) and we’re not flooding the sender
+        with key unit requests.
+
+    -   gst_video_decoder_request_sync_point() allows subclasses to
+        request a new sync point (e.g. if they choose to do their own
+        missing data handling). This will still honour the
+        "min-force-key-unit-interval" property if set.
+
+Improved support for custom minimal GStreamer builds
+
+-   Element registration and registration of other plugin features
+    inside plugin init functions has been improved in order to
+    facilitate minimal custom GStreamer builds.
+
+-   A number of new macros have been added to declare and create
+    per-element and per-pluginfeature register functions in all plugins,
+    and then call those from the per-plugin plugin_init functions:
+
+    -   GST_ELEMENT_REGISTER_DEFINE,
+        GST_DEVICE_PROVIDER_REGISTER_DEFINE,
+        GST_DYNAMIC_TYPE_REGISTER_DEFINE, GST_TYPE_FIND_REGISTER_DEFINE
+        for the actual registration call with GStreamer
+    -   GST_ELEMENT_REGISTER, GST_DEVICE_PROVIDER_REGISTER,
+        GST_DYNAMIC_TYPE_REGISTER, GST_PLUGIN_STATIC_REGISTER,
+        GST_TYPE_FIND_REGISTER to call the registration function defined
+        by the REGISTER_DEFINE macro
+    -   GST_ELEMENT_REGISTER_DECLARE,
+        GST_DEVICE_PROVIDER_REGISTER_DECLARE,
+        GST_DYNAMIC_TYPE_REGISTER_DECLARE,
+        GST_TYPE_FIND_REGISTER_DECLARE to declare the registration
+        function defined by the REGISTER_DEFINE macro
+    -   and various variants for advanced use cases.
+
+-   This means that applications can call the per-element and
+    per-pluginfeature registration functions for only the elements they
+    need instead of registering plugins as a whole with all kinds of
+    elements that may not be required (e.g. encoder and decoder instead
+    of just decoder). In case of static linking all unused functions and
+    their dependencies would be removed in this case by the linker,
+    which helps minimise binary size for custom builds.
+
+-   gst_init() will automatically call a gst_init_static_plugins()
+    function if one exists.
+
+-   See the GStreamer static build documentation and Stéphane’s blog
+    post Generate a minimal GStreamer build, tailored to your needs for
+    more details.
 
 New elements
 
--   this section will be filled in in due course
+-   New aesdec and aesenc elements for AES encryption and decryption in
+    a custom format.
+
+-   New encodebin2 element with dynamic/sometimes source pads in order
+    to support the option of doing the muxing outside of encodebin,
+    e.g. in combination with a splitmuxsink.
+
+-   New fakeaudiosink and videocodectestsink elements for testing and
+    debugging (see below for more details)
+
+-   rtpisacpay, rtpisacdepay: new RTP payloader and depayloader for iSAC
+    audio codec
+
+-   rtpst2022-1-fecdec, rtpst2022-1-fecenc: new elements providing SMPTE
+    2022-1 2-D Forward Error Correction. More details in Mathieu’s blog
+    post.
+
+-   isac: new plugin wrapping the Internet Speech Audio Codec reference
+    encoder and decoder from the WebRTC project.
+
+-   asio: plugin for Steinberg ASIO (Audio Streaming Input/Output) API
+
+-   gssrc, gssink: add source and sink for Google Cloud Storage
+
+-   onnx: new plugin to apply ONNX neural network models to video
+
+-   openaptx: aptX and aptX-HD codecs using libopenaptx (v0.2.0)
+
+-   qroverlay, debugqroverlay: new elements that allows overlaying data
+    on top of video in form of a QR code
+
+-   cvtracker: new OpenCV-based tracker element
+
+-   av1parse, vp9parse: new parsers for AV1 and VP9 video
+
+-   va: work on the new VA-API plugin implementation for
+    hardware-accelerated video decoding and encoding has continued at
+    pace, with various new decoders and filters having joined the
+    initial vah264dec:
+
+    -   vah265dec: VA-API H.265 decoder
+    -   vavp8dec: VA-API VP8 decoder
+    -   vavp9dec: VA-API VP9 decoder
+    -   vaav1dec: VA-API AV1 decoder
+    -   vampeg2dec: VA-API MPEG-2 decoder
+    -   vadeinterlace: : VA-API deinterlace filter
+    -   vapostproc: : VA-API postproc filter (color conversion,
+        resizing, cropping, color balance, video rotation, skin tone
+        enhancement, denoise, sharpen)
+
+    See Víctor’s blog post “GstVA in GStreamer 1.20” for more details
+    and what’s coming up next.
+
+-   vaapiav1dec: new AV1 decoder element (in gstreamer-vaapi)
+
+-   msdkav1dec: hardware-accelerated AV1 decoder using the Intel Media
+    SDK / oneVPL
+
+-   nvcodec plugin for NVIDIA NVCODEC API for hardware-accelerated video
+    encoding and decoding:
+
+    -   cudaconvert, cudascale: new CUDA based video color space convert
+        and rescale elements
+    -   cudaupload, cudadownload: new helper elements for memory
+        transfer between CUDA and system memory spaces
+    -   nvvp8sldec, nvvp9sldec: new GstCodecs-based VP8/VP9 decoders
+
+-   Various new hardware-accelerated elements for Windows:
+
+    -   d3d11screencapturesrc: new desktop capture element, including a
+        GstDeviceProvider implementation to enumerate/select target
+        monitors for capture.
+    -   d3d11av1dec and d3d11mpeg2dec: AV1 and MPEG-2 decoders
+    -   d3d11deinterlace: deinterlacing filter
+    -   d3d11compositor: video composing element
+    -   see Windows section below for more details
+
+-   new Rust plugins:
+
+    -   audiornnoise: Removes noise from an audio stream
+    -   awstranscribeparse: Parses AWS audio transcripts into timed text
+        buffers
+    -   ccdetect: Detects if valid closed captions are present in a
+        closed captions stream
+    -   cea608tojson: Converts CEA-608 Closed Captions to a JSON
+        representation
+    -   cmafmux: CMAF fragmented MP4 muxer
+    -   dashmp4mux: DASH fragmented MP4 muxer
+    -   isofmp4mux: ISO fragmented MP4 muxer
+    -   ebur128level: EBU R128 Loudness Level Measurement
+    -   ffv1dec: FFV1 video decoder
+    -   gtk4paintablesink: GTK4 video sink, which provides a
+        GdkPaintable that can be rendered in various widgets
+    -   hlssink3: HTTP Live Streaming sink
+    -   hrtfrender: Head-Related Transfer Function (HRTF) renderer
+    -   hsvdetector: HSV colorspace detector
+    -   hsvfilter: HSV colorspace filter
+    -   jsongstenc: Wraps buffers containing any valid top-level JSON
+        structures into higher level JSON objects, and outputs those as
+        ndjson
+    -   jsongstparse: Parses ndjson as output by jsongstenc
+    -   jsontovtt: converts JSON to WebVTT subtitles
+    -   regex: Applies regular expression operations on text
+    -   roundedcorners: Adds rounded corners to video
+    -   spotifyaudiosrc: Spotify source
+    -   textahead: Display upcoming text buffers ahead (e.g. for
+        Karaoke)
+    -   transcriberbin: passthrough bin that transcribes raw audio to
+        closed captions using awstranscriber and puts the captions as
+        metas onto the video
+    -   tttojson: Converts timed text to a JSON representation
+    -   uriplaylistbin: Playlist source bin
+    -   webpdec-rs: WebP image decoder with animation support
+
+-   New plugin codecalpha with elements to assist with WebM Alpha
+    decoding
+
+    -   codecalphademux: Split stream with GstVideoCodecAlphaMeta into
+        two streams
+    -   alphacombine: Combine two raw video stream (I420 or NV12) as one
+        stream with alpha channel (A420 or AV12)
+    -   vp8alphadecodebin: A bin to handle software decoding of VP8 with
+        alpha
+    -   vp9alphadecodebin: A bin to handle software decoding of VP9 with
+        alpha
+
+-   New hardware accelerated elements for Linux:
+
+    -   v4l2slmpeg2dec: Support for Linux Stateless MPEG2 decoders
+    -   v4l2slvp9dec: Support for Linux Stateless VP9 decoders
+    -   v4l2slvp8alphadecodebin: Support HW accelerated VP8 with alpha
+        layer decoding
+    -   v4l2slvp9alphadecodebin: Support HW accelerated VP9 with alpha
+        layer decoding
 
 New element features and additions
 
--   this section will be filled in in due course
+-   assrender: handle more font mime types; better interaction with
+    matroskademux for embedded fonts
+
+-   audiobuffersplit: Add support for specifying output buffer size in
+    bytes (not just duration)
+
+-   audiolatency: new "samplesperbuffer" property so users can configure
+    the number of samples per buffer. The default value is 240 samples
+    which is equivalent to 5ms latency with a sample rate of 48000,
+    which might be larger than actual buffer size of audio capture
+    device.
+
+-   audiomixer, audiointerleave, GstAudioAggregator: now keep a count of
+    samples that are dropped or processed as statistic and can be made
+    to post QoS messages on the bus whenever samples are dropped by
+    setting the "qos-messages" property on input pads.
+
+-   audiomixer, compositor: improved handling of new inputs added at
+    runtime. New API was added to the GstAggregator base class to allow
+    subclasses to opt into an aggregation mode where inactive pads are
+    ignored when processing input buffers
+    (gst_aggregator_set_ignore_inactive_pads(),
+    gst_aggregator_pad_is_inactive()). An “inactive pad” in this context
+    is a pad which, in live mode, hasn’t yet received a first buffer,
+    but has been waited on at least once. What would happen usually in
+    this case is that the aggregator would wait for data on this pad
+    every time, up to the maximum configured latency. This would
+    inadvertently push mixer elements in live mode to the configured
+    latency envelope and delay processing when new inputs are added at
+    runtime until these inputs have actually produced data. This is
+    usually undesirable. With this new API, new inputs can be added
+    (requested) and configured and they won’t delay the data processing.
+    Applications can opt into this new behaviour by setting the
+    "ignore-inactive-pads" property on compositor, audiomixer or other
+    GstAudioAggregator-based elements.
+
+-   cccombiner: implement “scheduling” of captions. So far cccombiner’s
+    behaviour was essentially that of a funnel: it strictly looked at
+    input timestamps to associate together video and caption buffers.
+    Now it will try to smoothly schedule caption buffers in order to
+    have exactly one per output video buffer. This might involve
+    rewriting input captions, for example when the input is CDP then
+    sequence counters are rewritten, time codes are dropped and
+    potentially re-injected if the input video frame had a time code
+    meta. This can also lead to the input drifting from synchronization,
+    when there isn’t enough padding in the input stream to catch up. In
+    that case the element will start dropping old caption buffers once
+    the number of buffers in its internal queue reaches a certain limit
+    (configurable via the "max-scheduled" property). The new original
+    funnel-like behaviour can be restored by setting the "scheduling"
+    property to FALSE.
+
+-   ccconverter: new "cdp-mode" property to specify which sections to
+    include in CDP packets (timecode, CC data, service info). Various
+    software, including ffmpeg’s Decklink support, fails parsing CDP
+    packets that contain anything but CC data in the CDP packets.
+
+-   clocksync: new "sync-to-first" property for automatic timestamp
+    offset setup: if set clocksync will set up the "ts-offset" value
+    based on the first buffer and the pipeline’s running time when the
+    first buffer arrived. The newly configured "ts-offset" in this case
+    would be the value that allows outputting the first buffer without
+    waiting on the clock. This is useful for example to feed a non-live
+    input into an already-running pipeline.
+
+-   compositor:
+
+    -   multi-threaded input conversion and compositing. Set the
+        "max-threads" property to activate this.
+    -   new "sizing-policy" property to support display aspect ratio
+        (DAR)-aware scaling. By default the image is scaled to fill the
+        configured destination rectangle without padding and without
+        keeping the aspect ratio. With sizing-policy=keep-aspect-ratio
+        the input image is scaled to fit the destination rectangle
+        specified by GstCompositorPad:{xpos, ypos, width, height}
+        properties preserving the aspect ratio. As a result, the image
+        will be centered in the destination rectangle with padding if
+        necessary.
+    -   new "zero-size-is-unscaled" property on input pads. By default
+        pad width=0 or pad height=0 mean that the stream should not be
+        scaled in that dimension. But if the "zero-size-is-unscaled"
+        property is set to FALSE a width or height of 0 is instead
+        interpreted to mean that the input image on that pad should not
+        be composited, which is useful when creating animations where an
+        input image is made smaller and smaller until it disappears.
+    -   improved handling of new inputs at runtime via
+        "ignore-inactive-pads"property (see above for details)
+    -   allow output format with alpha even if none of the inputs have
+        alpha (also glvideomixer and other GstVideoAggregator
+        subclasses)
+
+-   dashsink: add h265 codec support and signals for allowing custom
+    playlist/fragment output
+
+-   decodebin3:
+
+    -   improved decoder selection, especially for hardware decoders
+    -   make input activation “atomic” when adding inputs dynamically
+    -   better interleave handling: take into account decoder latency
+        for interleave size
+
+-   decklink:
+
+    -   Updated DeckLink SDK to 11.2 to support DeckLink 8K Pro
+    -   decklinkvideosrc:
+        -   More accurate and stable capture timestamps: use the
+            hardware reference clock time when the frame was finished
+            being captured instead of a clock time much further down the
+            road.
+        -   Automatically detect widescreen vs. normal NTSC/PAL
+
+-   encodebin:
+
+    -   add “smart encoding” support for H.265, VP8 and VP9 (i.e. only
+        re-encode where needed and otherwise pass through encoded video
+        as-is).
+    -   H264/H265 smart encoding improvements: respect user-specified
+        stream-format, but if not specified default to avc3/hvc1 with
+        in-band SPS/PPS/VPS signalling for more flexibility.
+    -   new encodebin2 element with dynamic/sometimes source pads in
+        order to support the option of doing the muxing outside of
+        encodebin, e.g. in combination with splitmuxsink.
+    -   add APIs to set element properties on encoding profiles (see
+        below)
+
+-   errorignore: new "ignore-eos" property to also ignore FLOW_EOS from
+    downstream elements
+
+-   giosrc: add support for growing source files: applications can
+    specify that the underlying file being read is growing by setting
+    the "is-growing" property. If set, the source won’t EOS when it
+    reaches the end of the file, but will instead start monitoring it
+    and will start reading data again whenever a change is detected. The
+    new "waiting-data" and "done-waiting-data" signals keep the
+    application informed about the current state.
+
+-   gtksink, gtkglsink:
+
+    -   scroll event support: forwarded as navigation events into the
+        pipeline
+    -   "video-aspect-ratio-override" property to force a specific
+        aspect ratio
+    -   "rotate-method" property and support automatic rotation based on
+        image tags
+
+-   identity: new "stats" property allows applications to retrieve the
+    number of bytes and buffers that have passed through so far.
+
+-   interlace: add support for more formats, esp 10-bit, 12-bit and
+    16-bit ones
+
+-   jack: new "low-latency" property for automatic latency-optimized
+    setting and "port-names" property to select ports explicitly
+
+-   jpegdec: support output conversion to RGB using libjpeg-turbo (for
+    certain input files)
+
+-   line21dec:
+
+    -   "mode" property to control whether and how detected closed
+        captions should be inserted in the list of existing close
+        caption metas on the input frame (if any): add, drop, or
+        replace.
+    -   "ntsc-only" property to only look for captions if video has NTSC
+        resolution
+
+-   line21enc: new "remove-caption-meta" to remove metas from output
+    buffers after encoding the captions into the video data; support for
+    CDP closed captions
+
+-   matroskademux, matroskamux: Add support for ffv1, a lossless
+    intra-frame video coding format.
+
+-   matroskamux: accept in-band SPS/PPS/VPS for H264 and H265
+    (i.e. stream-format avc3 and hev1) which allows on-the-fly
+    profile/level/resolution changes.
+
+-   matroskamux: new "cluster-timestamp-offset" property, useful for use
+    cases where the container timestamps should map to some absolute
+    wall clock time, for example.
+
+-   rtpsrc: add "caps" property to allow explicit setting of the caps
+    where needed
+
+-   mpegts: support SCTE-35 passthrough via new "send-scte35-events"
+    property on MPEG-TS demuxer tsdemux. When enabled, SCTE 35 sections
+    (eg ad placement opportunities) are forwarded as events donwstream
+    where they can be picked up again by mpegtsmux. This required a
+    semantic change in the SCTE-35 section API: timestamps are now in
+    running time instead of muxer pts.
+
+-   tsdemux: Handle PCR-less MPEG-TS streams; more robust timestamp
+    handling in certain corner cases and for poorly muxed streams.
+
+-   mpegtsmux:
+
+    -   More conformance improvements to make MPEG-TS analyzers happy:
+        -   PCR timing accuracy: Improvements to the way mpegtsmux
+            outputs PCR observations in CBR mode, so that a PCR
+            observation is always inserted when needed, so that we never
+            miss the configured pcr-interval, as that triggers various
+            MPEG-TS analyser errors.
+        -   Improved PCR/SI scheduling
+    -   Don’t write PCR until PAT/PMT are output to make sure streams
+        start cleanly with a PAT/PMT.
+    -   Allow overriding the automatic PMT PID selection via
+        application-supplied PMT_%d fields in the prog-map
+        structure/property.
+
+-   mp4mux:
+
+    -   new "first-moov-then-finalise" mode for fragmented output where
+        the output will start with a self-contained moov atom for the
+        first fragment, and then produce regular fragments. Then at the
+        end when the file is finalised, the initial moov is invalidated
+        and a new moov is written covering the entire file. This way the
+        file is a “fragmented mp4” file while it is still being written
+        out, and remains playable at all times, but at the end it is
+        turned into a regular mp4 file (with former fragment headers
+        remaining as unused junk data in the file).
+    -   support H.264 avc3 and H.265 hvc1 stream formats as input where
+        the codec data is signalled in-band inside the bitstream instead
+        of caps/file headers.
+    -   support profile/level/resolution changes for H264/H265 input
+        streams (i.e. codec data changing on the fly). Each codec_data
+        is put into its own SampleTableEntry inside the stsd, unless the
+        input is in avc3 stream format in which case it’s written
+        in-band an not in the headers.
+
+-   multifilesink: new ""min-keyframe-distance"" property to make
+    minimum distance between keyframes in next-file=key-frame mode
+    configurable instead of hard-coding it to 10 seconds.
+
+-   mxfdemux has seen a big refactoring to support non-frame wrappings
+    and more accurate timestamp/seek handling for some formats
+
+-   msdk plugin for hardware-accelerated video encoding and decoding
+    using the Intel Media SDK:
+
+    -   oneVPL support (Intel oneAPI Video Processing Library)
+    -   AV1 decoding support
+    -   H.264 decoder now supports constrained-high and progressive-high
+        profiles
+    -   H.264 encoder:
+        -   more configuration options (properties):
+            "intra-refresh-type", "min-qp" , "max-qp", "p-pyramid",
+            "dblk-idc"
+    -   H.265 encoder:
+        -   can output main-still-picture profile
+        -   now inserts HDR SEIs (mastering display colour volume and
+            content light level)
+        -   more configuration options (properties):
+            "intra-refresh-type", "min-qp" , "max-qp", "p-pyramid",
+            "b-pyramid", "dblk-idc", "transform-skip"
+        -   support for RGB 10bit format
+    -   External bitrate control in encoders
+    -   Video post proc element msdkvpp gained support for 12-bit pixel
+        formats P012_LE, Y212_LE and Y412_LE
+
+-   nvh264sldec: interlaced stream support
+
+-   openh264enc: support main, high, constrained-high and
+    progressive-high profiles
+
+-   openjpeg: support for multithreaded decoding and encoding
+
+-   rtspsrc: now supports IPv6 also for tunneled mode (RTSP-over-HTTP);
+    new "ignore-x-server-reply" property to ignore the
+    x-server-ip-address server header reply in case of HTTP tunneling,
+    as it is often broken.
+
+-   souphttpsrc: Runtime compatibility support for libsoup2 and
+    libsoup3. libsoup3 is the latest major version of libsoup, but
+    libsoup2 and libsoup3 can’t co-exist in the same process because
+    there is no namespacing or versioning for GObject types. As a
+    result, it would be awkward if the GStreamer souphttpsrc plugin
+    linked to a specific version of libsoup, because it would only work
+    with applications that use the same version of libsoup. To make this
+    work, the soup plugin now tries to determine the libsoup version
+    used by the application (and its other dependencies) at runtime on
+    systems where GStreamer is linked dynamically. libsoup3 support is
+    still considered somewhat experimental at this point.
+
+-   srtsrc, srtsink: add signals for the application to accept/reject
+    incoming connections
+
+-   timeoverlay: new elapsed-running-time time mode which shows the
+    running time since the first running time (and each flush-stop).
+
+-   udpsrc: new timestamping mode to retrieve packet receive timestamps
+    from the kernel via socket control messages (SO_TIMESTAMPNS) on
+    supported platforms
+
+-   uritranscodebin: new setup-source and element-setup signals for
+    applications to configure elements used
+
+-   v4l2codecs plugin gained support for 4x4 and 32x32 tile formats
+    enabling some platforms or direct renders. Important memory usage
+    improvement.
+
+-   v4l2slh264dec now implements the final Linux uAPI as shipped on
+    Linux 5.11 and later.
+
+-   valve: add "drop-mode" property and provide two new modes of
+    operation: in drop-mode=forward-sticky-events sticky events
+    (stream-start, segment, tags, caps, etc.) are forwarded downstream
+    even when dropping is enabled; drop-mode=transform-to-gap will in
+    addition also convert buffers into gap events when dropping is
+    enabled, which lets downstream elements know that time is advancing
+    and might allow for preroll in many scenarios. By default all events
+    and all buffers are dropped when dropping is enabled, which can
+    cause problems with caps negotiation not progressing or branches not
+    prerolling when dropping is enabled.
+
+-   videocrop: support for many more pixel formats, e.g. planar YUV
+    formats with > 8bits and GBR* video formats; can now also accept
+    video not backed by system memory as long as downstream supports the
+    GstCropMeta
+
+-   videotestsrc: new smpte-rp-219 pattern for SMPTE75 RP-219 conformant
+    color bars
+
+-   vp8enc: finish support for temporal scalability: two new properties
+    ("temporal-scalability-layer-flags",
+    "temporal-scalability-layer-sync-flags") and a unit change on the
+    "temporal-scalability-target-bitrate" property (now expects bps);
+    also make temporal scalability details available to RTP payloaders
+    as buffer metadata.
+
+-   vp9enc: new properties to tweak encoder performance:
+
+    -   "aq-mode" to configure adaptive quantization modes
+    -   "frame-parallel-decoding" to configure whether to create a
+        bitstream that reduces decoding dependencies between frames
+        which allows staged parallel processing of more than one video
+        frames in the decoder. (Defaults to TRUE)
+    -   "row-mt", "tile-columns" and "tile-rows" so multithreading can
+        be enabled on a per-tile basis, instead of on a per tile-column
+        basis. In combination with the new "tile-rows" property, this
+        allows the encoder to make much better use of the available CPU
+        power.
+
+-   vp9dec, vp9enc: add support for 10-bit 4:2:0 and 4:2:2 YUV, as well
+    as 8-bit 4:4:4
+
+-   vp8enc, vp9enc now default to “good quality” for the deadline
+    property rather then “best quality”. Having the deadline set to best
+    quality causes the encoder to be absurdly slow, most real-life users
+    will prefer good-enough quality with better performance instead.
+
+-   wpesrc:
+
+    -   implement audio support: a new sometimes source pad will be
+        created for each audio stream created by the web engine.
+    -   move wpesrc to wpevideosrc and add a wrapper bin wpesrc to also
+        support audio
+    -   also handles web:// URIs now (same as cefsrc)
+    -   post messages with the estimated load progress on the bus
 
-Plugin and library moves
+-   x265enc: add negative DTS support, which means timestamps are now
+    offset by 1h same as with x264enc
+
+RTP Payloaders and Depayloaders
+
+-   rtpisacpay, rtpisacdepay: new RTP payloader and depayloader for iSAC
+    audio codec
+
+-   rtph264depay:
+
+    -   new "request-keyframe" property to make the depayloader
+        automatically request a new keyframe from the sender on packet
+        loss, consistent with the new property on rtpvp8depay.
+    -   new "wait-for-keyframe" property to make depayloader wait for a
+        new keyframe at the beginning and after packet loss (only
+        effective if the depayloader outputs AUs), consistent with the
+        existing property on rtpvp8depay.
+
+-   rtpopuspay, rtpopusdepay: support libwebrtc-compatible multichannel
+    audio in addition to the previously supported multichannel audio
+    modes
+
+-   rtpopuspay: add DTX (Discontinuous Transmission) support
+
+-   rtpvp8depay: new "request-keyframe" property to make the depayloader
+    automatically request a new keyframe from the sender on packet loss.
+
+-   rtpvp8pay: temporal scaling support
 
--   this section will be filled in in due course
+-   rtpvp9depay: Improved SVC handling (aggregate all layers)
+
+RTP Infrastructure
+
+-   rtpst2022-1-fecdec, rtpst2022-1-fecenc: new elements providing SMPTE
+    2022-1 2-D Forward Error Correction. More details in Mathieu’s blog
+    post.
+
+-   rtpreddec: BUNDLE support
+
+-   rtpredenc, rtpulpfecenc: add support for Transport-wide Congestion
+    Control (TWCC)
+
+-   rtpsession: new "twcc-feedback-interval" property to allow RTCP TWCC
+    reports to be scheduled on a timer instead of per marker-bit.
+
+Plugin and library moves
 
 -   There were no plugin moves or library moves in this cycle.
 
@@ -56,8 +998,6 @@ Plugin removals
 
 The following elements or plugins have been removed:
 
--   this section will be filled in in due course
-
 -   The ofa audio fingerprinting plugin has been removed. The MusicIP
     database has been defunct for years so this plugin is likely neither
     useful nor used by anyone.
@@ -71,135 +1011,575 @@ The following elements or plugins have been removed:
 
 Miscellaneous API additions
 
--   this section will be filled in in due course
+Core
+
+-   gst_buffer_new_memdup() is a convenience function for the
+    widely-used gst_buffer_new_wrapped(g_memdup(data,size),size)
+    pattern.
+
+-   gst_caps_features_new_single() creates a new single GstCapsFeatures,
+    avoiding the need to use the vararg function with NULL terminator
+    for simple cases.
+
+-   gst_element_type_set_skip_documentation() can be used by plugins to
+    signal that certain elements should not be included in the GStreamer
+    plugin documentation. This is useful for plugins where elements are
+    registered dynamically based on hardware capabilities and/or where
+    the available plugins and properties vary from system to system.
+    This is used in the d3d11 plugin for example to ensure that only the
+    list of default elements is advertised in the documentation.
+
+-   gst_type_find_suggest_empty_simple() is a new convenience function
+    for typefinders for cases where there’s only a media type and no
+    other fields.
+
+-   New API to create elements and set properties at construction time,
+    which is not only convenient, but also allows GStreamer elements to
+    have construct-only properties: gst_element_factory_make_full(),
+    gst_element_factory_make_valist(),
+    gst_element_factory_make_with_properties(),
+    gst_element_factory_create_full(),
+    gst_element_factory_create_valist(),
+    gst_element_factory_create_with_properties().
+
+-   GstSharedTaskPool: new “shared” task pool subclass with slightly
+    different default behaviour than the existing GstTaskPool which
+    would create unlimited number of threads for new tasks. The shared
+    taskpool creates up to N threads (default: 1) and then distributes
+    pending tasks to those threads round-robin style, and blocks if no
+    thread is available. It is possible to join tasks. This can be used
+    by plugins to implement simple multi-threaded processing and is used
+    for the new multi-threaded video conversion and compositing done in
+    GstVideoAggregator, videoconverter and compositor.
+
+Plugins Base Utils library
+
+-   GstDiscoverer:
+
+    -   gst_discoverer_container_info_get_tags() was added to retrieve
+        global/container tags (vs. per-stream tags). Per-Stream tags can
+        be retrieved via the existing
+        gst_discoverer_stream_info_get_tags().
+        gst_discoverer_info_get_tags(), which for many files returns a
+        confusing mix of stream and container tags, has been deprecated
+        in favour of the container/stream-specific functions.
+    -   gst_discoverer_stream_info_get_stream_number() returns a unique
+        integer identifier for a given stream within the given
+        GstDiscoverer context. (If this matches the stream number inside
+        the container bitstream that’s by coincidence and not by
+        design.)
+
+-   gst_pb_utils_get_caps_description_flags() can be used to query
+    whether certain caps represent a container, audio, video, image,
+    subtitles, tags, or something else. This only works for formats
+    known to GStreamer.
+
+-   gst_pb_utils_get_file_extension_from_caps() returns a possible file
+    extension for given caps.
+
+-   gst_codec_utils_h264_get_profile_flags_level(): Parses profile,
+    flags, and level from H264 AvcC codec_data. The format of H264 AVCC
+    extradata/sequence_header is documented in the ITU-T H.264
+    specification section 7.3.2.1.1 as well as in ISO/IEC 14496-15
+    section 5.3.3.1.2.
+
+-   gst_codec_utils_caps_get_mime_codec() to convert caps to a RFC 6381
+    compatible MIME codec string codec. Useful for providing the codecs
+    field inside the Content-Type HTTP header for containerized formats,
+    such as mp4 or matroska.
+
+GStreamer OpenGL integration library and plugins
+
+-   glcolorconvert: added suppport for converting the video formats
+    A420, AV12, BGR, BGRA, RGBP and BGRP.
+
+-   Added support to GstGLBuffer for persistent buffer mappings where a
+    Pixel Buffer Object (PBO) can be mapped by both the CPU and the GPU.
+    This removes a memcpy() when uploading textures or vertices
+    particularly when software decoders (e.g. libav) are direct
+    rendering into our memory. Improves transfer performance
+    significantly. Requires OpenGL 4.4, GL_ARB_buffer_storage or
+    GL_EXT_buffer_storage
+
+-   Added various helper functions for handling 4x4 matrices of affine
+    transformations as used by GstVideoAffineTransformationMeta.
+
+-   Add support to GstGLContext for allowing the application to control
+    the config (EGLConfig, GLXConfig, etc) used when creating the OpenGL
+    context. This allows the ability to choose between RGB16 or RGB10A2
+    or RGBA8 back/front buffer configurations that were previously
+    hardcoded. GstGLContext also supports retrieving the configuration
+    it was created with or from an externally provide OpenGL context
+    handle. This infrastructure is also used to create a compatible
+    config from an application/externally provided OpenGL context in
+    order to improve compatibility with other OpenGL frameworks and GUI
+    toolkits. A new environment variable GST_GL_CONFIG was also added to
+    be able to request a specific configuration from the command line.
+    Note: different platforms will have different functionality
+    available.
+
+-   Add support for choosing between EGL and WGL at runtime when running
+    on Windows. Previously this was a build-time switch. Allows use in
+    e.g. Gtk applications on Windows that target EGL/ANGLE without
+    recompiling GStreamer. gst_gl_display_new_with_type() can be used by
+    applications to choose a specific display type to use.
+
+-   Build fixes to explicitly check for Broadcom-specific libraries on
+    older versions of the Raspberry Pi platform. The Broadcom OpenGL ES
+    and EGL libraries have different filenames. Using the vc4 Mesa
+    driver on the Raspberry Pi is not affected.
+
+-   Added support to glupload and gldownload for transferring RGBA
+    buffers using the memory:NVMM available on the Nvidia Tegra family
+    of embedded devices.
+
+-   Added support for choosing libOpenGL and libGLX as used in a GLVND
+    environment on unix-based platforms. This allows using desktop
+    OpenGL and EGL without pulling in any GLX symbols as would be
+    required with libGL.
+
+Video library
+
+-   New raw video formats:
+
+    -   AV12 (NV12 with alpha plane)
+    -   RGBP and BGRP (planar RGB formats)
+    -   ARGB64 variants with specified endianness instead of host
+        endianness:
+        -   ARGB64_LE, ARGB64_BE
+        -   RGBA64_BE, RGBA64_LE
+        -   BGRA64_BE, BGRA64_LE
+        -   ABGR64_BE, ABGR64_LE
+
+-   gst_video_orientation_from_tag() is new convenience API to parse the
+    image orientation from a GstTagList.
+
+-   GstVideoDecoder subframe support (see below)
+
+-   GstVideoCodecState now also carries some HDR metadata
+
+-   Ancillary video data: implement transform functions for AFD/Bar
+    metas, so they will be forwarded in more cases
+
+MPEG-TS library
+
+This library only handles section parsing and such, see above for
+changes to the actual mpegtsmux and mpegtsdemux elements.
+
+-   many additions and improvements to SCTE-35 section parsing
+-   new API for fetching extended descriptors:
+    gst_mpegts_find_descriptor_with_extension()
+-   add support for SIT sections (Selection Information Tables)
+-   expose event-from-section constructor gst_event_new_mpegts_section()
+-   parse Audio Preselection Descriptor needed for Dolby AC-4
+
+GstWebRTC library + webrtcbin
+
+-   Change the way in which sink pads and transceivers are matched
+    together to support easier usage. If a pad is created without a
+    specific index (i.e. using sink_%u as the pad template), then an
+    available compatible transceiver will be searched for. If a specific
+    index is requested (i.e. sink_1) then if a transceiver for that
+    m-line already exists, that transceiver must match the new sink pad
+    request. If there is no transceiver available in either scenario, a
+    new transceiver is created. If a mixture of both sink_1 and sink_%u
+    requests result in an impossible situation, an error will be
+    produced at pad request time or from create offer/answer.
+
+-   webrtcbin now uses regular ICE nomination instead of libnice’s
+    default of aggressive ICE nomination. Regular ICE nomination is the
+    default recommended by various relevant standards and improves
+    connectivity in specific network scenarios.
+
+-   Add support for limiting the port range used for RTP with the
+    addition of the min-rtp-port and max-rtp-port properties on the ICE
+    object.
+
+-   Expose the SCTP transport as a property on webrtcbin to more closely
+    match the WebRTC specification.
+
+-   Added support for taking into account the data channel transport
+    state when determining the value of the "connection-state" property.
+    Previous versions of the WebRTC spec did not include the data
+    channel state when computing this value.
+
+-   Add configuration for choosing the size of the underlying sockets
+    used for transporting media data
+
+-   Always advertise support for the transport-cc RTCP feedback protocol
+    as rtpbin supports it. For full support, the configured caps (input
+    or through codec-preferences) need to include the relevant RTP
+    header extension.
+
+-   Numerous fixes to caps and media handling to fail-fast when an
+    incompatible situation is detected.
+
+-   Improved support for attaching the required media after a remote
+    offer has been set.
+
+-   Add support for dynamically changing the amount of FEC used for a
+    particular stream.
+
+-   webrtcbin now stops further SDP processing at the first error it
+    encounters.
+
+-   Completed support for either local or the remote closing a data
+    channel.
+
+-   Various fixes when performing BUNDLEing of the media streams in
+    relation to RTX and FEC usage.
+
+-   Add support for writing out QoS DSCP marking on outgoing packets to
+    improve reliability in some network scenarios.
+
+-   Improvements to the statistics returned by the get-stats signal
+    including the addition of the raw statistics from the internal
+    RTPSource, the TWCC stats when available.
+
+-   The webrtc library does not expose any objects anymore with public
+    fields. Instead properties have been added to replace that
+    functionality. If you are accessing such fields in your application,
+    switch to the corresponding properties.
+
+GstCodecs and Video Parsers
+
+-   Support for render delays to improve throughput across all CODECs
+    (used with NVDEC and V4L2).
+-   lots of improvements to parsers and the codec parsing decoder base
+    classes (H264, H265, VP8, VP9, AV1, MPEG-2) used for various
+    hardware-accelerate decoder APIs.
+
+Bindings support
+
+-   gst_allocation_params_new() allocates a GstAllocationParams struct
+    on the heap. This should only be used by bindings (and freed via
+    gst_allocation_params_free() then). In C code you would allocate
+    this on the stack and only init it in place.
+
+-   gst_debug_log_literal() can be used to log a string to the debug log
+    without going through any printf format expansion and associated
+    overhead. This is mostly useful for bindings such as the Rust
+    bindings which may have done their own formatting already .
+
+-   Provide non-inlined versions of refcounting APIs for various
+    GStreamer mini objects, so that they can be consumed by bindings
+    (e.g. gstreamer-sharp): gst_buffer_ref, gst_buffer_unref,
+    gst_clear_buffer, gst_buffer_copy, gst_buffer_replace,
+    gst_buffer_list_ref, gst_buffer_list_unref, gst_clear_buffer_list,
+    gst_buffer_list_copy, gst_buffer_list_replace, gst_buffer_list_take,
+    gst_caps_ref, gst_caps_unref, gst_clear_caps, gst_caps_replace,
+    gst_caps_take, gst_context_ref, gst_context_unref, gst_context_copy,
+    gst_context_replace, gst_event_replace, gst_event_steal,
+    gst_event_take, gst_event_ref, gst_event_unref, gst_clear_event,
+    gst_event_copy, gst_memory_ref, gst_memory_unref, gst_message_ref,
+    gst_message_unref, gst_clear_message, gst_message_copy,
+    gst_message_replace, gst_message_take, gst_promise_ref,
+    gst_promise_unref, gst_query_ref, gst_query_unref, gst_clear_query,
+    gst_query_copy, gst_query_replace, gst_query_take, gst_sample_ref,
+    gst_sample_unref, gst_sample_copy, gst_tag_list_ref,
+    gst_tag_list_unref, gst_clear_tag_list, gst_tag_list_replace,
+    gst_tag_list_take, gst_uri_copy, gst_uri_ref, gst_uri_unref,
+    gst_clear_uri.
+
+-   expose a GType for GstMiniObject
+
+-   gst_device_provider_probe() now returns non-floating device object
+
+API Deprecations
+
+-   gst_element_get_request_pad() has been deprecated in favour of the
+    newly-added gst_element_request_pad_simple() which does the exact
+    same thing but has a less confusing name that hopefully makes clear
+    that the function request a new pad rather than just retrieves an
+    already-existing request pad.
+
+-   gst_discoverer_info_get_tags(), which for many files returns a
+    confusing mix of stream and container tags, has been deprecated in
+    favour of the container-specific and stream-specific functions,
+    gst_discoverer_container_info_get_tags() and
+    gst_discoverer_stream_info_get_tags().
+
+-   gst_video_sink_center_rect() was deprecated in favour of the more
+    generic newly-added gst_video_center_rect().
+
+-   The GST_MEMORY_FLAG_NO_SHARE flag has been deprecated, as it tends
+    to cause problems and prevents sub-buffering. If pooling or lifetime
+    tracking is required, memories should be allocated through a custom
+    GstAllocator instead of relying on the lifetime of the buffers the
+    memories were originally attached to, which is fragile anyway.
+
+-   The GstPlayer high-level playback library is being replaced with the
+    new GstPlay library (see above). GstPlayer should be considered
+    deprecated at this point and will be marked as such in the next
+    development cycle. Applications should be ported to GstPlay.
+
+-   Gstreamer Editing Services: ges_video_transition_set_border(),
+    ges_video_transition_get_border()
+    ges_video_transition_set_inverted()
+    ges_video_transition_is_inverted() have been deprecated, use
+    ges_timeline_element_set_children_properties() instead.
 
 Miscellaneous performance, latency and memory optimisations
 
--   this section will be filled in in due course
+More video conversion fast paths
 
-Miscellaneous other changes and enhancements
+-   v210 ↔ I420, YV12, Y42B, UYVY and YUY2
+-   A420 → RGB
+
+Less jitter when waiting on the system clock
+
+-   Better system clock wait accuracy, less jitter: where available,
+    clock_nanosleep is used for higher accuracy for waits below 500
+    usecs, and waits below 2ms will first use the regular waiting system
+    and then clock_nanosleep for the remainder. The various wait
+    implementation have a latency ranging from 50 to 500+ microseconds.
+    While this is not a major issue when dealing with a low number of
+    waits per second (for ex: video), it does introduce a non-negligible
+    jitter for synchronisation of higher packet rate systems.
+
+Video decoder subframe support
 
--   this section will be filled in in due course
+-   The GstVideoDecoder base class gained API to process input at the
+    sub-frame level. That way video decoders can start decoding slices
+    before they have received the full input frame in its entirety (to
+    the extent this is supported by the codec, of course). This helps
+    with CPU utilisation and reduces latency.
 
-Tracing framework and debugging improvements
+-   This functionality is now being used in the OpenJPEG JPEG 2000
+    decoder, the FFmpeg H.264 decoder (in case of NAL-aligned input) and
+    the OpenMAX H.264/H.265 decoders (in case of NAL-aligned input).
 
--   this section will be filled in in due course
+Miscellaneous other changes and enhancements
+
+-   GstDeviceMonitor no longer fails to start just because one of the
+    device providers failed to start. That could happen for example on
+    systems where the pulseaudio device provider is installed, but
+    pulseaudio isn’t actually running but ALSA is used for audio
+    instead. In the same vein the device monitor now keeps track of
+    which providers have been started (via the new
+    gst_device_provider_is_started()) and only stops actually running
+    device providers when stopping the device monitor.
+
+-   On embedded systems it can be useful to create a registry that can
+    be shared and read by multiple processes running as different users.
+    It is now possible to set the new GST_REGISTRY_MODE environment
+    variable to specify the file mode for the registry file, which by
+    default is set to be only user readable/writable.
+
+-   GstNetClientClock will signal lost sync in case the remote time
+    resets (e.g. because device power cycles), by emitting the “synced”
+    signal with synced=FALSE parameter, so applications can take action.
+
+-   gst_value_deserialize_with_pspec() allows deserialization with a
+    hint for what the target GType should be. This allows for example
+    passing arrays of flags through the command line or
+    gst_util_set_object_arg(), eg: foo="<bar,bar+baz>".
+
+-   It’s now allowed to create an empty GstVideoOverlayComposition
+    without any rectangles by passing a NULL rectangle to
+    gst_video_overlay_composition_new(). This is useful for bindings and
+    simplifies application code in some places.
+
+Tracing framework, debugging and testing improvements
+
+-   New factories tracer to list loaded elements (and other plugin
+    features). This can be useful to collect a list of elements needed
+    for an application, which then in turn can be used to create a
+    tailored minimal GStreamer build that contains just the elements
+    needed and nothing else.
+-   New plugin-feature-loaded tracing hook for use by tracers like the
+    new factories tracer
+
+-   GstHarness: Add gst_harness_set_live() so that harnesses can be set
+    to non-live and return is-live=false in latency queries if needed.
+    Default behaviour is to always return is-live=true in latency
+    queries.
+
+-   navseek: new "hold-eos" property. When enabled, the element will
+    hold back an EOS event until the next keystroke (via navigation
+    events). This can be used to keep a video sink showing the last
+    frame of a video pipeline until a key is pressed instead of tearing
+    it down immediately on EOS.
+
+-   New fakeaudiosink element: mimics an audio sink and can be used for
+    testing and CI pipelines on systems where no audio system is
+    installed or running. It differs from fakesink in that it only
+    support audio caps and syncs to the clock by default like a normal
+    audio sink. It also implements the GstStreamVolume interface like
+    most audio sinks do.
+
+-   New videocodectestsink element for video codec conformance testing:
+    Calculates MD5 checksums for video frames and skips any padding
+    whilst doing so. Can optionally also write back the video data with
+    padding removed into a file for easy byte-by-byte comparison with
+    reference data.
 
 Tools
 
--   this section will be filled in in due course
+gst-inspect-1.0
+
+-   Can sort the list of plugins by passing --sort=name as command line
+    option
+
+gst-launch-1.0
+
+-   will now error out on top-level properties that don’t exist and
+    which were silently ignored before
+-   On Windows the high-resolution clock is enabled now, which provides
+    better clock and timer performance on Windows (see Windows section
+    below for more details).
+
+gst-play-1.0
+
+-   New --start-position command line argument to start playback from
+    the specified position
+-   Audio can be muted/unmuted in interactive mode by pressing the m
+    key.
+-   On Windows the high-resolution clock is enabled now (see Windows
+    section below for more details)
+
+gst-device-monitor-1.0
+
+-   New --include-hidden command line argument to also show “hidden”
+    device providers
+
+ges-launch-1.0
+
+-   New interactive mode that allows seeking and such. Can be disabled
+    by passing the --no-interactive argument on the command line.
+-   Option to forward tags
+-   Allow using an existing clip to determine the rendering format (both
+    topology and profile) via new --profile-from command line argument.
 
 GStreamer RTSP server
 
--   this section will be filled in in due course
+-   GstRTSPMediaFactory gained API to disable RTCP
+    (gst_rtsp_media_factory_set_enable_rtcp(), "enable-rtcp" property).
+    Previously RTCP was always allowed for all RTSP medias. With this
+    change it is possible to disable RTCP completely, no matter if the
+    client wants to do RTCP or not.
+
+-   Make a mount point of / work correctly. While not allowed by the
+    RTSP 2 spec, the RTSP 1 spec is silent on this and it is used in the
+    wild. It is now possible to use / as a mount path in
+    gst-rtsp-server, e.g. rtsp://example.com/ would work with this now.
+    Note that query/fragment parts of the URI are not necessarily
+    correctly handled, and behaviour will differ between various
+    client/server implementations; so use it if you must but don’t bug
+    us if it doesn’t work with third party clients as you’d hoped.
+
+-   multithreading fixes (races, refcounting issues, deadlocks)
+
+-   ONVIF audio backchannel fixes
+
+-   ONVIF trick mode optimisations
+
+-   rtspclientsink: new "update-sdp" signal that allows updating the SDP
+    before sending it to the server via ANNOUNCE. This can be used to
+    add additional metadata to the SDP, for example. The order and
+    number of medias must not be changed, however.
 
 GStreamer VAAPI
 
--   this section will be filled in in due course
+-   new AV1 decoder element (vaapiav1dec)
+
+-   H264 decoder: handle stereoscopic 3D video with frame packing
+    arrangement SEI messages
+
+-   H265 encoder: added Screen Content Coding extensions support
+
+-   H265 decoder: gained MAIN_444_12 profile support (decoded to
+    Y412_LE), and 4:2:2 12-bits support (decoded to Y212_LE)
+
+-   vaapipostproc: gained BT2020 color standard support
+
+-   vaapidecode: now generates caps templates dynamically at runtime in
+    order to advertise actually supported caps instead of all
+    theoretically supported caps.
+
+-   GST_VAAPI_DRM_DEVICE environment variable to force a specified DRM
+    device when a DRM display is used. It is ignored when other types of
+    displays are used. By default /dev/dri/renderD128 is used for DRM
+    display.
 
 GStreamer OMX
 
--   this section will be filled in in due course
+-   subframe support in H.264/H.265 decoders
 
 GStreamer Editing Services and NLE
 
--   this section will be filled in in due course
+-   framepositioner: new "operator" property to access blending modes in
+    the compositor
+-   timeline: Implement snapping to markers
+-   smart-mixer: Add support for d3d11compositor and glvideomixer
+-   titleclip: add "draw-shadow" child property
+-   ges:// URI support to define a timeline from a description.
+-   command-line-formatter
+    -   Add track management to timeline description
+    -   Add keyframe support
+-   ges-launch-1.0:
+    -   Add an interactive mode where we can seek etc…
+    -   Add option to forward tags
+    -   Allow using an existing clip to determine the rendering format
+        (both topology and profile) via new --profile-from command line
+        argument.
+-   Fix static build
 
 GStreamer validate
 
--   this section will be filled in in due course
+-   report: Add a way to force backtraces on reports even if not a
+    critical issue (GST_VALIDATE_ISSUE_FLAGS_FORCE_BACKTRACE)
+-   Add a flag to gst_validate_replace_variables_in_string() allow
+    defining how to resolve variables in structs
+-   Add gst_validate_bin_monitor_get_scenario() to get the bin monitor
+    scenario, which is useful for applications that use Validate
+    directly.
+-   Add an expected-values parameter to wait, message-type=XX allowing
+    more precise filtering of the message we are waiting for.
+-   Add config file support: each test can now use a config file for the
+    given media file used to test.
+-   Add support to check properties of object properties
+-   scenario: Add an "action-done" signal to signal when an action is
+    done
+-   scenario: Add a "run-command" action type
+-   scenario: Allow forcing running action on idle from scenario file
+-   scenario: Allow iterating over arrays in foreach
+-   scenario: Rename ‘interlaced’ action to ‘non-blocking’
+-   scenario: Add a non-blocking flag to the wait signal
 
 GStreamer Python Bindings
 
--   this section will be filled in in due course
+-   Fixes for Python 3.10
+-   Various build fixes
+-   at least one known breaking change caused by g-i annotation changes
+    (see below)
 
 GStreamer C# Bindings
 
--   this section will be filled in in due course
+-   Fix GstDebugGraphDetails enum
+-   Updated to latests GtkSharp
+-   Updated to include GStreamer 1.20 API
 
 GStreamer Rust Bindings and Rust Plugins
 
-The GStreamer Rust bindings are released separately with a different
-release cadence that’s tied to gtk-rs, but the latest release has
-already been updated for the upcoming new GStreamer 1.20 API.
-
-gst-plugins-rs, the module containing GStreamer plugins written in Rust,
-has also seen lots of activity with many new elements and plugins.
-
-What follows is a list of elements and plugins available in
-gst-plugins-rs, so people don’t miss out on all those potentially useful
-elements that have no C equivalent.
-
--   FIXME: add new elements
-
-Rust audio plugins
-
--   audiornnoise: New element for audio denoising which implements the
-    noise removal algorithm of the Xiph RNNoise library, in Rust
--   rsaudioecho: Port of the audioecho element from gst-plugins-good
-    rsaudioloudnorm: Live audio loudness normalization element based on
-    the FFmpeg af_loudnorm filter
--   claxondec: FLAC lossless audio codec decoder element based on the
-    pure-Rust claxon implementation
--   csoundfilter: Audio filter that can use any filter defined via the
-    Csound audio programming language
--   lewtondec: Vorbis audio decoder element based on the pure-Rust
-    lewton implementation
-
-Rust video plugins
-
--   cdgdec/cdgparse: Decoder and parser for the CD+G video codec based
-    on a pure-Rust CD+G implementation, used for example by karaoke CDs
--   cea608overlay: CEA-608 Closed Captions overlay element
--   cea608tott: CEA-608 Closed Captions to timed-text (e.g. VTT or SRT
-    subtitles) converter
--   tttocea608: CEA-608 Closed Captions from timed-text converter
--   mccenc/mccparse: MacCaption Closed Caption format encoder and parser
--   sccenc/sccparse: Scenarist Closed Caption format encoder and parser
--   dav1dec: AV1 video decoder based on the dav1d decoder implementation
-    by the VLC project
--   rav1enc: AV1 video encoder based on the fast and pure-Rust rav1e
-    encoder implementation
--   rsflvdemux: Alternative to the flvdemux FLV demuxer element from
-    gst-plugins-good, not feature-equivalent yet
--   rsgifenc/rspngenc: GIF/PNG encoder elements based on the pure-Rust
-    implementations by the image-rs project
-
-Rust text plugins
-
--   textwrap: Element for line-wrapping timed text (e.g. subtitles) for
-    better screen-fitting, including hyphenation support for some
-    languages
-
-Rust network plugins
-
--   reqwesthttpsrc: HTTP(S) source element based on the Rust
-    reqwest/hyper HTTP implementations and almost feature-equivalent
-    with the main GStreamer HTTP source souphttpsrc
--   s3src/s3sink: Source/sink element for the Amazon S3 cloud storage
--   awstranscriber: Live audio to timed text transcription element using
-    the Amazon AWS Transcribe API
-
-Generic Rust plugins
-
--   sodiumencrypter/sodiumdecrypter: Encryption/decryption element based
-    on libsodium/NaCl
--   togglerecord: Recording element that allows to pause/resume
-    recordings easily and considers keyframe boundaries
--   fallbackswitch/fallbacksrc: Elements for handling potentially
-    failing (network) sources, restarting them on errors/timeout and
-    showing a fallback stream instead
--   threadshare: Set of elements that provide alternatives for various
-    existing GStreamer elements but allow to share the streaming threads
-    between each other to reduce the number of threads
--   rsfilesrc/rsfilesink: File source/sink elements as replacements for
-    the existing filesrc/filesink elements
+-   The GStreamer Rust bindings are released separately with a different
+    release cadence that’s tied to gtk-rs, but the latest release has
+    already been updated for the upcoming new GStreamer 1.20 API (v1_20
+    feature).
+
+-   gst-plugins-rs, the module containing GStreamer plugins written in
+    Rust, has also seen lots of activity with many new elements and
+    plugins. See the New Elements section above for a list of new Rust
+    elements.
 
 Build and Dependencies
 
 -   Meson 0.59 or newer is required to build GStreamer now.
 
--   FIXME: this section will be filled in in due course
+-   The GLib requirement has been bumped to GLib 2.56 or newer (from
+    March 2018).
+
+-   The wpe plugin now requires wpe >= 2.28 and wpebackend-fdo >= 1.8
 
 Explicit opt-in required for build of certain plugins with (A)GPL dependencies
 
@@ -211,11 +1591,9 @@ even if the required dependencies are available.
 See Building plugins with (A)GPL-licensed dependencies for more details
 and a non-exhaustive list of plugins affected.
 
-gst-build: replaced by Monorepo
+gst-build: replaced by mono repository
 
--   this section will be filled in in due course
-
--   FIXME: describe + link to Monorepo FAQ
+See mono repository section above and the GStreamer mono repository FAQ.
 
 Cerbero
 
@@ -223,76 +1601,254 @@ Cerbero is a meta build system used to build GStreamer plus dependencies
 on platforms where dependencies are not readily available, such as
 Windows, Android, iOS and macOS.
 
-General improvements
+General Cerbero improvements
 
--   this section will be filled in in due course
+-   Plugin removed: libvisual
+-   New plugins: rtpmanagerbad and rist
 
-macOS / iOS
+macOS / iOS specific Cerbero improvements
 
--   this section will be filled in in due course
+-   XCode 12 support
+-   macOS OS release support is now future-proof, similar to iOS
+-   macOS Apple Silicon (ARM64) cross-compile support has been added
+-   macOS Apple Silicon (ARM64) native support is currently experimental
 
-Windows
+Windows specific Cerbero improvements
 
--   this section will be filled in in due course
+-   Visual Studio 2022 support has been added
+-   bootstrap is faster since it requires building fewer build-tools
+    recipes on Windows
+-   package is faster due to better scheduling of recipe stages and
+    elimination of unnecessary autotools regeneration
+-   The following plugins are no longer built on Windows:
+    -   a52dec (another decoder is still available in libav)
+    -   dvdread
+    -   resindvd
 
 Windows MSI installer
 
--   this section will be filled in in due course
+-   no major changes
 
-Linux
+Linux specific Cerbero improvements
 
--   this section will be filled in in due course
+-   Fedora, Debian OS release support is now more future-proof
+-   Amazon Linux 2 support has been added
 
-Android
+Android specific Cerbero improvements
 
--   this section will be filled in in due course
+-   no major changes
 
 Platform-specific changes and improvements
 
 Android
 
--   this section will be filled in in due course
+-   No major changes
 
 macOS and iOS
 
--   this section will be filled in in due course
+-   applemedia: add ProRes support to vtenc and vtdec
 
 Windows
 
--   this section will be filled in in due course
+-   On Windows the high-resolution clock is enabled now in the
+    gst-launch-1.0 and gst-play-1.0 command line tools, which provides
+    better clock and timer performance on Windows, at the cost of higher
+    power consumption. By default, without the high-resolution clock
+    enabled, the timer precision on Windows is system-dependent and may
+    be as bad as 15ms which is not good enough for many multimedia
+    applications. Developers may want to do the same in their Windows
+    applications if they think it’s a good idea for their application
+    use case, and depending on the Windows version they target. This is
+    not done automatically by GStreamer because on older Windows
+    versions (pre-Windows 10) this affects a global Windows setting and
+    also there’s a power consumption vs. performance trade-off that may
+    differ from application to application.
+
+-   dxgiscreencapsrc now supports resolution changes
+
+-   The wasapi2 audio plugin was rewritten and now has a higher rank
+    than the old wasapi plugin since it has a number of additional
+    features such as automatic stream routing, and no
+    known-but-hard-to-fix issues. The plugin is always built if the
+    Windows 10 SDK is available now.
+
+-   The wasapi device providers now detect and notify dynamic device
+    additions/removals
+
+-   d3d11screencapturesrc: new desktop capture element, including
+    GstDeviceProvider implementation to enumerate/select target monitors
+    for capture.
+
+-   Direct3D11/DXVA decoder now supports AV1 and MPEG2 codecs
+    (d3d11av1dec, d3d11mpeg2dec)
+
+-   VP9 decoding got more reliable and stable thanks to a newly written
+    codec parser
+
+-   Support for decoding interlaced H.264/AVC streams
+
+-   Hardware-accelerated video deinterlacing (d3d11deinterlace) and
+    video mixing (d3d11compositor)
+
+-   Video mixing with the Direct3D11 API (d3d11compositor)
+
+-   MediaFoundation API based hardware encoders gained the ability to
+    receive Direct3D11 textures as an input
+
+-   Seungha’s blog post “GStreamer ❤ Windows: A primer on the cool stuff
+    you’ll find in the 1.20 release” describes many of the
+    Windows-related improvements in more detail
 
 Linux
 
--   this section will be filled in in due course
+-   bluez: LDAC Bluetooth audio codec support in a2dpsink and avdtpsink,
+    as well as an LDAC RTP payloader (rtpldacpay) and an LDAC audio
+    encoder (ldacenc)
+
+-   kmssink: gained support for NV24, NV61, RGB16/BGR16 formats;
+    auto-detect NVIDIA Tegra driver
 
 Documentation improvements
 
--   this section will be filled in in due course
+-   hardware-accelerated GPU plugins will now no longer always list all
+    the element variants for all available GPUs, since those are
+    system-dependent and it’s confusing for users to see those in the
+    documentation just because the GStreamer developer who generated the
+    docs had multiple GPUs to play with at the time. Instead just show
+    the default elements.
 
-Possibly Breaking Changes
+Possibly Breaking and Other Noteworthy Behavioural Changes
+
+-   gst_parse_launch(), gst_parse_bin_from_description() and friends
+    will now error out when setting properties that don’t exist on
+    top-level bins. They were silently ignored before.
+
+-   The GstWebRTC library does not expose any objects anymore with
+    public fields. Instead properties have been added to replace that
+    functionality. If you are accessing such fields in your application,
+    switch to the corresponding properties.
 
--   FIXME: this section will be filled in in due course
--   MPEG-TS SCTE-35 API changes (FIXME: flesh out)
--   gst_parse_launch() and friends now error out on non-existing
-    properties on top-level bins where they would silently fail and
-    ignore those before.
 -   playbin and uridecodebin now emit the source-setup signal before the
     element is added to the bin and linked so that the source element is
     already configured before any scheduling query comes in, which is
-    useful for elements such as appsrc or giostreamsrc. (Merge Request)
+    useful for elements such as appsrc or giostreamsrc.
+
+-   The source element inside urisourcebin (used inside uridecodebin3
+    which is used inside playbin3) is no longer called "source". This
+    shouldn’t affect anyone hopefully, because there’s a "setup-source"
+    signal to configure the source element and no one should rely on
+    names of internal elements anyway.
+
+-   The vp8enc element now expects bps (bits per second) for the
+    "temporal-scalability-target-bitrate" property, which is consistent
+    with the "target-bitrate" property. Since additional configuration
+    is required with modern libvpx to make temporal scaling work anyway,
+    chances are that very few people will have been using this property
+
+-   vp8enc and vp9enc now default to “good quality” for the "deadline"
+    property rather then “best quality”. Having the deadline set to best
+    quality causes the encoder to be absurdly slow, most real-life users
+    will want the good quality tradeoff instead.
+
+-   The experimental GstTranscoder library API in gst-plugins-bad was
+    changed from a GObject signal-based notification mechanism to a
+    GstBus/message-based mechanism akin to GstPlayer/GstPlay.
+
+-   MPEG-TS SCTE-35 API: semantic change for SCTE-35 splice commands:
+    timestamps passed by the application should be in running time now,
+    since users of the API can’t really be expected to predict the local
+    PTS of the muxer.
+
+-   The GstContext used by souphttpsrc to share the session between
+    multiple element instances has changed. Previously it provided
+    direct access to the internal SoupSession object, now it only
+    provides access to an opaque, internal type. This change is
+    necessary because SoupSession is not thread-safe at all and can’t be
+    shared safely between arbitrary external code and souphttpsrc.
+
+-   Python bindings: GObject-introspection related Annotation fixes have
+    led to a case of a GstVideo.VideoInfo-related function signature
+    changing in the Python bindings (possibly one or two other cases
+    too). This is for a function that should never have been exposed in
+    the first place though, so the bindings are being updated to throw
+    an exception in that case, and the correct replacement API has been
+    added in form of an override.
 
 Known Issues
 
--   this section will be filled in in due course
-
--   There are a couple of known WebRTC-related regressions/blockers:
-
-    -   webrtc: DTLS setup with Chrome is broken
-    -   webrtcbin: First keyframe is usually lost
+-   nothing in particular at this point (but also see possibly breaking
+    changes section above)
 
 Contributors
 
--   this section will be filled in in due course
+Aaron Boxer, Adam Leppky, Adam Williamson, Alba Mendez, Alejandro
+González, Aleksandr Slobodeniuk, Alexander Vandenbulcke, Alex Ashley,
+Alicia Boya García, Andika Triwidada, Andoni Morales Alastruey, Andrew
+Wesie, Andrey Moiseev, Antonio Ospite, Antonio Rojas, Arthur Crippa
+Búrigo, Arun Raghavan, Ashley Brighthope, Axel Kellermann, Baek, Bastien
+Nocera, Bastien Reboulet, Benjamin Gaignard, Bing Song, Binh Truong,
+Biswapriyo Nath, Brad Hards, Brad Smith, Brady J. Garvin, Branko
+Subasic, Camilo Celis Guzman, Chris Bass, ChrisDuncanAnyvision, Chris
+White, Corentin Damman, Daniel Almeida, Daniel Knobe, Daniel Stone,
+david, David Fernandez, David Keijser, David Phung, Devarsh Thakkar,
+Dinesh Manajipet, Dmitry Samoylov, Dmitry Shusharin, Dominique Martinet,
+Doug Nazar, Ederson de Souza, Edward Hervey, Emmanuel Gil Peyrot,
+Enrique Ocaña González, Ezequiel Garcia, Fabian Orccon, Fabrice
+Fontaine, Fernando Jimenez Moreno, Florian Karydes, Francisco Javier
+Velázquez-García, François Laignel, Frederich Munch, Fredrik Pålsson,
+George Kiagiadakis, Georg Lippitsch, Göran Jönsson, Guido Günther,
+Guillaume Desmottes, Guiqin Zou, Haakon Sporsheim, Haelwenn (lanodan)
+Monnier, Haihao Xiang, Haihua Hu, Havard Graff, He Junyan, Helmut
+Januschka, Henry Wilkes, Hosang Lee, Hou Qi, Ignacio Casal Quinteiro,
+Igor Kovalenko, Ilya Kreymer, Imanol Fernandez, Jacek Tomaszewski, Jade
+Macho, Jakub Adam, Jakub Janků, Jan Alexander Steffens (heftig), Jan
+Schmidt, Jason Carrete, Jason Pereira, Jay Douglass, Jeongki Kim, Jérôme
+Laheurte, Jimmi Holst Christensen, Johan Sternerup, John Hassell, John
+Lindgren, John-Mark Bell, Jonathan Matthew, Jordan Petridis, Jose
+Quaresma, Julian Bouzas, Julien, Kai Uwe Broulik, Kasper Steensig
+Jensen, Kellermann Axel, Kevin Song, Khem Raj, Knut Inge Hvidsten, Knut
+Saastad, Kristofer Björkström, Lars Lundqvist, Lawrence Troup, Lim Siew
+Hoon, Lucas Stach, Ludvig Rappe, Luis Paulo Fernandes de Barros, Luke
+Yelavich, Mads Buvik Sandvei, Marc Leeman, Marco Felsch, Marek Vasut,
+Marian Cichy, Marijn Suijten, Marius Vlad, Markus Ebner, Mart Raudsepp,
+Matej Knopp, Mathieu Duponchelle, Matthew Waters, Matthieu De Beule,
+Mengkejiergeli Ba, Michael de Gans, Michael Olbrich, Michael Tretter,
+Michal Dzik, Miguel Paris, Mikhail Fludkov, mkba, Nazar Mokrynskyi,
+Nicholas Jackson, Nicola Murino, Nicolas Dufresne, Niklas Hambüchen,
+Nikolay Sivov, Nirbheek Chauhan, Olivier Blin, Olivier Crete, Olivier
+Crête, Paul Goulpié, Per Förlin, Peter Boba, P H, Philippe Normand,
+Philipp Zabel, Pieter Willem Jordaan, Piotrek Brzeziński, Rafał
+Dzięgiel, Rafostar, raghavendra, Raghavendra, Raju Babannavar, Raleigh
+Littles III, Randy Li, Randy Li (ayaka), Ratchanan Srirattanamet, Raul
+Tambre, reed.lawrence, Ricky Tang, Robert Rosengren, Robert Swain, Robin
+Burchell, Roman Sivriver, R S Nikhil Krishna, Ruben Gonzalez, Ruslan
+Khamidullin, Sanchayan Maity, Scott Moreau, Sebastian Dröge, Sergei
+Kovalev, Seungha Yang, Sid Sethupathi, sohwan.park, Sonny Piers, Staz M,
+Stefan Brüns, Stéphane Cerveau, Stephan Hesse, Stian Selnes, Stirling
+Westrup, Théo MAILLART, Thibault Saunier, Tim, Timo Wischer, Tim-Philipp
+Müller, Tim Schneider, Tobias Ronge, Tom Schoonjans, Tulio Beloqui,
+tyler-aicradle, U. Artie Eoff, Ung, Val Doroshchuk, VaL Doroshchuk,
+Víctor Manuel Jáquez Leal, Vivek R, Vivia Nikolaidou, Vivienne
+Watermeier, Vladimir Menshakov, Will Miller, Wim Taymans, Xabier
+Rodriguez Calvar, Xavier Claessens, Xℹ Ruoyao, Yacine Bandou, Yinhang
+Liu, youngh.lee, youngsoo.lee, yychao, Zebediah Figura, Zhang yuankun,
+Zhang Yuankun, Zhao, Zhao Zhili, , Aleksandar Topic, Antonio Ospite,
+Bastien Nocera, Benjamin Gaignard, Brad Hards, Carlos Falgueras García,
+Célestin Marot, Corentin Damman, Corentin Noël, Daniel Almeida, Daniel
+Knobe, Danny Smith, Dave Piché, Dmitry Osipenko, Fabrice Fontaine,
+fjmax, Florian Zwoch, Guillaume Desmottes, Haihua Hu, Heinrich Kruger,
+He Junyan, Jakub Adam, James Cowgill, Jan Alexander Steffens (heftig),
+Jean Felder, Jeongki Kim, Jiri Uncovsky, Joe Todd, Jordan Petridis,
+Krystian Wojtas, Marc-André Lureau, Marcin Kolny, Marc Leeman, Mark
+Nauwelaerts, Martin Reboredo, Mathieu Duponchelle, Matthew Waters,
+Mengkejiergeli Ba, Michael Gruner, Nicolas Dufresne, Nirbheek Chauhan,
+Olivier Crête, Philippe Normand, Rafał Dzięgiel, Ralf Sippl, Robert
+Mader, Sanchayan Maity, Sangchul Lee, Sebastian Dröge, Seungha Yang,
+Stéphane Cerveau, Teh Yule Kim, Thibault Saunier, Thomas Klausner, Timo
+Wischer, Tim-Philipp Müller, Tobias Reineke, Tomasz Andrzejak, Trung Do,
+Tyler Compton, Ung, Víctor Manuel Jáquez Leal, Vivia Nikolaidou, Wim
+Taymans, wngecn, Wonchul Lee, wuchang li, Xavier Claessens, Xi Ruoyao,
+Yoshiharu Hirose, Zhao,
 
 … and many others who have contributed bug reports, translations, sent
 suggestions or helped testing.
@@ -307,7 +1863,7 @@ the git 1.20 branch, which will be a stable branch.
 
 1.20.0
 
-1.20.0 is scheduled to be released around October/November 2021.
+1.20.0 is scheduled to be released around early February 2022.
 
 Schedule for 1.22
 
@@ -315,7 +1871,9 @@ Our next major feature release will be 1.22, and 1.21 will be the
 unstable development version leading up to the stable 1.22 release. The
 development of 1.21/1.22 will happen in the git main branch.
 
-The plan for the 1.22 development cycle is yet to be confirmed.
+The plan for the 1.22 development cycle is yet to be confirmed. Assuming
+no major project-wide reorganisations in the 1.22 cycle we might try and
+aim for a release around August 2022.
 
 1.22 will be backwards-compatible to the stable 1.20, 1.18, 1.16, 1.14,
 1.12, 1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series.
@@ -323,6 +1881,7 @@ The plan for the 1.22 development cycle is yet to be confirmed.
 ------------------------------------------------------------------------
 
 These release notes have been prepared by Tim-Philipp Müller with
-contributions from …
+contributions from Matthew Waters, Nicolas Dufresne, Nirbheek Chauhan,
+Sebastian Dröge and Seungha Yang.
 
 License: CC BY-SA 4.0
diff --git a/subprojects/gst-libav/RELEASE b/subprojects/gst-libav/RELEASE
index b5cf87f29b..0349d12b7d 100644
--- a/subprojects/gst-libav/RELEASE
+++ b/subprojects/gst-libav/RELEASE
@@ -1,4 +1,4 @@
-This is GStreamer gst-libav 1.19.3.
+This is GStreamer gst-libav 1.19.90.
 
 GStreamer 1.19 is the development branch leading up to the next major
 stable version which will be 1.20.
diff --git a/subprojects/gst-libav/gst-libav.doap b/subprojects/gst-libav/gst-libav.doap
index 086244f593..2bd3df03a2 100644
--- a/subprojects/gst-libav/gst-libav.doap
+++ b/subprojects/gst-libav/gst-libav.doap
@@ -34,6 +34,16 @@ colorspace conversion elements.
 
  <release>
   <Version>
+   <revision>1.19.90</revision>
+   <branch>main</branch>
+   <name></name>
+   <created>2022-01-28</created>
+   <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-libav/gst-libav-1.19.90.tar.xz" />
+  </Version>
+ </release>
+
+ <release>
+  <Version>
    <revision>1.19.3</revision>
    <branch>main</branch>
    <name></name>
diff --git a/subprojects/gst-libav/meson.build b/subprojects/gst-libav/meson.build
index 296f57ee59..0d6b77d43e 100644
--- a/subprojects/gst-libav/meson.build
+++ b/subprojects/gst-libav/meson.build
@@ -1,5 +1,5 @@
 project('gst-libav', 'c', 'cpp',
-  version : '1.19.3.1',
+  version : '1.19.90',
   meson_version : '>= 0.59',
   default_options : [ 'warning_level=1',
                       'buildtype=debugoptimized' ])
diff --git a/subprojects/gst-omx/ChangeLog b/subprojects/gst-omx/ChangeLog
index 59ce16a248..730011d05a 100644
--- a/subprojects/gst-omx/ChangeLog
+++ b/subprojects/gst-omx/ChangeLog
@@ -1,3 +1,18 @@
+=== release 1.19.90 ===
+
+2022-01-28 14:28:35 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* NEWS:
+	* RELEASE:
+	* gst-omx.doap:
+	* meson.build:
+	  Release 1.19.90
+
+2022-01-28 14:28:28 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* ChangeLog:
+	  Update ChangeLogs for 1.19.90
+
 2022-01-05 02:07:59 +0530  Nirbheek Chauhan <nirbheek@centricular.com>
 
 	* meson.build:
diff --git a/subprojects/gst-omx/NEWS b/subprojects/gst-omx/NEWS
index e9f5227c15..1a512c1848 100644
--- a/subprojects/gst-omx/NEWS
+++ b/subprojects/gst-omx/NEWS
@@ -1,23 +1,19 @@
 GStreamer 1.20 Release Notes
 
 GStreamer 1.20 has not been released yet. It is scheduled for release in
-November 2021.
+late January / early February 2022.
 
 1.19.x is the unstable development version that is being developed in
-the git main branch and which will eventually result in 1.20, and 1.19.3
-is the current development release in that series
-
-It is expected that feature freeze will be in early October 2021,
-followed by one or two 1.19.9x pre-releases and the new 1.20 stable
-release around October/November 2021.
+the git main branch and which will eventually result in 1.20, and
+1.19.90 is the first release candidate in that series (1.20rc1).
 
 1.20 will be backwards-compatible to the stable 1.18, 1.16, 1.14, 1.12,
-1.10, 1.8, 1.6,, 1.4, 1.2 and 1.0 release series.
+1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series.
 
 See https://gstreamer.freedesktop.org/releases/1.20/ for the latest
 version of this document.
 
-Last updated: Monday 1 November 2021, 01:00 UTC (log)
+Last updated: Wednesday 26 January 2022, 01:00 UTC (log)
 
 Introduction
 
@@ -30,25 +26,971 @@ fixes and other improvements.
 
 Highlights
 
--   this section will be completed in due course
+-   Development in GitLab was switched to a single git repository
+    containing all the modules
+-   GstPlay: new high-level playback library, replaces GstPlayer
+-   WebM Alpha decoding support
+-   Encoding profiles can now be tweaked with additional
+    application-specified element properties
+-   Compositor: multi-threaded video conversion and mixing
+-   RTP header extensions: unified support in RTP depayloader and
+    payloader base classes
+-   SMPTE 2022-1 2-D Forward Error Correction support
+-   Smart encoding (passthrough) support for VP8, VP9, H.265 in
+    encodebin and transcodebin
+-   Runtime compatibility support for libsoup2 and libsoup3 (libsoup3
+    support experimental)
+-   Video decoder subframe support
+-   Video decoder automatic packet-loss, data corruption, and keyframe
+    request handling for RTP / WebRTC / RTSP
+-   MP4 and Matroska muxers now support profile/level/resolution changes
+    for H264/H265 input streams (i.e. codec data changing on the fly)
+-   MP4 muxing mode that initially creates a fragmented mp4 which is
+    converted to a regular mp4 on EOS
+-   Audio support for the WebKit Port for Embedded (WPE) web page source
+    element
+-   CUDA based video color space convert and rescale elements and
+    upload/download elements
+-   NVIDIA memory:NVMM support for OpenGL glupload and gldownload
+    elements
+-   Many WebRTC improvements
+-   The new VA-API plugin implemention fleshed out with more decoders
+    and new postproc elements
+-   AppSink API to retrieve events in addition to buffers and buffer
+    lists
+-   AppSrc gained more configuration options for the internal queue
+    (leakiness, limits in buffers and time, getters to read current
+    levels)
+-   Updated Rust bindings and many new Rust plugins
+-   Improved support for custom minimal GStreamer builds
+-   Support build against FFmpeg 5.0
+-   Linux Stateless CODEC support gained MPEG2 and VP9
+-   Windows Direct3D11/DXVA decoder gained AV1 and MPEG2 support
+-   Lots of new plugins, features, performance improvements and bug
+    fixes
 
 Major new features and changes
 
 Noteworthy new features and API
 
--   this section will be filled in in due course
+-   gst_element_get_request_pad() has been deprecated in favour of the
+    newly-added gst_element_request_pad_simple() which does the exact
+    same thing but has a less confusing name that hopefully makes clear
+    that the function request a new pad rather than just retrieves an
+    already-existing request pad.
+
+Development in GitLab was switched to a single git repository containing all the modules
+
+The GStreamer multimedia framework is a set of libraries and plugins
+split into a number of distinct modules which are released independently
+and which have so far been developed in separate git repositories in
+freedesktop.org GitLab.
+
+In addition to these separate git repositories there was a gst-build
+module that would use the Meson build systems’s subproject feature to
+download each individual module and then build everything in one go. It
+would also provide an uninstalled development environment that made it
+easy to work on GStreamer and use or test versions other than the
+system-installed GStreamer version.
+
+All of these modules have now (as of 28 September 2021) been merged into
+a single git repository (“Mono repository” or “monorepo”) which should
+simplify development workflows and continuous integration, especially
+where changes need to be made to multiple modules at once.
+
+This mono repository merge will primarily affect GStreamer developers
+and contributors and anyone who has workflows based on the GStreamer git
+repositories.
+
+The Rust bindings and Rust plugins modules have not been merged into the
+mono repository at this time because they follow a different release
+cycle.
+
+The mono repository lives in the existing GStreamer core git repository
+in GitLab in the new main branch and all future development will happen
+on this branch.
+
+Modules will continue to be released as separate tarballs.
+
+For more details, please see the GStreamer mono repository FAQ.
+
+GstPlay: new high-level playback library replacing GstPlayer
+
+-   GstPlay is a new high-level playback library that replaces the older
+    GstPlayer API. It is basically the same API as GstPlayer but
+    refactored to use bus messages for application notifications instead
+    of GObject signals. There is still a signal adapter object for those
+    who prefer signals. Since the existing GstPlayer API is already in
+    use in various applications, it didn’t seem like a good idea to
+    break it entirely. Instead a new API was added, and it is expected
+    that this new GstPlay API will be moved to gst-plugins-base in
+    future.
+
+-   The existing GstPlayer API is scheduled for deprecation and will be
+    removed at some point in the future (e.g. in GStreamer 1.24), so
+    application developers are urged to migrate to the new GstPlay API
+    at their earliest convenience.
+
+WebM alpha decoding
+
+-   Implement WebM alpha decoding (VP8/VP9 with alpha), which required
+    support and additions in various places. This is supported both with
+    software decoders and hardware-accelerated decoders.
+
+-   VP8/VP9 don’t support alpha components natively in the codec, so the
+    way this is implemented in WebM is by encoding the alpha plane with
+    transparency data as a separate VP8/VP9 stream. Inside the WebM
+    container (a variant of Matroska) this is coded as a single video
+    track with the “normal” VP8/VP9 video data making up the main video
+    data and each frame of video having an encoded alpha frame attached
+    to it as extra data ("BlockAdditional").
+
+-   matroskademux has been extended extract this per-frame alpha side
+    data and attach it in form of a GstVideoCodecAlphaMeta to the
+    regular video buffers. Note that this new meta is specific to this
+    VP8/VP9 alpha support and can’t be used to just add alpha support to
+    other codecs that don’t support it. Lastly, matroskademux also
+    advertises the fact that the streams contain alpha in the caps.
+
+-   The new codecalpha plugin contains various bits of infrastructure to
+    support autoplugging and debugging:
+
+    -   codecalphademux splits out the alpha stream from the metas on
+        the regular VP8/VP9 buffers
+    -   alphacombine takes two decoded raw video streams (one alpha, one
+        the regular video) and combines it into a video stream with
+        alpha
+    -   vp8alphadecodebin + vp9alphadecodebin are wrapper bins that use
+        the regular vp8dec and vp9dec software decoders to decode
+        regular and alpha streams and combine them again. To decodebin
+        these look like regular decoders which ju
+    -   The V4L2 CODEC plugin has stateless VP8/VP9 decoders that can
+        decode both alpha and non-alpha stream with a single decoder
+        instance
+
+-   A new AV12 video format was added which is basically NV12 with an
+    alpha plane, which is more convenient for many hardware-accelerated
+    decoders.
+
+-   Watch Nicolas Dufresne’s LCA 2022 talk “Bringing WebM Alpha support
+    to GStreamer” for all the details and a demo.
+
+RTP Header Extensions Base Class and Automatic Header Extension Handling in RTP Payloaders and Depayloaders
+
+-   RTP Header Extensions are specified in RFC 5285 and provide a way to
+    add small pieces of data to RTP packets in between the RTP header
+    and the RTP payload. This is often used for per-frame metadata,
+    extended timestamps or other application-specific extra data. There
+    are several commonly-used extensions specified in various RFCs, but
+    senders are free to put any kind of data in there, as long as sender
+    and receiver both know what that data is. Receivers that don’t know
+    about the header extensions will just skip the extra data without
+    ever looking at it. These header extensions can often be combined
+    with any kind of payload format, so may need to be supported by many
+    RTP payloader and depayloader elements.
+
+-   Inserting and extracting RTP header extension data has so far been a
+    bit inconvenient in GStreamer: There are functions to add and
+    retrieve RTP header extension data from RTP packets, but nothing
+    works automatically, even for common extensions. People would have
+    to do the insertion/extraction either in custom elements
+    before/after the RTP payloader/depayloader, or inside pad probes,
+    which isn’t very nice.
+
+-   This release adds various pieces of new infrastructure for generic
+    RTP header extension handling, as well as some implementations for
+    common extensions:
+
+    -   GstRTPHeaderExtension is a new helper base class for reading and
+        writing RTP header extensions. Nominally this subclasses
+        GstElement, but only so these extensions are stored in the
+        registry where they can be looked up by URI or name. They don’t
+        have pads and don’t get added to the pipeline graph as an
+        element.
+
+    -   "add-extension" and "clear-extension" action signals on RTP
+        payloaders and depayloaders for manual extension management
+
+    -   The "request-extension" signal will be emitted if an extension
+        is encountered that requires explicit mapping by the application
+
+    -   new "auto-header-extension" property on RTP payloaders and
+        depayloaders for automatic handling of known header extensions.
+        This is enabled by default. The extensions must be signalled via
+        caps / SDP.
+
+    -   RTP header extension implementations:
+
+        -   rtphdrextclientaudiolevel: Client-to-Mixer Audio Level
+            Indication (RFC 6464) (also see below)
+        -   rtphdrextcolorspace: Color Space extension, extends RTP
+            packets with color space and high dynamic range (HDR)
+            information
+        -   rtphdrexttwcc: Transport Wide Congestion Control support
+
+-   gst_rtp_buffer_remove_extension_data() is a new helper function to
+    remove an RTP header extension from an RTP buffer
+
+-   The existing gst_rtp_buffer_set_extension_data() now also supports
+    shrinking the extension data in size
+
+AppSink and AppSrc improvements
+
+-   appsink: new API to pull events out of appsink in addition to
+    buffers and buffer lists.
+
+    There was previously no way for users to receive incoming events
+    from appsink properly serialised with the data flow, even if they
+    are serialised events. The reason for that is that the only way to
+    intercept events was via a pad probe on the appsink sink pad, but
+    there is also internal queuing inside of appsink, so it’s difficult
+    to ascertain the right order of everything in all cases.
+
+    There is now a new "new-serialized-event" signal which will be
+    emitted when there’s a new event pending (just like the existing
+    "new-sample" signal). The "emit-signals" property must be set to
+    TRUE in order to activate this (but it’s also fine to just pull from
+    the application thread without using the signals).
+
+    gst_app_sink_pull_object() and gst_app_sink_try_pull_object() can be
+    used to pull out either an event or a new sample carrying a buffer
+    or buffer list, whatever is next in the queue.
+
+    EOS events will be filtered and will not be returned. EOS handling
+    can be done the ususal way, same as with _pull_sample().
+
+-   appsrc: allow configuration of internal queue limits in time and
+    buffers and add leaky mode.
+
+    There is internal queuing inside appsrc so the application thread
+    can push data into the element which will then be picked up by the
+    source element’s streaming thread and pushed into the pipeline from
+    that streaming thread. This queue is unlimited by default and until
+    now it was only possible to set a maximum size limit in bytes. When
+    that byte limit is reached, the pushing thread (application thread)
+    would be blocked until more space becomes available.
+
+    A limit in bytes is not particularly useful for many use cases, so
+    now it is possible to also configure limits in time and buffers
+    using the new "max-time" and "max-buffers" properties. Of course
+    there are also matching new read-only"current-level-buffers" and
+    "current-level-time properties" properties to query the current fill
+    level of the internal queue in time and buffers.
+
+    And as if that wasn’t enough the internal queue can also be
+    configured as leaky using the new "leaky-type" property. That way
+    when the queue is full the application thread won’t be blocked when
+    it tries to push in more data, but instead either the new buffer
+    will be dropped or the oldest data in the queue will be dropped.
+
+Better string serialization of nested GstCaps and GstStructures
+
+-   New string serialisation format for structs and caps that can handle
+    nested structs and caps properly by using brackets to delimit nested
+    items (e.g. some-struct, some-field=[nested-struct, nested=true]).
+    Unlike the default format the new variant can also support more than
+    one level of nesting. For backwards-compatibility reasons the old
+    format is still output by default when serialising caps and structs
+    using the existing API. The new functions gst_caps_serialize() and
+    gst_structure_serialize() can be used to output strings in the new
+    format.
+
+Convenience API for custom GstMetas
+
+-   New convenience API to register and create custom GstMetas:
+    gst_meta_register_custom() and gst_buffer_add_custom_meta(). Such
+    custom meta is backed by a GstStructure and does not require that
+    users of the API expose their GstMeta implementation as public API
+    for other components to make use of it. In addition, it provides a
+    simpler interface by ignoring the impl vs. api distinction that the
+    regular API exposes. This new API is meant to be the meta
+    counterpart to custom events and messages, and to be more convenient
+    than the lower-level API when the absolute best performance isn’t a
+    requirement. The reason it’s less performant than a “proper” meta is
+    that a proper meta is just a C struct in the end whereas this goes
+    through the GstStructure API which has a bit more overhead, which
+    for most scenarios is negligible however. This new API is useful for
+    experimentation or proprietary metas, but also has some limitations:
+    it can only be used if there’s a single producer of these metas;
+    it’s not allowed to register the same custom meta multiple times or
+    from multiple places.
+
+Additional Element Properties on Encoding Profiles
+
+-   GstEncodingProfile: The new "element-properties" and
+    gst_encoding_profile_set_element_properties() API allows
+    applications to set additional element properties on encoding
+    profiles to configure muxers and encoders. So far the encoding
+    profile template was the only place where this could be specified,
+    but often what applications want to do is take a ready-made encoding
+    profile shipped by GStreamer or the application and then tweak the
+    settings on top of that, which is now possible with this API. Since
+    applications can’t always know in advance what encoder element will
+    be used in the end, it’s even possible to specify properties on a
+    per-element basis.
+
+    Encoding Profiles are used in the encodebin, transcodebin and
+    camerabin elements and APIs to configure output formats (containers
+    and elementary streams).
+
+Audio Level Indication Meta for RFC 6464
+
+-   New GstAudioLevelMeta containing Audio Level Indication as per RFC
+    6464
+
+-   The level element has been updated to add GstAudioLevelMeta on
+    buffers if the "audio-level-meta" property is set to TRUE. This can
+    then in turn be picked up by RTP payloaders to signal the audio
+    level to receivers through RTP header extensions (see above).
+
+-   New Client-to-Mixer Audio Level Indication (RFC6464) RTP Header
+    Extension which should be automatically created and used by RTP
+    payloaders and depayloaders if their "auto-header-extension"
+    property is enabled and if the extension is part of the RTP caps.
+
+Automatic packet loss, data corruption and keyframe request handling for video decoders
+
+-   The GstVideoDecoder base class has gained various new APIs to
+    automatically handle packet loss and data corruption better by
+    default, especially in RTP, RTSP and WebRTC streaming scenarios, and
+    to give subclasses more control about how they want to handle
+    missing data:
+
+    -   Video decoder subclasses can mark output frames as corrupted via
+        the new GST_VIDEO_CODEC_FRAME_FLAG_CORRUPTED flag
+
+    -   A new "discard-corrupted-frames" property allows applications to
+        configure decoders so that corrupted frames are directly
+        discarded instead of being forwarded inside the pipeline. This
+        is a replacement for the "output-corrupt" property of the FFmpeg
+        decoders.
+
+    -   RTP depayloaders can now signal to decoders that data is missing
+        when sending GAP events for lost packets. GAP events can be sent
+        for various reason in a GStreamer pipeline. Often they are just
+        used to let downstream elements know that there isn’t a buffer
+        available at the moment, so downstream elements can move on
+        instead of waiting for one. They are also sent by RTP
+        depayloaders in the case that packets are missing, however, and
+        so far a decoder was not able to differentiate the two cases.
+        This has been remedied now: GAP events can be decorated with
+        gst_event_set_gap_flags() and GST_GAP_FLAG_MISSING_DATA to let
+        decoders now what happened, and decoders can then use that in
+        some cases to handle missing data better.
+
+    -   The GstVideoDecoder::handle_missing_data vfunc was added to
+        inform subclasses about packet loss or missing data and let them
+        handle it in their own way if they like.
+
+    -   gst_video_decoder_set_needs_sync_point() lets subclasses signal
+        that they need the stream to start with a sync point. If
+        enabled, the base class will discard all non-sync point frames
+        in the beginning and after a flush and does not pass them to the
+        subclass. Furthermore, if the first frame is not a sync point,
+        the base class will try and request a sync frame from upstream
+        by sending a force-key-unit event (see next items).
+
+    -   New "automatic-request-sync-points" and
+        "automatic-request-sync-point-flags" properties to automatically
+        request sync points when needed, e.g. on packet loss or if the
+        first frame is not a keyframe. Applications may want to enable
+        this on decoders operating in e.g. RTP/WebRTC/RTSP receiver
+        pipelines.
+
+    -   The new "min-force-key-unit-interval" property can be used to
+        ensure there’s a minimal interval between keyframe requests to
+        upstream (and/or the sender) and we’re not flooding the sender
+        with key unit requests.
+
+    -   gst_video_decoder_request_sync_point() allows subclasses to
+        request a new sync point (e.g. if they choose to do their own
+        missing data handling). This will still honour the
+        "min-force-key-unit-interval" property if set.
+
+Improved support for custom minimal GStreamer builds
+
+-   Element registration and registration of other plugin features
+    inside plugin init functions has been improved in order to
+    facilitate minimal custom GStreamer builds.
+
+-   A number of new macros have been added to declare and create
+    per-element and per-pluginfeature register functions in all plugins,
+    and then call those from the per-plugin plugin_init functions:
+
+    -   GST_ELEMENT_REGISTER_DEFINE,
+        GST_DEVICE_PROVIDER_REGISTER_DEFINE,
+        GST_DYNAMIC_TYPE_REGISTER_DEFINE, GST_TYPE_FIND_REGISTER_DEFINE
+        for the actual registration call with GStreamer
+    -   GST_ELEMENT_REGISTER, GST_DEVICE_PROVIDER_REGISTER,
+        GST_DYNAMIC_TYPE_REGISTER, GST_PLUGIN_STATIC_REGISTER,
+        GST_TYPE_FIND_REGISTER to call the registration function defined
+        by the REGISTER_DEFINE macro
+    -   GST_ELEMENT_REGISTER_DECLARE,
+        GST_DEVICE_PROVIDER_REGISTER_DECLARE,
+        GST_DYNAMIC_TYPE_REGISTER_DECLARE,
+        GST_TYPE_FIND_REGISTER_DECLARE to declare the registration
+        function defined by the REGISTER_DEFINE macro
+    -   and various variants for advanced use cases.
+
+-   This means that applications can call the per-element and
+    per-pluginfeature registration functions for only the elements they
+    need instead of registering plugins as a whole with all kinds of
+    elements that may not be required (e.g. encoder and decoder instead
+    of just decoder). In case of static linking all unused functions and
+    their dependencies would be removed in this case by the linker,
+    which helps minimise binary size for custom builds.
+
+-   gst_init() will automatically call a gst_init_static_plugins()
+    function if one exists.
+
+-   See the GStreamer static build documentation and Stéphane’s blog
+    post Generate a minimal GStreamer build, tailored to your needs for
+    more details.
 
 New elements
 
--   this section will be filled in in due course
+-   New aesdec and aesenc elements for AES encryption and decryption in
+    a custom format.
+
+-   New encodebin2 element with dynamic/sometimes source pads in order
+    to support the option of doing the muxing outside of encodebin,
+    e.g. in combination with a splitmuxsink.
+
+-   New fakeaudiosink and videocodectestsink elements for testing and
+    debugging (see below for more details)
+
+-   rtpisacpay, rtpisacdepay: new RTP payloader and depayloader for iSAC
+    audio codec
+
+-   rtpst2022-1-fecdec, rtpst2022-1-fecenc: new elements providing SMPTE
+    2022-1 2-D Forward Error Correction. More details in Mathieu’s blog
+    post.
+
+-   isac: new plugin wrapping the Internet Speech Audio Codec reference
+    encoder and decoder from the WebRTC project.
+
+-   asio: plugin for Steinberg ASIO (Audio Streaming Input/Output) API
+
+-   gssrc, gssink: add source and sink for Google Cloud Storage
+
+-   onnx: new plugin to apply ONNX neural network models to video
+
+-   openaptx: aptX and aptX-HD codecs using libopenaptx (v0.2.0)
+
+-   qroverlay, debugqroverlay: new elements that allows overlaying data
+    on top of video in form of a QR code
+
+-   cvtracker: new OpenCV-based tracker element
+
+-   av1parse, vp9parse: new parsers for AV1 and VP9 video
+
+-   va: work on the new VA-API plugin implementation for
+    hardware-accelerated video decoding and encoding has continued at
+    pace, with various new decoders and filters having joined the
+    initial vah264dec:
+
+    -   vah265dec: VA-API H.265 decoder
+    -   vavp8dec: VA-API VP8 decoder
+    -   vavp9dec: VA-API VP9 decoder
+    -   vaav1dec: VA-API AV1 decoder
+    -   vampeg2dec: VA-API MPEG-2 decoder
+    -   vadeinterlace: : VA-API deinterlace filter
+    -   vapostproc: : VA-API postproc filter (color conversion,
+        resizing, cropping, color balance, video rotation, skin tone
+        enhancement, denoise, sharpen)
+
+    See Víctor’s blog post “GstVA in GStreamer 1.20” for more details
+    and what’s coming up next.
+
+-   vaapiav1dec: new AV1 decoder element (in gstreamer-vaapi)
+
+-   msdkav1dec: hardware-accelerated AV1 decoder using the Intel Media
+    SDK / oneVPL
+
+-   nvcodec plugin for NVIDIA NVCODEC API for hardware-accelerated video
+    encoding and decoding:
+
+    -   cudaconvert, cudascale: new CUDA based video color space convert
+        and rescale elements
+    -   cudaupload, cudadownload: new helper elements for memory
+        transfer between CUDA and system memory spaces
+    -   nvvp8sldec, nvvp9sldec: new GstCodecs-based VP8/VP9 decoders
+
+-   Various new hardware-accelerated elements for Windows:
+
+    -   d3d11screencapturesrc: new desktop capture element, including a
+        GstDeviceProvider implementation to enumerate/select target
+        monitors for capture.
+    -   d3d11av1dec and d3d11mpeg2dec: AV1 and MPEG-2 decoders
+    -   d3d11deinterlace: deinterlacing filter
+    -   d3d11compositor: video composing element
+    -   see Windows section below for more details
+
+-   new Rust plugins:
+
+    -   audiornnoise: Removes noise from an audio stream
+    -   awstranscribeparse: Parses AWS audio transcripts into timed text
+        buffers
+    -   ccdetect: Detects if valid closed captions are present in a
+        closed captions stream
+    -   cea608tojson: Converts CEA-608 Closed Captions to a JSON
+        representation
+    -   cmafmux: CMAF fragmented MP4 muxer
+    -   dashmp4mux: DASH fragmented MP4 muxer
+    -   isofmp4mux: ISO fragmented MP4 muxer
+    -   ebur128level: EBU R128 Loudness Level Measurement
+    -   ffv1dec: FFV1 video decoder
+    -   gtk4paintablesink: GTK4 video sink, which provides a
+        GdkPaintable that can be rendered in various widgets
+    -   hlssink3: HTTP Live Streaming sink
+    -   hrtfrender: Head-Related Transfer Function (HRTF) renderer
+    -   hsvdetector: HSV colorspace detector
+    -   hsvfilter: HSV colorspace filter
+    -   jsongstenc: Wraps buffers containing any valid top-level JSON
+        structures into higher level JSON objects, and outputs those as
+        ndjson
+    -   jsongstparse: Parses ndjson as output by jsongstenc
+    -   jsontovtt: converts JSON to WebVTT subtitles
+    -   regex: Applies regular expression operations on text
+    -   roundedcorners: Adds rounded corners to video
+    -   spotifyaudiosrc: Spotify source
+    -   textahead: Display upcoming text buffers ahead (e.g. for
+        Karaoke)
+    -   transcriberbin: passthrough bin that transcribes raw audio to
+        closed captions using awstranscriber and puts the captions as
+        metas onto the video
+    -   tttojson: Converts timed text to a JSON representation
+    -   uriplaylistbin: Playlist source bin
+    -   webpdec-rs: WebP image decoder with animation support
+
+-   New plugin codecalpha with elements to assist with WebM Alpha
+    decoding
+
+    -   codecalphademux: Split stream with GstVideoCodecAlphaMeta into
+        two streams
+    -   alphacombine: Combine two raw video stream (I420 or NV12) as one
+        stream with alpha channel (A420 or AV12)
+    -   vp8alphadecodebin: A bin to handle software decoding of VP8 with
+        alpha
+    -   vp9alphadecodebin: A bin to handle software decoding of VP9 with
+        alpha
+
+-   New hardware accelerated elements for Linux:
+
+    -   v4l2slmpeg2dec: Support for Linux Stateless MPEG2 decoders
+    -   v4l2slvp9dec: Support for Linux Stateless VP9 decoders
+    -   v4l2slvp8alphadecodebin: Support HW accelerated VP8 with alpha
+        layer decoding
+    -   v4l2slvp9alphadecodebin: Support HW accelerated VP9 with alpha
+        layer decoding
 
 New element features and additions
 
--   this section will be filled in in due course
+-   assrender: handle more font mime types; better interaction with
+    matroskademux for embedded fonts
+
+-   audiobuffersplit: Add support for specifying output buffer size in
+    bytes (not just duration)
+
+-   audiolatency: new "samplesperbuffer" property so users can configure
+    the number of samples per buffer. The default value is 240 samples
+    which is equivalent to 5ms latency with a sample rate of 48000,
+    which might be larger than actual buffer size of audio capture
+    device.
+
+-   audiomixer, audiointerleave, GstAudioAggregator: now keep a count of
+    samples that are dropped or processed as statistic and can be made
+    to post QoS messages on the bus whenever samples are dropped by
+    setting the "qos-messages" property on input pads.
+
+-   audiomixer, compositor: improved handling of new inputs added at
+    runtime. New API was added to the GstAggregator base class to allow
+    subclasses to opt into an aggregation mode where inactive pads are
+    ignored when processing input buffers
+    (gst_aggregator_set_ignore_inactive_pads(),
+    gst_aggregator_pad_is_inactive()). An “inactive pad” in this context
+    is a pad which, in live mode, hasn’t yet received a first buffer,
+    but has been waited on at least once. What would happen usually in
+    this case is that the aggregator would wait for data on this pad
+    every time, up to the maximum configured latency. This would
+    inadvertently push mixer elements in live mode to the configured
+    latency envelope and delay processing when new inputs are added at
+    runtime until these inputs have actually produced data. This is
+    usually undesirable. With this new API, new inputs can be added
+    (requested) and configured and they won’t delay the data processing.
+    Applications can opt into this new behaviour by setting the
+    "ignore-inactive-pads" property on compositor, audiomixer or other
+    GstAudioAggregator-based elements.
+
+-   cccombiner: implement “scheduling” of captions. So far cccombiner’s
+    behaviour was essentially that of a funnel: it strictly looked at
+    input timestamps to associate together video and caption buffers.
+    Now it will try to smoothly schedule caption buffers in order to
+    have exactly one per output video buffer. This might involve
+    rewriting input captions, for example when the input is CDP then
+    sequence counters are rewritten, time codes are dropped and
+    potentially re-injected if the input video frame had a time code
+    meta. This can also lead to the input drifting from synchronization,
+    when there isn’t enough padding in the input stream to catch up. In
+    that case the element will start dropping old caption buffers once
+    the number of buffers in its internal queue reaches a certain limit
+    (configurable via the "max-scheduled" property). The new original
+    funnel-like behaviour can be restored by setting the "scheduling"
+    property to FALSE.
+
+-   ccconverter: new "cdp-mode" property to specify which sections to
+    include in CDP packets (timecode, CC data, service info). Various
+    software, including ffmpeg’s Decklink support, fails parsing CDP
+    packets that contain anything but CC data in the CDP packets.
+
+-   clocksync: new "sync-to-first" property for automatic timestamp
+    offset setup: if set clocksync will set up the "ts-offset" value
+    based on the first buffer and the pipeline’s running time when the
+    first buffer arrived. The newly configured "ts-offset" in this case
+    would be the value that allows outputting the first buffer without
+    waiting on the clock. This is useful for example to feed a non-live
+    input into an already-running pipeline.
+
+-   compositor:
+
+    -   multi-threaded input conversion and compositing. Set the
+        "max-threads" property to activate this.
+    -   new "sizing-policy" property to support display aspect ratio
+        (DAR)-aware scaling. By default the image is scaled to fill the
+        configured destination rectangle without padding and without
+        keeping the aspect ratio. With sizing-policy=keep-aspect-ratio
+        the input image is scaled to fit the destination rectangle
+        specified by GstCompositorPad:{xpos, ypos, width, height}
+        properties preserving the aspect ratio. As a result, the image
+        will be centered in the destination rectangle with padding if
+        necessary.
+    -   new "zero-size-is-unscaled" property on input pads. By default
+        pad width=0 or pad height=0 mean that the stream should not be
+        scaled in that dimension. But if the "zero-size-is-unscaled"
+        property is set to FALSE a width or height of 0 is instead
+        interpreted to mean that the input image on that pad should not
+        be composited, which is useful when creating animations where an
+        input image is made smaller and smaller until it disappears.
+    -   improved handling of new inputs at runtime via
+        "ignore-inactive-pads"property (see above for details)
+    -   allow output format with alpha even if none of the inputs have
+        alpha (also glvideomixer and other GstVideoAggregator
+        subclasses)
+
+-   dashsink: add h265 codec support and signals for allowing custom
+    playlist/fragment output
+
+-   decodebin3:
+
+    -   improved decoder selection, especially for hardware decoders
+    -   make input activation “atomic” when adding inputs dynamically
+    -   better interleave handling: take into account decoder latency
+        for interleave size
+
+-   decklink:
+
+    -   Updated DeckLink SDK to 11.2 to support DeckLink 8K Pro
+    -   decklinkvideosrc:
+        -   More accurate and stable capture timestamps: use the
+            hardware reference clock time when the frame was finished
+            being captured instead of a clock time much further down the
+            road.
+        -   Automatically detect widescreen vs. normal NTSC/PAL
+
+-   encodebin:
+
+    -   add “smart encoding” support for H.265, VP8 and VP9 (i.e. only
+        re-encode where needed and otherwise pass through encoded video
+        as-is).
+    -   H264/H265 smart encoding improvements: respect user-specified
+        stream-format, but if not specified default to avc3/hvc1 with
+        in-band SPS/PPS/VPS signalling for more flexibility.
+    -   new encodebin2 element with dynamic/sometimes source pads in
+        order to support the option of doing the muxing outside of
+        encodebin, e.g. in combination with splitmuxsink.
+    -   add APIs to set element properties on encoding profiles (see
+        below)
+
+-   errorignore: new "ignore-eos" property to also ignore FLOW_EOS from
+    downstream elements
+
+-   giosrc: add support for growing source files: applications can
+    specify that the underlying file being read is growing by setting
+    the "is-growing" property. If set, the source won’t EOS when it
+    reaches the end of the file, but will instead start monitoring it
+    and will start reading data again whenever a change is detected. The
+    new "waiting-data" and "done-waiting-data" signals keep the
+    application informed about the current state.
+
+-   gtksink, gtkglsink:
+
+    -   scroll event support: forwarded as navigation events into the
+        pipeline
+    -   "video-aspect-ratio-override" property to force a specific
+        aspect ratio
+    -   "rotate-method" property and support automatic rotation based on
+        image tags
+
+-   identity: new "stats" property allows applications to retrieve the
+    number of bytes and buffers that have passed through so far.
+
+-   interlace: add support for more formats, esp 10-bit, 12-bit and
+    16-bit ones
+
+-   jack: new "low-latency" property for automatic latency-optimized
+    setting and "port-names" property to select ports explicitly
+
+-   jpegdec: support output conversion to RGB using libjpeg-turbo (for
+    certain input files)
+
+-   line21dec:
+
+    -   "mode" property to control whether and how detected closed
+        captions should be inserted in the list of existing close
+        caption metas on the input frame (if any): add, drop, or
+        replace.
+    -   "ntsc-only" property to only look for captions if video has NTSC
+        resolution
+
+-   line21enc: new "remove-caption-meta" to remove metas from output
+    buffers after encoding the captions into the video data; support for
+    CDP closed captions
+
+-   matroskademux, matroskamux: Add support for ffv1, a lossless
+    intra-frame video coding format.
+
+-   matroskamux: accept in-band SPS/PPS/VPS for H264 and H265
+    (i.e. stream-format avc3 and hev1) which allows on-the-fly
+    profile/level/resolution changes.
+
+-   matroskamux: new "cluster-timestamp-offset" property, useful for use
+    cases where the container timestamps should map to some absolute
+    wall clock time, for example.
+
+-   rtpsrc: add "caps" property to allow explicit setting of the caps
+    where needed
+
+-   mpegts: support SCTE-35 passthrough via new "send-scte35-events"
+    property on MPEG-TS demuxer tsdemux. When enabled, SCTE 35 sections
+    (eg ad placement opportunities) are forwarded as events donwstream
+    where they can be picked up again by mpegtsmux. This required a
+    semantic change in the SCTE-35 section API: timestamps are now in
+    running time instead of muxer pts.
+
+-   tsdemux: Handle PCR-less MPEG-TS streams; more robust timestamp
+    handling in certain corner cases and for poorly muxed streams.
+
+-   mpegtsmux:
+
+    -   More conformance improvements to make MPEG-TS analyzers happy:
+        -   PCR timing accuracy: Improvements to the way mpegtsmux
+            outputs PCR observations in CBR mode, so that a PCR
+            observation is always inserted when needed, so that we never
+            miss the configured pcr-interval, as that triggers various
+            MPEG-TS analyser errors.
+        -   Improved PCR/SI scheduling
+    -   Don’t write PCR until PAT/PMT are output to make sure streams
+        start cleanly with a PAT/PMT.
+    -   Allow overriding the automatic PMT PID selection via
+        application-supplied PMT_%d fields in the prog-map
+        structure/property.
+
+-   mp4mux:
+
+    -   new "first-moov-then-finalise" mode for fragmented output where
+        the output will start with a self-contained moov atom for the
+        first fragment, and then produce regular fragments. Then at the
+        end when the file is finalised, the initial moov is invalidated
+        and a new moov is written covering the entire file. This way the
+        file is a “fragmented mp4” file while it is still being written
+        out, and remains playable at all times, but at the end it is
+        turned into a regular mp4 file (with former fragment headers
+        remaining as unused junk data in the file).
+    -   support H.264 avc3 and H.265 hvc1 stream formats as input where
+        the codec data is signalled in-band inside the bitstream instead
+        of caps/file headers.
+    -   support profile/level/resolution changes for H264/H265 input
+        streams (i.e. codec data changing on the fly). Each codec_data
+        is put into its own SampleTableEntry inside the stsd, unless the
+        input is in avc3 stream format in which case it’s written
+        in-band an not in the headers.
+
+-   multifilesink: new ""min-keyframe-distance"" property to make
+    minimum distance between keyframes in next-file=key-frame mode
+    configurable instead of hard-coding it to 10 seconds.
+
+-   mxfdemux has seen a big refactoring to support non-frame wrappings
+    and more accurate timestamp/seek handling for some formats
+
+-   msdk plugin for hardware-accelerated video encoding and decoding
+    using the Intel Media SDK:
+
+    -   oneVPL support (Intel oneAPI Video Processing Library)
+    -   AV1 decoding support
+    -   H.264 decoder now supports constrained-high and progressive-high
+        profiles
+    -   H.264 encoder:
+        -   more configuration options (properties):
+            "intra-refresh-type", "min-qp" , "max-qp", "p-pyramid",
+            "dblk-idc"
+    -   H.265 encoder:
+        -   can output main-still-picture profile
+        -   now inserts HDR SEIs (mastering display colour volume and
+            content light level)
+        -   more configuration options (properties):
+            "intra-refresh-type", "min-qp" , "max-qp", "p-pyramid",
+            "b-pyramid", "dblk-idc", "transform-skip"
+        -   support for RGB 10bit format
+    -   External bitrate control in encoders
+    -   Video post proc element msdkvpp gained support for 12-bit pixel
+        formats P012_LE, Y212_LE and Y412_LE
+
+-   nvh264sldec: interlaced stream support
+
+-   openh264enc: support main, high, constrained-high and
+    progressive-high profiles
+
+-   openjpeg: support for multithreaded decoding and encoding
+
+-   rtspsrc: now supports IPv6 also for tunneled mode (RTSP-over-HTTP);
+    new "ignore-x-server-reply" property to ignore the
+    x-server-ip-address server header reply in case of HTTP tunneling,
+    as it is often broken.
+
+-   souphttpsrc: Runtime compatibility support for libsoup2 and
+    libsoup3. libsoup3 is the latest major version of libsoup, but
+    libsoup2 and libsoup3 can’t co-exist in the same process because
+    there is no namespacing or versioning for GObject types. As a
+    result, it would be awkward if the GStreamer souphttpsrc plugin
+    linked to a specific version of libsoup, because it would only work
+    with applications that use the same version of libsoup. To make this
+    work, the soup plugin now tries to determine the libsoup version
+    used by the application (and its other dependencies) at runtime on
+    systems where GStreamer is linked dynamically. libsoup3 support is
+    still considered somewhat experimental at this point.
+
+-   srtsrc, srtsink: add signals for the application to accept/reject
+    incoming connections
+
+-   timeoverlay: new elapsed-running-time time mode which shows the
+    running time since the first running time (and each flush-stop).
+
+-   udpsrc: new timestamping mode to retrieve packet receive timestamps
+    from the kernel via socket control messages (SO_TIMESTAMPNS) on
+    supported platforms
+
+-   uritranscodebin: new setup-source and element-setup signals for
+    applications to configure elements used
+
+-   v4l2codecs plugin gained support for 4x4 and 32x32 tile formats
+    enabling some platforms or direct renders. Important memory usage
+    improvement.
+
+-   v4l2slh264dec now implements the final Linux uAPI as shipped on
+    Linux 5.11 and later.
+
+-   valve: add "drop-mode" property and provide two new modes of
+    operation: in drop-mode=forward-sticky-events sticky events
+    (stream-start, segment, tags, caps, etc.) are forwarded downstream
+    even when dropping is enabled; drop-mode=transform-to-gap will in
+    addition also convert buffers into gap events when dropping is
+    enabled, which lets downstream elements know that time is advancing
+    and might allow for preroll in many scenarios. By default all events
+    and all buffers are dropped when dropping is enabled, which can
+    cause problems with caps negotiation not progressing or branches not
+    prerolling when dropping is enabled.
+
+-   videocrop: support for many more pixel formats, e.g. planar YUV
+    formats with > 8bits and GBR* video formats; can now also accept
+    video not backed by system memory as long as downstream supports the
+    GstCropMeta
+
+-   videotestsrc: new smpte-rp-219 pattern for SMPTE75 RP-219 conformant
+    color bars
+
+-   vp8enc: finish support for temporal scalability: two new properties
+    ("temporal-scalability-layer-flags",
+    "temporal-scalability-layer-sync-flags") and a unit change on the
+    "temporal-scalability-target-bitrate" property (now expects bps);
+    also make temporal scalability details available to RTP payloaders
+    as buffer metadata.
+
+-   vp9enc: new properties to tweak encoder performance:
+
+    -   "aq-mode" to configure adaptive quantization modes
+    -   "frame-parallel-decoding" to configure whether to create a
+        bitstream that reduces decoding dependencies between frames
+        which allows staged parallel processing of more than one video
+        frames in the decoder. (Defaults to TRUE)
+    -   "row-mt", "tile-columns" and "tile-rows" so multithreading can
+        be enabled on a per-tile basis, instead of on a per tile-column
+        basis. In combination with the new "tile-rows" property, this
+        allows the encoder to make much better use of the available CPU
+        power.
+
+-   vp9dec, vp9enc: add support for 10-bit 4:2:0 and 4:2:2 YUV, as well
+    as 8-bit 4:4:4
+
+-   vp8enc, vp9enc now default to “good quality” for the deadline
+    property rather then “best quality”. Having the deadline set to best
+    quality causes the encoder to be absurdly slow, most real-life users
+    will prefer good-enough quality with better performance instead.
+
+-   wpesrc:
+
+    -   implement audio support: a new sometimes source pad will be
+        created for each audio stream created by the web engine.
+    -   move wpesrc to wpevideosrc and add a wrapper bin wpesrc to also
+        support audio
+    -   also handles web:// URIs now (same as cefsrc)
+    -   post messages with the estimated load progress on the bus
 
-Plugin and library moves
+-   x265enc: add negative DTS support, which means timestamps are now
+    offset by 1h same as with x264enc
+
+RTP Payloaders and Depayloaders
+
+-   rtpisacpay, rtpisacdepay: new RTP payloader and depayloader for iSAC
+    audio codec
+
+-   rtph264depay:
+
+    -   new "request-keyframe" property to make the depayloader
+        automatically request a new keyframe from the sender on packet
+        loss, consistent with the new property on rtpvp8depay.
+    -   new "wait-for-keyframe" property to make depayloader wait for a
+        new keyframe at the beginning and after packet loss (only
+        effective if the depayloader outputs AUs), consistent with the
+        existing property on rtpvp8depay.
+
+-   rtpopuspay, rtpopusdepay: support libwebrtc-compatible multichannel
+    audio in addition to the previously supported multichannel audio
+    modes
+
+-   rtpopuspay: add DTX (Discontinuous Transmission) support
+
+-   rtpvp8depay: new "request-keyframe" property to make the depayloader
+    automatically request a new keyframe from the sender on packet loss.
+
+-   rtpvp8pay: temporal scaling support
 
--   this section will be filled in in due course
+-   rtpvp9depay: Improved SVC handling (aggregate all layers)
+
+RTP Infrastructure
+
+-   rtpst2022-1-fecdec, rtpst2022-1-fecenc: new elements providing SMPTE
+    2022-1 2-D Forward Error Correction. More details in Mathieu’s blog
+    post.
+
+-   rtpreddec: BUNDLE support
+
+-   rtpredenc, rtpulpfecenc: add support for Transport-wide Congestion
+    Control (TWCC)
+
+-   rtpsession: new "twcc-feedback-interval" property to allow RTCP TWCC
+    reports to be scheduled on a timer instead of per marker-bit.
+
+Plugin and library moves
 
 -   There were no plugin moves or library moves in this cycle.
 
@@ -56,8 +998,6 @@ Plugin removals
 
 The following elements or plugins have been removed:
 
--   this section will be filled in in due course
-
 -   The ofa audio fingerprinting plugin has been removed. The MusicIP
     database has been defunct for years so this plugin is likely neither
     useful nor used by anyone.
@@ -71,135 +1011,575 @@ The following elements or plugins have been removed:
 
 Miscellaneous API additions
 
--   this section will be filled in in due course
+Core
+
+-   gst_buffer_new_memdup() is a convenience function for the
+    widely-used gst_buffer_new_wrapped(g_memdup(data,size),size)
+    pattern.
+
+-   gst_caps_features_new_single() creates a new single GstCapsFeatures,
+    avoiding the need to use the vararg function with NULL terminator
+    for simple cases.
+
+-   gst_element_type_set_skip_documentation() can be used by plugins to
+    signal that certain elements should not be included in the GStreamer
+    plugin documentation. This is useful for plugins where elements are
+    registered dynamically based on hardware capabilities and/or where
+    the available plugins and properties vary from system to system.
+    This is used in the d3d11 plugin for example to ensure that only the
+    list of default elements is advertised in the documentation.
+
+-   gst_type_find_suggest_empty_simple() is a new convenience function
+    for typefinders for cases where there’s only a media type and no
+    other fields.
+
+-   New API to create elements and set properties at construction time,
+    which is not only convenient, but also allows GStreamer elements to
+    have construct-only properties: gst_element_factory_make_full(),
+    gst_element_factory_make_valist(),
+    gst_element_factory_make_with_properties(),
+    gst_element_factory_create_full(),
+    gst_element_factory_create_valist(),
+    gst_element_factory_create_with_properties().
+
+-   GstSharedTaskPool: new “shared” task pool subclass with slightly
+    different default behaviour than the existing GstTaskPool which
+    would create unlimited number of threads for new tasks. The shared
+    taskpool creates up to N threads (default: 1) and then distributes
+    pending tasks to those threads round-robin style, and blocks if no
+    thread is available. It is possible to join tasks. This can be used
+    by plugins to implement simple multi-threaded processing and is used
+    for the new multi-threaded video conversion and compositing done in
+    GstVideoAggregator, videoconverter and compositor.
+
+Plugins Base Utils library
+
+-   GstDiscoverer:
+
+    -   gst_discoverer_container_info_get_tags() was added to retrieve
+        global/container tags (vs. per-stream tags). Per-Stream tags can
+        be retrieved via the existing
+        gst_discoverer_stream_info_get_tags().
+        gst_discoverer_info_get_tags(), which for many files returns a
+        confusing mix of stream and container tags, has been deprecated
+        in favour of the container/stream-specific functions.
+    -   gst_discoverer_stream_info_get_stream_number() returns a unique
+        integer identifier for a given stream within the given
+        GstDiscoverer context. (If this matches the stream number inside
+        the container bitstream that’s by coincidence and not by
+        design.)
+
+-   gst_pb_utils_get_caps_description_flags() can be used to query
+    whether certain caps represent a container, audio, video, image,
+    subtitles, tags, or something else. This only works for formats
+    known to GStreamer.
+
+-   gst_pb_utils_get_file_extension_from_caps() returns a possible file
+    extension for given caps.
+
+-   gst_codec_utils_h264_get_profile_flags_level(): Parses profile,
+    flags, and level from H264 AvcC codec_data. The format of H264 AVCC
+    extradata/sequence_header is documented in the ITU-T H.264
+    specification section 7.3.2.1.1 as well as in ISO/IEC 14496-15
+    section 5.3.3.1.2.
+
+-   gst_codec_utils_caps_get_mime_codec() to convert caps to a RFC 6381
+    compatible MIME codec string codec. Useful for providing the codecs
+    field inside the Content-Type HTTP header for containerized formats,
+    such as mp4 or matroska.
+
+GStreamer OpenGL integration library and plugins
+
+-   glcolorconvert: added suppport for converting the video formats
+    A420, AV12, BGR, BGRA, RGBP and BGRP.
+
+-   Added support to GstGLBuffer for persistent buffer mappings where a
+    Pixel Buffer Object (PBO) can be mapped by both the CPU and the GPU.
+    This removes a memcpy() when uploading textures or vertices
+    particularly when software decoders (e.g. libav) are direct
+    rendering into our memory. Improves transfer performance
+    significantly. Requires OpenGL 4.4, GL_ARB_buffer_storage or
+    GL_EXT_buffer_storage
+
+-   Added various helper functions for handling 4x4 matrices of affine
+    transformations as used by GstVideoAffineTransformationMeta.
+
+-   Add support to GstGLContext for allowing the application to control
+    the config (EGLConfig, GLXConfig, etc) used when creating the OpenGL
+    context. This allows the ability to choose between RGB16 or RGB10A2
+    or RGBA8 back/front buffer configurations that were previously
+    hardcoded. GstGLContext also supports retrieving the configuration
+    it was created with or from an externally provide OpenGL context
+    handle. This infrastructure is also used to create a compatible
+    config from an application/externally provided OpenGL context in
+    order to improve compatibility with other OpenGL frameworks and GUI
+    toolkits. A new environment variable GST_GL_CONFIG was also added to
+    be able to request a specific configuration from the command line.
+    Note: different platforms will have different functionality
+    available.
+
+-   Add support for choosing between EGL and WGL at runtime when running
+    on Windows. Previously this was a build-time switch. Allows use in
+    e.g. Gtk applications on Windows that target EGL/ANGLE without
+    recompiling GStreamer. gst_gl_display_new_with_type() can be used by
+    applications to choose a specific display type to use.
+
+-   Build fixes to explicitly check for Broadcom-specific libraries on
+    older versions of the Raspberry Pi platform. The Broadcom OpenGL ES
+    and EGL libraries have different filenames. Using the vc4 Mesa
+    driver on the Raspberry Pi is not affected.
+
+-   Added support to glupload and gldownload for transferring RGBA
+    buffers using the memory:NVMM available on the Nvidia Tegra family
+    of embedded devices.
+
+-   Added support for choosing libOpenGL and libGLX as used in a GLVND
+    environment on unix-based platforms. This allows using desktop
+    OpenGL and EGL without pulling in any GLX symbols as would be
+    required with libGL.
+
+Video library
+
+-   New raw video formats:
+
+    -   AV12 (NV12 with alpha plane)
+    -   RGBP and BGRP (planar RGB formats)
+    -   ARGB64 variants with specified endianness instead of host
+        endianness:
+        -   ARGB64_LE, ARGB64_BE
+        -   RGBA64_BE, RGBA64_LE
+        -   BGRA64_BE, BGRA64_LE
+        -   ABGR64_BE, ABGR64_LE
+
+-   gst_video_orientation_from_tag() is new convenience API to parse the
+    image orientation from a GstTagList.
+
+-   GstVideoDecoder subframe support (see below)
+
+-   GstVideoCodecState now also carries some HDR metadata
+
+-   Ancillary video data: implement transform functions for AFD/Bar
+    metas, so they will be forwarded in more cases
+
+MPEG-TS library
+
+This library only handles section parsing and such, see above for
+changes to the actual mpegtsmux and mpegtsdemux elements.
+
+-   many additions and improvements to SCTE-35 section parsing
+-   new API for fetching extended descriptors:
+    gst_mpegts_find_descriptor_with_extension()
+-   add support for SIT sections (Selection Information Tables)
+-   expose event-from-section constructor gst_event_new_mpegts_section()
+-   parse Audio Preselection Descriptor needed for Dolby AC-4
+
+GstWebRTC library + webrtcbin
+
+-   Change the way in which sink pads and transceivers are matched
+    together to support easier usage. If a pad is created without a
+    specific index (i.e. using sink_%u as the pad template), then an
+    available compatible transceiver will be searched for. If a specific
+    index is requested (i.e. sink_1) then if a transceiver for that
+    m-line already exists, that transceiver must match the new sink pad
+    request. If there is no transceiver available in either scenario, a
+    new transceiver is created. If a mixture of both sink_1 and sink_%u
+    requests result in an impossible situation, an error will be
+    produced at pad request time or from create offer/answer.
+
+-   webrtcbin now uses regular ICE nomination instead of libnice’s
+    default of aggressive ICE nomination. Regular ICE nomination is the
+    default recommended by various relevant standards and improves
+    connectivity in specific network scenarios.
+
+-   Add support for limiting the port range used for RTP with the
+    addition of the min-rtp-port and max-rtp-port properties on the ICE
+    object.
+
+-   Expose the SCTP transport as a property on webrtcbin to more closely
+    match the WebRTC specification.
+
+-   Added support for taking into account the data channel transport
+    state when determining the value of the "connection-state" property.
+    Previous versions of the WebRTC spec did not include the data
+    channel state when computing this value.
+
+-   Add configuration for choosing the size of the underlying sockets
+    used for transporting media data
+
+-   Always advertise support for the transport-cc RTCP feedback protocol
+    as rtpbin supports it. For full support, the configured caps (input
+    or through codec-preferences) need to include the relevant RTP
+    header extension.
+
+-   Numerous fixes to caps and media handling to fail-fast when an
+    incompatible situation is detected.
+
+-   Improved support for attaching the required media after a remote
+    offer has been set.
+
+-   Add support for dynamically changing the amount of FEC used for a
+    particular stream.
+
+-   webrtcbin now stops further SDP processing at the first error it
+    encounters.
+
+-   Completed support for either local or the remote closing a data
+    channel.
+
+-   Various fixes when performing BUNDLEing of the media streams in
+    relation to RTX and FEC usage.
+
+-   Add support for writing out QoS DSCP marking on outgoing packets to
+    improve reliability in some network scenarios.
+
+-   Improvements to the statistics returned by the get-stats signal
+    including the addition of the raw statistics from the internal
+    RTPSource, the TWCC stats when available.
+
+-   The webrtc library does not expose any objects anymore with public
+    fields. Instead properties have been added to replace that
+    functionality. If you are accessing such fields in your application,
+    switch to the corresponding properties.
+
+GstCodecs and Video Parsers
+
+-   Support for render delays to improve throughput across all CODECs
+    (used with NVDEC and V4L2).
+-   lots of improvements to parsers and the codec parsing decoder base
+    classes (H264, H265, VP8, VP9, AV1, MPEG-2) used for various
+    hardware-accelerate decoder APIs.
+
+Bindings support
+
+-   gst_allocation_params_new() allocates a GstAllocationParams struct
+    on the heap. This should only be used by bindings (and freed via
+    gst_allocation_params_free() then). In C code you would allocate
+    this on the stack and only init it in place.
+
+-   gst_debug_log_literal() can be used to log a string to the debug log
+    without going through any printf format expansion and associated
+    overhead. This is mostly useful for bindings such as the Rust
+    bindings which may have done their own formatting already .
+
+-   Provide non-inlined versions of refcounting APIs for various
+    GStreamer mini objects, so that they can be consumed by bindings
+    (e.g. gstreamer-sharp): gst_buffer_ref, gst_buffer_unref,
+    gst_clear_buffer, gst_buffer_copy, gst_buffer_replace,
+    gst_buffer_list_ref, gst_buffer_list_unref, gst_clear_buffer_list,
+    gst_buffer_list_copy, gst_buffer_list_replace, gst_buffer_list_take,
+    gst_caps_ref, gst_caps_unref, gst_clear_caps, gst_caps_replace,
+    gst_caps_take, gst_context_ref, gst_context_unref, gst_context_copy,
+    gst_context_replace, gst_event_replace, gst_event_steal,
+    gst_event_take, gst_event_ref, gst_event_unref, gst_clear_event,
+    gst_event_copy, gst_memory_ref, gst_memory_unref, gst_message_ref,
+    gst_message_unref, gst_clear_message, gst_message_copy,
+    gst_message_replace, gst_message_take, gst_promise_ref,
+    gst_promise_unref, gst_query_ref, gst_query_unref, gst_clear_query,
+    gst_query_copy, gst_query_replace, gst_query_take, gst_sample_ref,
+    gst_sample_unref, gst_sample_copy, gst_tag_list_ref,
+    gst_tag_list_unref, gst_clear_tag_list, gst_tag_list_replace,
+    gst_tag_list_take, gst_uri_copy, gst_uri_ref, gst_uri_unref,
+    gst_clear_uri.
+
+-   expose a GType for GstMiniObject
+
+-   gst_device_provider_probe() now returns non-floating device object
+
+API Deprecations
+
+-   gst_element_get_request_pad() has been deprecated in favour of the
+    newly-added gst_element_request_pad_simple() which does the exact
+    same thing but has a less confusing name that hopefully makes clear
+    that the function request a new pad rather than just retrieves an
+    already-existing request pad.
+
+-   gst_discoverer_info_get_tags(), which for many files returns a
+    confusing mix of stream and container tags, has been deprecated in
+    favour of the container-specific and stream-specific functions,
+    gst_discoverer_container_info_get_tags() and
+    gst_discoverer_stream_info_get_tags().
+
+-   gst_video_sink_center_rect() was deprecated in favour of the more
+    generic newly-added gst_video_center_rect().
+
+-   The GST_MEMORY_FLAG_NO_SHARE flag has been deprecated, as it tends
+    to cause problems and prevents sub-buffering. If pooling or lifetime
+    tracking is required, memories should be allocated through a custom
+    GstAllocator instead of relying on the lifetime of the buffers the
+    memories were originally attached to, which is fragile anyway.
+
+-   The GstPlayer high-level playback library is being replaced with the
+    new GstPlay library (see above). GstPlayer should be considered
+    deprecated at this point and will be marked as such in the next
+    development cycle. Applications should be ported to GstPlay.
+
+-   Gstreamer Editing Services: ges_video_transition_set_border(),
+    ges_video_transition_get_border()
+    ges_video_transition_set_inverted()
+    ges_video_transition_is_inverted() have been deprecated, use
+    ges_timeline_element_set_children_properties() instead.
 
 Miscellaneous performance, latency and memory optimisations
 
--   this section will be filled in in due course
+More video conversion fast paths
 
-Miscellaneous other changes and enhancements
+-   v210 ↔ I420, YV12, Y42B, UYVY and YUY2
+-   A420 → RGB
+
+Less jitter when waiting on the system clock
+
+-   Better system clock wait accuracy, less jitter: where available,
+    clock_nanosleep is used for higher accuracy for waits below 500
+    usecs, and waits below 2ms will first use the regular waiting system
+    and then clock_nanosleep for the remainder. The various wait
+    implementation have a latency ranging from 50 to 500+ microseconds.
+    While this is not a major issue when dealing with a low number of
+    waits per second (for ex: video), it does introduce a non-negligible
+    jitter for synchronisation of higher packet rate systems.
+
+Video decoder subframe support
 
--   this section will be filled in in due course
+-   The GstVideoDecoder base class gained API to process input at the
+    sub-frame level. That way video decoders can start decoding slices
+    before they have received the full input frame in its entirety (to
+    the extent this is supported by the codec, of course). This helps
+    with CPU utilisation and reduces latency.
 
-Tracing framework and debugging improvements
+-   This functionality is now being used in the OpenJPEG JPEG 2000
+    decoder, the FFmpeg H.264 decoder (in case of NAL-aligned input) and
+    the OpenMAX H.264/H.265 decoders (in case of NAL-aligned input).
 
--   this section will be filled in in due course
+Miscellaneous other changes and enhancements
+
+-   GstDeviceMonitor no longer fails to start just because one of the
+    device providers failed to start. That could happen for example on
+    systems where the pulseaudio device provider is installed, but
+    pulseaudio isn’t actually running but ALSA is used for audio
+    instead. In the same vein the device monitor now keeps track of
+    which providers have been started (via the new
+    gst_device_provider_is_started()) and only stops actually running
+    device providers when stopping the device monitor.
+
+-   On embedded systems it can be useful to create a registry that can
+    be shared and read by multiple processes running as different users.
+    It is now possible to set the new GST_REGISTRY_MODE environment
+    variable to specify the file mode for the registry file, which by
+    default is set to be only user readable/writable.
+
+-   GstNetClientClock will signal lost sync in case the remote time
+    resets (e.g. because device power cycles), by emitting the “synced”
+    signal with synced=FALSE parameter, so applications can take action.
+
+-   gst_value_deserialize_with_pspec() allows deserialization with a
+    hint for what the target GType should be. This allows for example
+    passing arrays of flags through the command line or
+    gst_util_set_object_arg(), eg: foo="<bar,bar+baz>".
+
+-   It’s now allowed to create an empty GstVideoOverlayComposition
+    without any rectangles by passing a NULL rectangle to
+    gst_video_overlay_composition_new(). This is useful for bindings and
+    simplifies application code in some places.
+
+Tracing framework, debugging and testing improvements
+
+-   New factories tracer to list loaded elements (and other plugin
+    features). This can be useful to collect a list of elements needed
+    for an application, which then in turn can be used to create a
+    tailored minimal GStreamer build that contains just the elements
+    needed and nothing else.
+-   New plugin-feature-loaded tracing hook for use by tracers like the
+    new factories tracer
+
+-   GstHarness: Add gst_harness_set_live() so that harnesses can be set
+    to non-live and return is-live=false in latency queries if needed.
+    Default behaviour is to always return is-live=true in latency
+    queries.
+
+-   navseek: new "hold-eos" property. When enabled, the element will
+    hold back an EOS event until the next keystroke (via navigation
+    events). This can be used to keep a video sink showing the last
+    frame of a video pipeline until a key is pressed instead of tearing
+    it down immediately on EOS.
+
+-   New fakeaudiosink element: mimics an audio sink and can be used for
+    testing and CI pipelines on systems where no audio system is
+    installed or running. It differs from fakesink in that it only
+    support audio caps and syncs to the clock by default like a normal
+    audio sink. It also implements the GstStreamVolume interface like
+    most audio sinks do.
+
+-   New videocodectestsink element for video codec conformance testing:
+    Calculates MD5 checksums for video frames and skips any padding
+    whilst doing so. Can optionally also write back the video data with
+    padding removed into a file for easy byte-by-byte comparison with
+    reference data.
 
 Tools
 
--   this section will be filled in in due course
+gst-inspect-1.0
+
+-   Can sort the list of plugins by passing --sort=name as command line
+    option
+
+gst-launch-1.0
+
+-   will now error out on top-level properties that don’t exist and
+    which were silently ignored before
+-   On Windows the high-resolution clock is enabled now, which provides
+    better clock and timer performance on Windows (see Windows section
+    below for more details).
+
+gst-play-1.0
+
+-   New --start-position command line argument to start playback from
+    the specified position
+-   Audio can be muted/unmuted in interactive mode by pressing the m
+    key.
+-   On Windows the high-resolution clock is enabled now (see Windows
+    section below for more details)
+
+gst-device-monitor-1.0
+
+-   New --include-hidden command line argument to also show “hidden”
+    device providers
+
+ges-launch-1.0
+
+-   New interactive mode that allows seeking and such. Can be disabled
+    by passing the --no-interactive argument on the command line.
+-   Option to forward tags
+-   Allow using an existing clip to determine the rendering format (both
+    topology and profile) via new --profile-from command line argument.
 
 GStreamer RTSP server
 
--   this section will be filled in in due course
+-   GstRTSPMediaFactory gained API to disable RTCP
+    (gst_rtsp_media_factory_set_enable_rtcp(), "enable-rtcp" property).
+    Previously RTCP was always allowed for all RTSP medias. With this
+    change it is possible to disable RTCP completely, no matter if the
+    client wants to do RTCP or not.
+
+-   Make a mount point of / work correctly. While not allowed by the
+    RTSP 2 spec, the RTSP 1 spec is silent on this and it is used in the
+    wild. It is now possible to use / as a mount path in
+    gst-rtsp-server, e.g. rtsp://example.com/ would work with this now.
+    Note that query/fragment parts of the URI are not necessarily
+    correctly handled, and behaviour will differ between various
+    client/server implementations; so use it if you must but don’t bug
+    us if it doesn’t work with third party clients as you’d hoped.
+
+-   multithreading fixes (races, refcounting issues, deadlocks)
+
+-   ONVIF audio backchannel fixes
+
+-   ONVIF trick mode optimisations
+
+-   rtspclientsink: new "update-sdp" signal that allows updating the SDP
+    before sending it to the server via ANNOUNCE. This can be used to
+    add additional metadata to the SDP, for example. The order and
+    number of medias must not be changed, however.
 
 GStreamer VAAPI
 
--   this section will be filled in in due course
+-   new AV1 decoder element (vaapiav1dec)
+
+-   H264 decoder: handle stereoscopic 3D video with frame packing
+    arrangement SEI messages
+
+-   H265 encoder: added Screen Content Coding extensions support
+
+-   H265 decoder: gained MAIN_444_12 profile support (decoded to
+    Y412_LE), and 4:2:2 12-bits support (decoded to Y212_LE)
+
+-   vaapipostproc: gained BT2020 color standard support
+
+-   vaapidecode: now generates caps templates dynamically at runtime in
+    order to advertise actually supported caps instead of all
+    theoretically supported caps.
+
+-   GST_VAAPI_DRM_DEVICE environment variable to force a specified DRM
+    device when a DRM display is used. It is ignored when other types of
+    displays are used. By default /dev/dri/renderD128 is used for DRM
+    display.
 
 GStreamer OMX
 
--   this section will be filled in in due course
+-   subframe support in H.264/H.265 decoders
 
 GStreamer Editing Services and NLE
 
--   this section will be filled in in due course
+-   framepositioner: new "operator" property to access blending modes in
+    the compositor
+-   timeline: Implement snapping to markers
+-   smart-mixer: Add support for d3d11compositor and glvideomixer
+-   titleclip: add "draw-shadow" child property
+-   ges:// URI support to define a timeline from a description.
+-   command-line-formatter
+    -   Add track management to timeline description
+    -   Add keyframe support
+-   ges-launch-1.0:
+    -   Add an interactive mode where we can seek etc…
+    -   Add option to forward tags
+    -   Allow using an existing clip to determine the rendering format
+        (both topology and profile) via new --profile-from command line
+        argument.
+-   Fix static build
 
 GStreamer validate
 
--   this section will be filled in in due course
+-   report: Add a way to force backtraces on reports even if not a
+    critical issue (GST_VALIDATE_ISSUE_FLAGS_FORCE_BACKTRACE)
+-   Add a flag to gst_validate_replace_variables_in_string() allow
+    defining how to resolve variables in structs
+-   Add gst_validate_bin_monitor_get_scenario() to get the bin monitor
+    scenario, which is useful for applications that use Validate
+    directly.
+-   Add an expected-values parameter to wait, message-type=XX allowing
+    more precise filtering of the message we are waiting for.
+-   Add config file support: each test can now use a config file for the
+    given media file used to test.
+-   Add support to check properties of object properties
+-   scenario: Add an "action-done" signal to signal when an action is
+    done
+-   scenario: Add a "run-command" action type
+-   scenario: Allow forcing running action on idle from scenario file
+-   scenario: Allow iterating over arrays in foreach
+-   scenario: Rename ‘interlaced’ action to ‘non-blocking’
+-   scenario: Add a non-blocking flag to the wait signal
 
 GStreamer Python Bindings
 
--   this section will be filled in in due course
+-   Fixes for Python 3.10
+-   Various build fixes
+-   at least one known breaking change caused by g-i annotation changes
+    (see below)
 
 GStreamer C# Bindings
 
--   this section will be filled in in due course
+-   Fix GstDebugGraphDetails enum
+-   Updated to latests GtkSharp
+-   Updated to include GStreamer 1.20 API
 
 GStreamer Rust Bindings and Rust Plugins
 
-The GStreamer Rust bindings are released separately with a different
-release cadence that’s tied to gtk-rs, but the latest release has
-already been updated for the upcoming new GStreamer 1.20 API.
-
-gst-plugins-rs, the module containing GStreamer plugins written in Rust,
-has also seen lots of activity with many new elements and plugins.
-
-What follows is a list of elements and plugins available in
-gst-plugins-rs, so people don’t miss out on all those potentially useful
-elements that have no C equivalent.
-
--   FIXME: add new elements
-
-Rust audio plugins
-
--   audiornnoise: New element for audio denoising which implements the
-    noise removal algorithm of the Xiph RNNoise library, in Rust
--   rsaudioecho: Port of the audioecho element from gst-plugins-good
-    rsaudioloudnorm: Live audio loudness normalization element based on
-    the FFmpeg af_loudnorm filter
--   claxondec: FLAC lossless audio codec decoder element based on the
-    pure-Rust claxon implementation
--   csoundfilter: Audio filter that can use any filter defined via the
-    Csound audio programming language
--   lewtondec: Vorbis audio decoder element based on the pure-Rust
-    lewton implementation
-
-Rust video plugins
-
--   cdgdec/cdgparse: Decoder and parser for the CD+G video codec based
-    on a pure-Rust CD+G implementation, used for example by karaoke CDs
--   cea608overlay: CEA-608 Closed Captions overlay element
--   cea608tott: CEA-608 Closed Captions to timed-text (e.g. VTT or SRT
-    subtitles) converter
--   tttocea608: CEA-608 Closed Captions from timed-text converter
--   mccenc/mccparse: MacCaption Closed Caption format encoder and parser
--   sccenc/sccparse: Scenarist Closed Caption format encoder and parser
--   dav1dec: AV1 video decoder based on the dav1d decoder implementation
-    by the VLC project
--   rav1enc: AV1 video encoder based on the fast and pure-Rust rav1e
-    encoder implementation
--   rsflvdemux: Alternative to the flvdemux FLV demuxer element from
-    gst-plugins-good, not feature-equivalent yet
--   rsgifenc/rspngenc: GIF/PNG encoder elements based on the pure-Rust
-    implementations by the image-rs project
-
-Rust text plugins
-
--   textwrap: Element for line-wrapping timed text (e.g. subtitles) for
-    better screen-fitting, including hyphenation support for some
-    languages
-
-Rust network plugins
-
--   reqwesthttpsrc: HTTP(S) source element based on the Rust
-    reqwest/hyper HTTP implementations and almost feature-equivalent
-    with the main GStreamer HTTP source souphttpsrc
--   s3src/s3sink: Source/sink element for the Amazon S3 cloud storage
--   awstranscriber: Live audio to timed text transcription element using
-    the Amazon AWS Transcribe API
-
-Generic Rust plugins
-
--   sodiumencrypter/sodiumdecrypter: Encryption/decryption element based
-    on libsodium/NaCl
--   togglerecord: Recording element that allows to pause/resume
-    recordings easily and considers keyframe boundaries
--   fallbackswitch/fallbacksrc: Elements for handling potentially
-    failing (network) sources, restarting them on errors/timeout and
-    showing a fallback stream instead
--   threadshare: Set of elements that provide alternatives for various
-    existing GStreamer elements but allow to share the streaming threads
-    between each other to reduce the number of threads
--   rsfilesrc/rsfilesink: File source/sink elements as replacements for
-    the existing filesrc/filesink elements
+-   The GStreamer Rust bindings are released separately with a different
+    release cadence that’s tied to gtk-rs, but the latest release has
+    already been updated for the upcoming new GStreamer 1.20 API (v1_20
+    feature).
+
+-   gst-plugins-rs, the module containing GStreamer plugins written in
+    Rust, has also seen lots of activity with many new elements and
+    plugins. See the New Elements section above for a list of new Rust
+    elements.
 
 Build and Dependencies
 
 -   Meson 0.59 or newer is required to build GStreamer now.
 
--   FIXME: this section will be filled in in due course
+-   The GLib requirement has been bumped to GLib 2.56 or newer (from
+    March 2018).
+
+-   The wpe plugin now requires wpe >= 2.28 and wpebackend-fdo >= 1.8
 
 Explicit opt-in required for build of certain plugins with (A)GPL dependencies
 
@@ -211,11 +1591,9 @@ even if the required dependencies are available.
 See Building plugins with (A)GPL-licensed dependencies for more details
 and a non-exhaustive list of plugins affected.
 
-gst-build: replaced by Monorepo
+gst-build: replaced by mono repository
 
--   this section will be filled in in due course
-
--   FIXME: describe + link to Monorepo FAQ
+See mono repository section above and the GStreamer mono repository FAQ.
 
 Cerbero
 
@@ -223,76 +1601,254 @@ Cerbero is a meta build system used to build GStreamer plus dependencies
 on platforms where dependencies are not readily available, such as
 Windows, Android, iOS and macOS.
 
-General improvements
+General Cerbero improvements
 
--   this section will be filled in in due course
+-   Plugin removed: libvisual
+-   New plugins: rtpmanagerbad and rist
 
-macOS / iOS
+macOS / iOS specific Cerbero improvements
 
--   this section will be filled in in due course
+-   XCode 12 support
+-   macOS OS release support is now future-proof, similar to iOS
+-   macOS Apple Silicon (ARM64) cross-compile support has been added
+-   macOS Apple Silicon (ARM64) native support is currently experimental
 
-Windows
+Windows specific Cerbero improvements
 
--   this section will be filled in in due course
+-   Visual Studio 2022 support has been added
+-   bootstrap is faster since it requires building fewer build-tools
+    recipes on Windows
+-   package is faster due to better scheduling of recipe stages and
+    elimination of unnecessary autotools regeneration
+-   The following plugins are no longer built on Windows:
+    -   a52dec (another decoder is still available in libav)
+    -   dvdread
+    -   resindvd
 
 Windows MSI installer
 
--   this section will be filled in in due course
+-   no major changes
 
-Linux
+Linux specific Cerbero improvements
 
--   this section will be filled in in due course
+-   Fedora, Debian OS release support is now more future-proof
+-   Amazon Linux 2 support has been added
 
-Android
+Android specific Cerbero improvements
 
--   this section will be filled in in due course
+-   no major changes
 
 Platform-specific changes and improvements
 
 Android
 
--   this section will be filled in in due course
+-   No major changes
 
 macOS and iOS
 
--   this section will be filled in in due course
+-   applemedia: add ProRes support to vtenc and vtdec
 
 Windows
 
--   this section will be filled in in due course
+-   On Windows the high-resolution clock is enabled now in the
+    gst-launch-1.0 and gst-play-1.0 command line tools, which provides
+    better clock and timer performance on Windows, at the cost of higher
+    power consumption. By default, without the high-resolution clock
+    enabled, the timer precision on Windows is system-dependent and may
+    be as bad as 15ms which is not good enough for many multimedia
+    applications. Developers may want to do the same in their Windows
+    applications if they think it’s a good idea for their application
+    use case, and depending on the Windows version they target. This is
+    not done automatically by GStreamer because on older Windows
+    versions (pre-Windows 10) this affects a global Windows setting and
+    also there’s a power consumption vs. performance trade-off that may
+    differ from application to application.
+
+-   dxgiscreencapsrc now supports resolution changes
+
+-   The wasapi2 audio plugin was rewritten and now has a higher rank
+    than the old wasapi plugin since it has a number of additional
+    features such as automatic stream routing, and no
+    known-but-hard-to-fix issues. The plugin is always built if the
+    Windows 10 SDK is available now.
+
+-   The wasapi device providers now detect and notify dynamic device
+    additions/removals
+
+-   d3d11screencapturesrc: new desktop capture element, including
+    GstDeviceProvider implementation to enumerate/select target monitors
+    for capture.
+
+-   Direct3D11/DXVA decoder now supports AV1 and MPEG2 codecs
+    (d3d11av1dec, d3d11mpeg2dec)
+
+-   VP9 decoding got more reliable and stable thanks to a newly written
+    codec parser
+
+-   Support for decoding interlaced H.264/AVC streams
+
+-   Hardware-accelerated video deinterlacing (d3d11deinterlace) and
+    video mixing (d3d11compositor)
+
+-   Video mixing with the Direct3D11 API (d3d11compositor)
+
+-   MediaFoundation API based hardware encoders gained the ability to
+    receive Direct3D11 textures as an input
+
+-   Seungha’s blog post “GStreamer ❤ Windows: A primer on the cool stuff
+    you’ll find in the 1.20 release” describes many of the
+    Windows-related improvements in more detail
 
 Linux
 
--   this section will be filled in in due course
+-   bluez: LDAC Bluetooth audio codec support in a2dpsink and avdtpsink,
+    as well as an LDAC RTP payloader (rtpldacpay) and an LDAC audio
+    encoder (ldacenc)
+
+-   kmssink: gained support for NV24, NV61, RGB16/BGR16 formats;
+    auto-detect NVIDIA Tegra driver
 
 Documentation improvements
 
--   this section will be filled in in due course
+-   hardware-accelerated GPU plugins will now no longer always list all
+    the element variants for all available GPUs, since those are
+    system-dependent and it’s confusing for users to see those in the
+    documentation just because the GStreamer developer who generated the
+    docs had multiple GPUs to play with at the time. Instead just show
+    the default elements.
 
-Possibly Breaking Changes
+Possibly Breaking and Other Noteworthy Behavioural Changes
+
+-   gst_parse_launch(), gst_parse_bin_from_description() and friends
+    will now error out when setting properties that don’t exist on
+    top-level bins. They were silently ignored before.
+
+-   The GstWebRTC library does not expose any objects anymore with
+    public fields. Instead properties have been added to replace that
+    functionality. If you are accessing such fields in your application,
+    switch to the corresponding properties.
 
--   FIXME: this section will be filled in in due course
--   MPEG-TS SCTE-35 API changes (FIXME: flesh out)
--   gst_parse_launch() and friends now error out on non-existing
-    properties on top-level bins where they would silently fail and
-    ignore those before.
 -   playbin and uridecodebin now emit the source-setup signal before the
     element is added to the bin and linked so that the source element is
     already configured before any scheduling query comes in, which is
-    useful for elements such as appsrc or giostreamsrc. (Merge Request)
+    useful for elements such as appsrc or giostreamsrc.
+
+-   The source element inside urisourcebin (used inside uridecodebin3
+    which is used inside playbin3) is no longer called "source". This
+    shouldn’t affect anyone hopefully, because there’s a "setup-source"
+    signal to configure the source element and no one should rely on
+    names of internal elements anyway.
+
+-   The vp8enc element now expects bps (bits per second) for the
+    "temporal-scalability-target-bitrate" property, which is consistent
+    with the "target-bitrate" property. Since additional configuration
+    is required with modern libvpx to make temporal scaling work anyway,
+    chances are that very few people will have been using this property
+
+-   vp8enc and vp9enc now default to “good quality” for the "deadline"
+    property rather then “best quality”. Having the deadline set to best
+    quality causes the encoder to be absurdly slow, most real-life users
+    will want the good quality tradeoff instead.
+
+-   The experimental GstTranscoder library API in gst-plugins-bad was
+    changed from a GObject signal-based notification mechanism to a
+    GstBus/message-based mechanism akin to GstPlayer/GstPlay.
+
+-   MPEG-TS SCTE-35 API: semantic change for SCTE-35 splice commands:
+    timestamps passed by the application should be in running time now,
+    since users of the API can’t really be expected to predict the local
+    PTS of the muxer.
+
+-   The GstContext used by souphttpsrc to share the session between
+    multiple element instances has changed. Previously it provided
+    direct access to the internal SoupSession object, now it only
+    provides access to an opaque, internal type. This change is
+    necessary because SoupSession is not thread-safe at all and can’t be
+    shared safely between arbitrary external code and souphttpsrc.
+
+-   Python bindings: GObject-introspection related Annotation fixes have
+    led to a case of a GstVideo.VideoInfo-related function signature
+    changing in the Python bindings (possibly one or two other cases
+    too). This is for a function that should never have been exposed in
+    the first place though, so the bindings are being updated to throw
+    an exception in that case, and the correct replacement API has been
+    added in form of an override.
 
 Known Issues
 
--   this section will be filled in in due course
-
--   There are a couple of known WebRTC-related regressions/blockers:
-
-    -   webrtc: DTLS setup with Chrome is broken
-    -   webrtcbin: First keyframe is usually lost
+-   nothing in particular at this point (but also see possibly breaking
+    changes section above)
 
 Contributors
 
--   this section will be filled in in due course
+Aaron Boxer, Adam Leppky, Adam Williamson, Alba Mendez, Alejandro
+González, Aleksandr Slobodeniuk, Alexander Vandenbulcke, Alex Ashley,
+Alicia Boya García, Andika Triwidada, Andoni Morales Alastruey, Andrew
+Wesie, Andrey Moiseev, Antonio Ospite, Antonio Rojas, Arthur Crippa
+Búrigo, Arun Raghavan, Ashley Brighthope, Axel Kellermann, Baek, Bastien
+Nocera, Bastien Reboulet, Benjamin Gaignard, Bing Song, Binh Truong,
+Biswapriyo Nath, Brad Hards, Brad Smith, Brady J. Garvin, Branko
+Subasic, Camilo Celis Guzman, Chris Bass, ChrisDuncanAnyvision, Chris
+White, Corentin Damman, Daniel Almeida, Daniel Knobe, Daniel Stone,
+david, David Fernandez, David Keijser, David Phung, Devarsh Thakkar,
+Dinesh Manajipet, Dmitry Samoylov, Dmitry Shusharin, Dominique Martinet,
+Doug Nazar, Ederson de Souza, Edward Hervey, Emmanuel Gil Peyrot,
+Enrique Ocaña González, Ezequiel Garcia, Fabian Orccon, Fabrice
+Fontaine, Fernando Jimenez Moreno, Florian Karydes, Francisco Javier
+Velázquez-García, François Laignel, Frederich Munch, Fredrik Pålsson,
+George Kiagiadakis, Georg Lippitsch, Göran Jönsson, Guido Günther,
+Guillaume Desmottes, Guiqin Zou, Haakon Sporsheim, Haelwenn (lanodan)
+Monnier, Haihao Xiang, Haihua Hu, Havard Graff, He Junyan, Helmut
+Januschka, Henry Wilkes, Hosang Lee, Hou Qi, Ignacio Casal Quinteiro,
+Igor Kovalenko, Ilya Kreymer, Imanol Fernandez, Jacek Tomaszewski, Jade
+Macho, Jakub Adam, Jakub Janků, Jan Alexander Steffens (heftig), Jan
+Schmidt, Jason Carrete, Jason Pereira, Jay Douglass, Jeongki Kim, Jérôme
+Laheurte, Jimmi Holst Christensen, Johan Sternerup, John Hassell, John
+Lindgren, John-Mark Bell, Jonathan Matthew, Jordan Petridis, Jose
+Quaresma, Julian Bouzas, Julien, Kai Uwe Broulik, Kasper Steensig
+Jensen, Kellermann Axel, Kevin Song, Khem Raj, Knut Inge Hvidsten, Knut
+Saastad, Kristofer Björkström, Lars Lundqvist, Lawrence Troup, Lim Siew
+Hoon, Lucas Stach, Ludvig Rappe, Luis Paulo Fernandes de Barros, Luke
+Yelavich, Mads Buvik Sandvei, Marc Leeman, Marco Felsch, Marek Vasut,
+Marian Cichy, Marijn Suijten, Marius Vlad, Markus Ebner, Mart Raudsepp,
+Matej Knopp, Mathieu Duponchelle, Matthew Waters, Matthieu De Beule,
+Mengkejiergeli Ba, Michael de Gans, Michael Olbrich, Michael Tretter,
+Michal Dzik, Miguel Paris, Mikhail Fludkov, mkba, Nazar Mokrynskyi,
+Nicholas Jackson, Nicola Murino, Nicolas Dufresne, Niklas Hambüchen,
+Nikolay Sivov, Nirbheek Chauhan, Olivier Blin, Olivier Crete, Olivier
+Crête, Paul Goulpié, Per Förlin, Peter Boba, P H, Philippe Normand,
+Philipp Zabel, Pieter Willem Jordaan, Piotrek Brzeziński, Rafał
+Dzięgiel, Rafostar, raghavendra, Raghavendra, Raju Babannavar, Raleigh
+Littles III, Randy Li, Randy Li (ayaka), Ratchanan Srirattanamet, Raul
+Tambre, reed.lawrence, Ricky Tang, Robert Rosengren, Robert Swain, Robin
+Burchell, Roman Sivriver, R S Nikhil Krishna, Ruben Gonzalez, Ruslan
+Khamidullin, Sanchayan Maity, Scott Moreau, Sebastian Dröge, Sergei
+Kovalev, Seungha Yang, Sid Sethupathi, sohwan.park, Sonny Piers, Staz M,
+Stefan Brüns, Stéphane Cerveau, Stephan Hesse, Stian Selnes, Stirling
+Westrup, Théo MAILLART, Thibault Saunier, Tim, Timo Wischer, Tim-Philipp
+Müller, Tim Schneider, Tobias Ronge, Tom Schoonjans, Tulio Beloqui,
+tyler-aicradle, U. Artie Eoff, Ung, Val Doroshchuk, VaL Doroshchuk,
+Víctor Manuel Jáquez Leal, Vivek R, Vivia Nikolaidou, Vivienne
+Watermeier, Vladimir Menshakov, Will Miller, Wim Taymans, Xabier
+Rodriguez Calvar, Xavier Claessens, Xℹ Ruoyao, Yacine Bandou, Yinhang
+Liu, youngh.lee, youngsoo.lee, yychao, Zebediah Figura, Zhang yuankun,
+Zhang Yuankun, Zhao, Zhao Zhili, , Aleksandar Topic, Antonio Ospite,
+Bastien Nocera, Benjamin Gaignard, Brad Hards, Carlos Falgueras García,
+Célestin Marot, Corentin Damman, Corentin Noël, Daniel Almeida, Daniel
+Knobe, Danny Smith, Dave Piché, Dmitry Osipenko, Fabrice Fontaine,
+fjmax, Florian Zwoch, Guillaume Desmottes, Haihua Hu, Heinrich Kruger,
+He Junyan, Jakub Adam, James Cowgill, Jan Alexander Steffens (heftig),
+Jean Felder, Jeongki Kim, Jiri Uncovsky, Joe Todd, Jordan Petridis,
+Krystian Wojtas, Marc-André Lureau, Marcin Kolny, Marc Leeman, Mark
+Nauwelaerts, Martin Reboredo, Mathieu Duponchelle, Matthew Waters,
+Mengkejiergeli Ba, Michael Gruner, Nicolas Dufresne, Nirbheek Chauhan,
+Olivier Crête, Philippe Normand, Rafał Dzięgiel, Ralf Sippl, Robert
+Mader, Sanchayan Maity, Sangchul Lee, Sebastian Dröge, Seungha Yang,
+Stéphane Cerveau, Teh Yule Kim, Thibault Saunier, Thomas Klausner, Timo
+Wischer, Tim-Philipp Müller, Tobias Reineke, Tomasz Andrzejak, Trung Do,
+Tyler Compton, Ung, Víctor Manuel Jáquez Leal, Vivia Nikolaidou, Wim
+Taymans, wngecn, Wonchul Lee, wuchang li, Xavier Claessens, Xi Ruoyao,
+Yoshiharu Hirose, Zhao,
 
 … and many others who have contributed bug reports, translations, sent
 suggestions or helped testing.
@@ -307,7 +1863,7 @@ the git 1.20 branch, which will be a stable branch.
 
 1.20.0
 
-1.20.0 is scheduled to be released around October/November 2021.
+1.20.0 is scheduled to be released around early February 2022.
 
 Schedule for 1.22
 
@@ -315,7 +1871,9 @@ Our next major feature release will be 1.22, and 1.21 will be the
 unstable development version leading up to the stable 1.22 release. The
 development of 1.21/1.22 will happen in the git main branch.
 
-The plan for the 1.22 development cycle is yet to be confirmed.
+The plan for the 1.22 development cycle is yet to be confirmed. Assuming
+no major project-wide reorganisations in the 1.22 cycle we might try and
+aim for a release around August 2022.
 
 1.22 will be backwards-compatible to the stable 1.20, 1.18, 1.16, 1.14,
 1.12, 1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series.
@@ -323,6 +1881,7 @@ The plan for the 1.22 development cycle is yet to be confirmed.
 ------------------------------------------------------------------------
 
 These release notes have been prepared by Tim-Philipp Müller with
-contributions from …
+contributions from Matthew Waters, Nicolas Dufresne, Nirbheek Chauhan,
+Sebastian Dröge and Seungha Yang.
 
 License: CC BY-SA 4.0
diff --git a/subprojects/gst-omx/RELEASE b/subprojects/gst-omx/RELEASE
index 4bf15720ea..b5aa9980f7 100644
--- a/subprojects/gst-omx/RELEASE
+++ b/subprojects/gst-omx/RELEASE
@@ -1,4 +1,4 @@
-This is GStreamer gst-omx 1.19.3.
+This is GStreamer gst-omx 1.19.90.
 
 GStreamer 1.19 is the development branch leading up to the next major
 stable version which will be 1.20.
diff --git a/subprojects/gst-omx/gst-omx.doap b/subprojects/gst-omx/gst-omx.doap
index 6b4b3f3147..e11d11723c 100644
--- a/subprojects/gst-omx/gst-omx.doap
+++ b/subprojects/gst-omx/gst-omx.doap
@@ -33,6 +33,16 @@ a basic collection of elements
 
  <release>
   <Version>
+   <revision>1.19.90</revision>
+   <branch>main</branch>
+   <name></name>
+   <created>2022-01-28</created>
+   <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-omx/gst-omx-1.19.90.tar.xz" />
+  </Version>
+ </release>
+
+ <release>
+  <Version>
    <revision>1.19.3</revision>
    <branch>main</branch>
    <name></name>
diff --git a/subprojects/gst-omx/meson.build b/subprojects/gst-omx/meson.build
index c0ad4eaa85..2fd4f5df6c 100644
--- a/subprojects/gst-omx/meson.build
+++ b/subprojects/gst-omx/meson.build
@@ -1,5 +1,5 @@
 project('gst-omx', 'c',
-  version : '1.19.3.1',
+  version : '1.19.90',
   meson_version : '>= 0.59',
   default_options : [ 'warning_level=1',
                       'buildtype=debugoptimized' ])
diff --git a/subprojects/gst-plugins-bad/ChangeLog b/subprojects/gst-plugins-bad/ChangeLog
index a2257d9693..722afdd8ea 100644
--- a/subprojects/gst-plugins-bad/ChangeLog
+++ b/subprojects/gst-plugins-bad/ChangeLog
@@ -1,3 +1,18 @@
+=== release 1.19.90 ===
+
+2022-01-28 14:28:35 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* NEWS:
+	* RELEASE:
+	* gst-plugins-bad.doap:
+	* meson.build:
+	  Release 1.19.90
+
+2022-01-28 14:28:28 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* ChangeLog:
+	  Update ChangeLogs for 1.19.90
+
 2022-01-27 11:22:54 +0100  Víctor Manuel Jáquez Leal <vjaquez@igalia.com>
 
 	* sys/va/gstvavpp.c:
diff --git a/subprojects/gst-plugins-bad/NEWS b/subprojects/gst-plugins-bad/NEWS
index e9f5227c15..1a512c1848 100644
--- a/subprojects/gst-plugins-bad/NEWS
+++ b/subprojects/gst-plugins-bad/NEWS
@@ -1,23 +1,19 @@
 GStreamer 1.20 Release Notes
 
 GStreamer 1.20 has not been released yet. It is scheduled for release in
-November 2021.
+late January / early February 2022.
 
 1.19.x is the unstable development version that is being developed in
-the git main branch and which will eventually result in 1.20, and 1.19.3
-is the current development release in that series
-
-It is expected that feature freeze will be in early October 2021,
-followed by one or two 1.19.9x pre-releases and the new 1.20 stable
-release around October/November 2021.
+the git main branch and which will eventually result in 1.20, and
+1.19.90 is the first release candidate in that series (1.20rc1).
 
 1.20 will be backwards-compatible to the stable 1.18, 1.16, 1.14, 1.12,
-1.10, 1.8, 1.6,, 1.4, 1.2 and 1.0 release series.
+1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series.
 
 See https://gstreamer.freedesktop.org/releases/1.20/ for the latest
 version of this document.
 
-Last updated: Monday 1 November 2021, 01:00 UTC (log)
+Last updated: Wednesday 26 January 2022, 01:00 UTC (log)
 
 Introduction
 
@@ -30,25 +26,971 @@ fixes and other improvements.
 
 Highlights
 
--   this section will be completed in due course
+-   Development in GitLab was switched to a single git repository
+    containing all the modules
+-   GstPlay: new high-level playback library, replaces GstPlayer
+-   WebM Alpha decoding support
+-   Encoding profiles can now be tweaked with additional
+    application-specified element properties
+-   Compositor: multi-threaded video conversion and mixing
+-   RTP header extensions: unified support in RTP depayloader and
+    payloader base classes
+-   SMPTE 2022-1 2-D Forward Error Correction support
+-   Smart encoding (passthrough) support for VP8, VP9, H.265 in
+    encodebin and transcodebin
+-   Runtime compatibility support for libsoup2 and libsoup3 (libsoup3
+    support experimental)
+-   Video decoder subframe support
+-   Video decoder automatic packet-loss, data corruption, and keyframe
+    request handling for RTP / WebRTC / RTSP
+-   MP4 and Matroska muxers now support profile/level/resolution changes
+    for H264/H265 input streams (i.e. codec data changing on the fly)
+-   MP4 muxing mode that initially creates a fragmented mp4 which is
+    converted to a regular mp4 on EOS
+-   Audio support for the WebKit Port for Embedded (WPE) web page source
+    element
+-   CUDA based video color space convert and rescale elements and
+    upload/download elements
+-   NVIDIA memory:NVMM support for OpenGL glupload and gldownload
+    elements
+-   Many WebRTC improvements
+-   The new VA-API plugin implemention fleshed out with more decoders
+    and new postproc elements
+-   AppSink API to retrieve events in addition to buffers and buffer
+    lists
+-   AppSrc gained more configuration options for the internal queue
+    (leakiness, limits in buffers and time, getters to read current
+    levels)
+-   Updated Rust bindings and many new Rust plugins
+-   Improved support for custom minimal GStreamer builds
+-   Support build against FFmpeg 5.0
+-   Linux Stateless CODEC support gained MPEG2 and VP9
+-   Windows Direct3D11/DXVA decoder gained AV1 and MPEG2 support
+-   Lots of new plugins, features, performance improvements and bug
+    fixes
 
 Major new features and changes
 
 Noteworthy new features and API
 
--   this section will be filled in in due course
+-   gst_element_get_request_pad() has been deprecated in favour of the
+    newly-added gst_element_request_pad_simple() which does the exact
+    same thing but has a less confusing name that hopefully makes clear
+    that the function request a new pad rather than just retrieves an
+    already-existing request pad.
+
+Development in GitLab was switched to a single git repository containing all the modules
+
+The GStreamer multimedia framework is a set of libraries and plugins
+split into a number of distinct modules which are released independently
+and which have so far been developed in separate git repositories in
+freedesktop.org GitLab.
+
+In addition to these separate git repositories there was a gst-build
+module that would use the Meson build systems’s subproject feature to
+download each individual module and then build everything in one go. It
+would also provide an uninstalled development environment that made it
+easy to work on GStreamer and use or test versions other than the
+system-installed GStreamer version.
+
+All of these modules have now (as of 28 September 2021) been merged into
+a single git repository (“Mono repository” or “monorepo”) which should
+simplify development workflows and continuous integration, especially
+where changes need to be made to multiple modules at once.
+
+This mono repository merge will primarily affect GStreamer developers
+and contributors and anyone who has workflows based on the GStreamer git
+repositories.
+
+The Rust bindings and Rust plugins modules have not been merged into the
+mono repository at this time because they follow a different release
+cycle.
+
+The mono repository lives in the existing GStreamer core git repository
+in GitLab in the new main branch and all future development will happen
+on this branch.
+
+Modules will continue to be released as separate tarballs.
+
+For more details, please see the GStreamer mono repository FAQ.
+
+GstPlay: new high-level playback library replacing GstPlayer
+
+-   GstPlay is a new high-level playback library that replaces the older
+    GstPlayer API. It is basically the same API as GstPlayer but
+    refactored to use bus messages for application notifications instead
+    of GObject signals. There is still a signal adapter object for those
+    who prefer signals. Since the existing GstPlayer API is already in
+    use in various applications, it didn’t seem like a good idea to
+    break it entirely. Instead a new API was added, and it is expected
+    that this new GstPlay API will be moved to gst-plugins-base in
+    future.
+
+-   The existing GstPlayer API is scheduled for deprecation and will be
+    removed at some point in the future (e.g. in GStreamer 1.24), so
+    application developers are urged to migrate to the new GstPlay API
+    at their earliest convenience.
+
+WebM alpha decoding
+
+-   Implement WebM alpha decoding (VP8/VP9 with alpha), which required
+    support and additions in various places. This is supported both with
+    software decoders and hardware-accelerated decoders.
+
+-   VP8/VP9 don’t support alpha components natively in the codec, so the
+    way this is implemented in WebM is by encoding the alpha plane with
+    transparency data as a separate VP8/VP9 stream. Inside the WebM
+    container (a variant of Matroska) this is coded as a single video
+    track with the “normal” VP8/VP9 video data making up the main video
+    data and each frame of video having an encoded alpha frame attached
+    to it as extra data ("BlockAdditional").
+
+-   matroskademux has been extended extract this per-frame alpha side
+    data and attach it in form of a GstVideoCodecAlphaMeta to the
+    regular video buffers. Note that this new meta is specific to this
+    VP8/VP9 alpha support and can’t be used to just add alpha support to
+    other codecs that don’t support it. Lastly, matroskademux also
+    advertises the fact that the streams contain alpha in the caps.
+
+-   The new codecalpha plugin contains various bits of infrastructure to
+    support autoplugging and debugging:
+
+    -   codecalphademux splits out the alpha stream from the metas on
+        the regular VP8/VP9 buffers
+    -   alphacombine takes two decoded raw video streams (one alpha, one
+        the regular video) and combines it into a video stream with
+        alpha
+    -   vp8alphadecodebin + vp9alphadecodebin are wrapper bins that use
+        the regular vp8dec and vp9dec software decoders to decode
+        regular and alpha streams and combine them again. To decodebin
+        these look like regular decoders which ju
+    -   The V4L2 CODEC plugin has stateless VP8/VP9 decoders that can
+        decode both alpha and non-alpha stream with a single decoder
+        instance
+
+-   A new AV12 video format was added which is basically NV12 with an
+    alpha plane, which is more convenient for many hardware-accelerated
+    decoders.
+
+-   Watch Nicolas Dufresne’s LCA 2022 talk “Bringing WebM Alpha support
+    to GStreamer” for all the details and a demo.
+
+RTP Header Extensions Base Class and Automatic Header Extension Handling in RTP Payloaders and Depayloaders
+
+-   RTP Header Extensions are specified in RFC 5285 and provide a way to
+    add small pieces of data to RTP packets in between the RTP header
+    and the RTP payload. This is often used for per-frame metadata,
+    extended timestamps or other application-specific extra data. There
+    are several commonly-used extensions specified in various RFCs, but
+    senders are free to put any kind of data in there, as long as sender
+    and receiver both know what that data is. Receivers that don’t know
+    about the header extensions will just skip the extra data without
+    ever looking at it. These header extensions can often be combined
+    with any kind of payload format, so may need to be supported by many
+    RTP payloader and depayloader elements.
+
+-   Inserting and extracting RTP header extension data has so far been a
+    bit inconvenient in GStreamer: There are functions to add and
+    retrieve RTP header extension data from RTP packets, but nothing
+    works automatically, even for common extensions. People would have
+    to do the insertion/extraction either in custom elements
+    before/after the RTP payloader/depayloader, or inside pad probes,
+    which isn’t very nice.
+
+-   This release adds various pieces of new infrastructure for generic
+    RTP header extension handling, as well as some implementations for
+    common extensions:
+
+    -   GstRTPHeaderExtension is a new helper base class for reading and
+        writing RTP header extensions. Nominally this subclasses
+        GstElement, but only so these extensions are stored in the
+        registry where they can be looked up by URI or name. They don’t
+        have pads and don’t get added to the pipeline graph as an
+        element.
+
+    -   "add-extension" and "clear-extension" action signals on RTP
+        payloaders and depayloaders for manual extension management
+
+    -   The "request-extension" signal will be emitted if an extension
+        is encountered that requires explicit mapping by the application
+
+    -   new "auto-header-extension" property on RTP payloaders and
+        depayloaders for automatic handling of known header extensions.
+        This is enabled by default. The extensions must be signalled via
+        caps / SDP.
+
+    -   RTP header extension implementations:
+
+        -   rtphdrextclientaudiolevel: Client-to-Mixer Audio Level
+            Indication (RFC 6464) (also see below)
+        -   rtphdrextcolorspace: Color Space extension, extends RTP
+            packets with color space and high dynamic range (HDR)
+            information
+        -   rtphdrexttwcc: Transport Wide Congestion Control support
+
+-   gst_rtp_buffer_remove_extension_data() is a new helper function to
+    remove an RTP header extension from an RTP buffer
+
+-   The existing gst_rtp_buffer_set_extension_data() now also supports
+    shrinking the extension data in size
+
+AppSink and AppSrc improvements
+
+-   appsink: new API to pull events out of appsink in addition to
+    buffers and buffer lists.
+
+    There was previously no way for users to receive incoming events
+    from appsink properly serialised with the data flow, even if they
+    are serialised events. The reason for that is that the only way to
+    intercept events was via a pad probe on the appsink sink pad, but
+    there is also internal queuing inside of appsink, so it’s difficult
+    to ascertain the right order of everything in all cases.
+
+    There is now a new "new-serialized-event" signal which will be
+    emitted when there’s a new event pending (just like the existing
+    "new-sample" signal). The "emit-signals" property must be set to
+    TRUE in order to activate this (but it’s also fine to just pull from
+    the application thread without using the signals).
+
+    gst_app_sink_pull_object() and gst_app_sink_try_pull_object() can be
+    used to pull out either an event or a new sample carrying a buffer
+    or buffer list, whatever is next in the queue.
+
+    EOS events will be filtered and will not be returned. EOS handling
+    can be done the ususal way, same as with _pull_sample().
+
+-   appsrc: allow configuration of internal queue limits in time and
+    buffers and add leaky mode.
+
+    There is internal queuing inside appsrc so the application thread
+    can push data into the element which will then be picked up by the
+    source element’s streaming thread and pushed into the pipeline from
+    that streaming thread. This queue is unlimited by default and until
+    now it was only possible to set a maximum size limit in bytes. When
+    that byte limit is reached, the pushing thread (application thread)
+    would be blocked until more space becomes available.
+
+    A limit in bytes is not particularly useful for many use cases, so
+    now it is possible to also configure limits in time and buffers
+    using the new "max-time" and "max-buffers" properties. Of course
+    there are also matching new read-only"current-level-buffers" and
+    "current-level-time properties" properties to query the current fill
+    level of the internal queue in time and buffers.
+
+    And as if that wasn’t enough the internal queue can also be
+    configured as leaky using the new "leaky-type" property. That way
+    when the queue is full the application thread won’t be blocked when
+    it tries to push in more data, but instead either the new buffer
+    will be dropped or the oldest data in the queue will be dropped.
+
+Better string serialization of nested GstCaps and GstStructures
+
+-   New string serialisation format for structs and caps that can handle
+    nested structs and caps properly by using brackets to delimit nested
+    items (e.g. some-struct, some-field=[nested-struct, nested=true]).
+    Unlike the default format the new variant can also support more than
+    one level of nesting. For backwards-compatibility reasons the old
+    format is still output by default when serialising caps and structs
+    using the existing API. The new functions gst_caps_serialize() and
+    gst_structure_serialize() can be used to output strings in the new
+    format.
+
+Convenience API for custom GstMetas
+
+-   New convenience API to register and create custom GstMetas:
+    gst_meta_register_custom() and gst_buffer_add_custom_meta(). Such
+    custom meta is backed by a GstStructure and does not require that
+    users of the API expose their GstMeta implementation as public API
+    for other components to make use of it. In addition, it provides a
+    simpler interface by ignoring the impl vs. api distinction that the
+    regular API exposes. This new API is meant to be the meta
+    counterpart to custom events and messages, and to be more convenient
+    than the lower-level API when the absolute best performance isn’t a
+    requirement. The reason it’s less performant than a “proper” meta is
+    that a proper meta is just a C struct in the end whereas this goes
+    through the GstStructure API which has a bit more overhead, which
+    for most scenarios is negligible however. This new API is useful for
+    experimentation or proprietary metas, but also has some limitations:
+    it can only be used if there’s a single producer of these metas;
+    it’s not allowed to register the same custom meta multiple times or
+    from multiple places.
+
+Additional Element Properties on Encoding Profiles
+
+-   GstEncodingProfile: The new "element-properties" and
+    gst_encoding_profile_set_element_properties() API allows
+    applications to set additional element properties on encoding
+    profiles to configure muxers and encoders. So far the encoding
+    profile template was the only place where this could be specified,
+    but often what applications want to do is take a ready-made encoding
+    profile shipped by GStreamer or the application and then tweak the
+    settings on top of that, which is now possible with this API. Since
+    applications can’t always know in advance what encoder element will
+    be used in the end, it’s even possible to specify properties on a
+    per-element basis.
+
+    Encoding Profiles are used in the encodebin, transcodebin and
+    camerabin elements and APIs to configure output formats (containers
+    and elementary streams).
+
+Audio Level Indication Meta for RFC 6464
+
+-   New GstAudioLevelMeta containing Audio Level Indication as per RFC
+    6464
+
+-   The level element has been updated to add GstAudioLevelMeta on
+    buffers if the "audio-level-meta" property is set to TRUE. This can
+    then in turn be picked up by RTP payloaders to signal the audio
+    level to receivers through RTP header extensions (see above).
+
+-   New Client-to-Mixer Audio Level Indication (RFC6464) RTP Header
+    Extension which should be automatically created and used by RTP
+    payloaders and depayloaders if their "auto-header-extension"
+    property is enabled and if the extension is part of the RTP caps.
+
+Automatic packet loss, data corruption and keyframe request handling for video decoders
+
+-   The GstVideoDecoder base class has gained various new APIs to
+    automatically handle packet loss and data corruption better by
+    default, especially in RTP, RTSP and WebRTC streaming scenarios, and
+    to give subclasses more control about how they want to handle
+    missing data:
+
+    -   Video decoder subclasses can mark output frames as corrupted via
+        the new GST_VIDEO_CODEC_FRAME_FLAG_CORRUPTED flag
+
+    -   A new "discard-corrupted-frames" property allows applications to
+        configure decoders so that corrupted frames are directly
+        discarded instead of being forwarded inside the pipeline. This
+        is a replacement for the "output-corrupt" property of the FFmpeg
+        decoders.
+
+    -   RTP depayloaders can now signal to decoders that data is missing
+        when sending GAP events for lost packets. GAP events can be sent
+        for various reason in a GStreamer pipeline. Often they are just
+        used to let downstream elements know that there isn’t a buffer
+        available at the moment, so downstream elements can move on
+        instead of waiting for one. They are also sent by RTP
+        depayloaders in the case that packets are missing, however, and
+        so far a decoder was not able to differentiate the two cases.
+        This has been remedied now: GAP events can be decorated with
+        gst_event_set_gap_flags() and GST_GAP_FLAG_MISSING_DATA to let
+        decoders now what happened, and decoders can then use that in
+        some cases to handle missing data better.
+
+    -   The GstVideoDecoder::handle_missing_data vfunc was added to
+        inform subclasses about packet loss or missing data and let them
+        handle it in their own way if they like.
+
+    -   gst_video_decoder_set_needs_sync_point() lets subclasses signal
+        that they need the stream to start with a sync point. If
+        enabled, the base class will discard all non-sync point frames
+        in the beginning and after a flush and does not pass them to the
+        subclass. Furthermore, if the first frame is not a sync point,
+        the base class will try and request a sync frame from upstream
+        by sending a force-key-unit event (see next items).
+
+    -   New "automatic-request-sync-points" and
+        "automatic-request-sync-point-flags" properties to automatically
+        request sync points when needed, e.g. on packet loss or if the
+        first frame is not a keyframe. Applications may want to enable
+        this on decoders operating in e.g. RTP/WebRTC/RTSP receiver
+        pipelines.
+
+    -   The new "min-force-key-unit-interval" property can be used to
+        ensure there’s a minimal interval between keyframe requests to
+        upstream (and/or the sender) and we’re not flooding the sender
+        with key unit requests.
+
+    -   gst_video_decoder_request_sync_point() allows subclasses to
+        request a new sync point (e.g. if they choose to do their own
+        missing data handling). This will still honour the
+        "min-force-key-unit-interval" property if set.
+
+Improved support for custom minimal GStreamer builds
+
+-   Element registration and registration of other plugin features
+    inside plugin init functions has been improved in order to
+    facilitate minimal custom GStreamer builds.
+
+-   A number of new macros have been added to declare and create
+    per-element and per-pluginfeature register functions in all plugins,
+    and then call those from the per-plugin plugin_init functions:
+
+    -   GST_ELEMENT_REGISTER_DEFINE,
+        GST_DEVICE_PROVIDER_REGISTER_DEFINE,
+        GST_DYNAMIC_TYPE_REGISTER_DEFINE, GST_TYPE_FIND_REGISTER_DEFINE
+        for the actual registration call with GStreamer
+    -   GST_ELEMENT_REGISTER, GST_DEVICE_PROVIDER_REGISTER,
+        GST_DYNAMIC_TYPE_REGISTER, GST_PLUGIN_STATIC_REGISTER,
+        GST_TYPE_FIND_REGISTER to call the registration function defined
+        by the REGISTER_DEFINE macro
+    -   GST_ELEMENT_REGISTER_DECLARE,
+        GST_DEVICE_PROVIDER_REGISTER_DECLARE,
+        GST_DYNAMIC_TYPE_REGISTER_DECLARE,
+        GST_TYPE_FIND_REGISTER_DECLARE to declare the registration
+        function defined by the REGISTER_DEFINE macro
+    -   and various variants for advanced use cases.
+
+-   This means that applications can call the per-element and
+    per-pluginfeature registration functions for only the elements they
+    need instead of registering plugins as a whole with all kinds of
+    elements that may not be required (e.g. encoder and decoder instead
+    of just decoder). In case of static linking all unused functions and
+    their dependencies would be removed in this case by the linker,
+    which helps minimise binary size for custom builds.
+
+-   gst_init() will automatically call a gst_init_static_plugins()
+    function if one exists.
+
+-   See the GStreamer static build documentation and Stéphane’s blog
+    post Generate a minimal GStreamer build, tailored to your needs for
+    more details.
 
 New elements
 
--   this section will be filled in in due course
+-   New aesdec and aesenc elements for AES encryption and decryption in
+    a custom format.
+
+-   New encodebin2 element with dynamic/sometimes source pads in order
+    to support the option of doing the muxing outside of encodebin,
+    e.g. in combination with a splitmuxsink.
+
+-   New fakeaudiosink and videocodectestsink elements for testing and
+    debugging (see below for more details)
+
+-   rtpisacpay, rtpisacdepay: new RTP payloader and depayloader for iSAC
+    audio codec
+
+-   rtpst2022-1-fecdec, rtpst2022-1-fecenc: new elements providing SMPTE
+    2022-1 2-D Forward Error Correction. More details in Mathieu’s blog
+    post.
+
+-   isac: new plugin wrapping the Internet Speech Audio Codec reference
+    encoder and decoder from the WebRTC project.
+
+-   asio: plugin for Steinberg ASIO (Audio Streaming Input/Output) API
+
+-   gssrc, gssink: add source and sink for Google Cloud Storage
+
+-   onnx: new plugin to apply ONNX neural network models to video
+
+-   openaptx: aptX and aptX-HD codecs using libopenaptx (v0.2.0)
+
+-   qroverlay, debugqroverlay: new elements that allows overlaying data
+    on top of video in form of a QR code
+
+-   cvtracker: new OpenCV-based tracker element
+
+-   av1parse, vp9parse: new parsers for AV1 and VP9 video
+
+-   va: work on the new VA-API plugin implementation for
+    hardware-accelerated video decoding and encoding has continued at
+    pace, with various new decoders and filters having joined the
+    initial vah264dec:
+
+    -   vah265dec: VA-API H.265 decoder
+    -   vavp8dec: VA-API VP8 decoder
+    -   vavp9dec: VA-API VP9 decoder
+    -   vaav1dec: VA-API AV1 decoder
+    -   vampeg2dec: VA-API MPEG-2 decoder
+    -   vadeinterlace: : VA-API deinterlace filter
+    -   vapostproc: : VA-API postproc filter (color conversion,
+        resizing, cropping, color balance, video rotation, skin tone
+        enhancement, denoise, sharpen)
+
+    See Víctor’s blog post “GstVA in GStreamer 1.20” for more details
+    and what’s coming up next.
+
+-   vaapiav1dec: new AV1 decoder element (in gstreamer-vaapi)
+
+-   msdkav1dec: hardware-accelerated AV1 decoder using the Intel Media
+    SDK / oneVPL
+
+-   nvcodec plugin for NVIDIA NVCODEC API for hardware-accelerated video
+    encoding and decoding:
+
+    -   cudaconvert, cudascale: new CUDA based video color space convert
+        and rescale elements
+    -   cudaupload, cudadownload: new helper elements for memory
+        transfer between CUDA and system memory spaces
+    -   nvvp8sldec, nvvp9sldec: new GstCodecs-based VP8/VP9 decoders
+
+-   Various new hardware-accelerated elements for Windows:
+
+    -   d3d11screencapturesrc: new desktop capture element, including a
+        GstDeviceProvider implementation to enumerate/select target
+        monitors for capture.
+    -   d3d11av1dec and d3d11mpeg2dec: AV1 and MPEG-2 decoders
+    -   d3d11deinterlace: deinterlacing filter
+    -   d3d11compositor: video composing element
+    -   see Windows section below for more details
+
+-   new Rust plugins:
+
+    -   audiornnoise: Removes noise from an audio stream
+    -   awstranscribeparse: Parses AWS audio transcripts into timed text
+        buffers
+    -   ccdetect: Detects if valid closed captions are present in a
+        closed captions stream
+    -   cea608tojson: Converts CEA-608 Closed Captions to a JSON
+        representation
+    -   cmafmux: CMAF fragmented MP4 muxer
+    -   dashmp4mux: DASH fragmented MP4 muxer
+    -   isofmp4mux: ISO fragmented MP4 muxer
+    -   ebur128level: EBU R128 Loudness Level Measurement
+    -   ffv1dec: FFV1 video decoder
+    -   gtk4paintablesink: GTK4 video sink, which provides a
+        GdkPaintable that can be rendered in various widgets
+    -   hlssink3: HTTP Live Streaming sink
+    -   hrtfrender: Head-Related Transfer Function (HRTF) renderer
+    -   hsvdetector: HSV colorspace detector
+    -   hsvfilter: HSV colorspace filter
+    -   jsongstenc: Wraps buffers containing any valid top-level JSON
+        structures into higher level JSON objects, and outputs those as
+        ndjson
+    -   jsongstparse: Parses ndjson as output by jsongstenc
+    -   jsontovtt: converts JSON to WebVTT subtitles
+    -   regex: Applies regular expression operations on text
+    -   roundedcorners: Adds rounded corners to video
+    -   spotifyaudiosrc: Spotify source
+    -   textahead: Display upcoming text buffers ahead (e.g. for
+        Karaoke)
+    -   transcriberbin: passthrough bin that transcribes raw audio to
+        closed captions using awstranscriber and puts the captions as
+        metas onto the video
+    -   tttojson: Converts timed text to a JSON representation
+    -   uriplaylistbin: Playlist source bin
+    -   webpdec-rs: WebP image decoder with animation support
+
+-   New plugin codecalpha with elements to assist with WebM Alpha
+    decoding
+
+    -   codecalphademux: Split stream with GstVideoCodecAlphaMeta into
+        two streams
+    -   alphacombine: Combine two raw video stream (I420 or NV12) as one
+        stream with alpha channel (A420 or AV12)
+    -   vp8alphadecodebin: A bin to handle software decoding of VP8 with
+        alpha
+    -   vp9alphadecodebin: A bin to handle software decoding of VP9 with
+        alpha
+
+-   New hardware accelerated elements for Linux:
+
+    -   v4l2slmpeg2dec: Support for Linux Stateless MPEG2 decoders
+    -   v4l2slvp9dec: Support for Linux Stateless VP9 decoders
+    -   v4l2slvp8alphadecodebin: Support HW accelerated VP8 with alpha
+        layer decoding
+    -   v4l2slvp9alphadecodebin: Support HW accelerated VP9 with alpha
+        layer decoding
 
 New element features and additions
 
--   this section will be filled in in due course
+-   assrender: handle more font mime types; better interaction with
+    matroskademux for embedded fonts
+
+-   audiobuffersplit: Add support for specifying output buffer size in
+    bytes (not just duration)
+
+-   audiolatency: new "samplesperbuffer" property so users can configure
+    the number of samples per buffer. The default value is 240 samples
+    which is equivalent to 5ms latency with a sample rate of 48000,
+    which might be larger than actual buffer size of audio capture
+    device.
+
+-   audiomixer, audiointerleave, GstAudioAggregator: now keep a count of
+    samples that are dropped or processed as statistic and can be made
+    to post QoS messages on the bus whenever samples are dropped by
+    setting the "qos-messages" property on input pads.
+
+-   audiomixer, compositor: improved handling of new inputs added at
+    runtime. New API was added to the GstAggregator base class to allow
+    subclasses to opt into an aggregation mode where inactive pads are
+    ignored when processing input buffers
+    (gst_aggregator_set_ignore_inactive_pads(),
+    gst_aggregator_pad_is_inactive()). An “inactive pad” in this context
+    is a pad which, in live mode, hasn’t yet received a first buffer,
+    but has been waited on at least once. What would happen usually in
+    this case is that the aggregator would wait for data on this pad
+    every time, up to the maximum configured latency. This would
+    inadvertently push mixer elements in live mode to the configured
+    latency envelope and delay processing when new inputs are added at
+    runtime until these inputs have actually produced data. This is
+    usually undesirable. With this new API, new inputs can be added
+    (requested) and configured and they won’t delay the data processing.
+    Applications can opt into this new behaviour by setting the
+    "ignore-inactive-pads" property on compositor, audiomixer or other
+    GstAudioAggregator-based elements.
+
+-   cccombiner: implement “scheduling” of captions. So far cccombiner’s
+    behaviour was essentially that of a funnel: it strictly looked at
+    input timestamps to associate together video and caption buffers.
+    Now it will try to smoothly schedule caption buffers in order to
+    have exactly one per output video buffer. This might involve
+    rewriting input captions, for example when the input is CDP then
+    sequence counters are rewritten, time codes are dropped and
+    potentially re-injected if the input video frame had a time code
+    meta. This can also lead to the input drifting from synchronization,
+    when there isn’t enough padding in the input stream to catch up. In
+    that case the element will start dropping old caption buffers once
+    the number of buffers in its internal queue reaches a certain limit
+    (configurable via the "max-scheduled" property). The new original
+    funnel-like behaviour can be restored by setting the "scheduling"
+    property to FALSE.
+
+-   ccconverter: new "cdp-mode" property to specify which sections to
+    include in CDP packets (timecode, CC data, service info). Various
+    software, including ffmpeg’s Decklink support, fails parsing CDP
+    packets that contain anything but CC data in the CDP packets.
+
+-   clocksync: new "sync-to-first" property for automatic timestamp
+    offset setup: if set clocksync will set up the "ts-offset" value
+    based on the first buffer and the pipeline’s running time when the
+    first buffer arrived. The newly configured "ts-offset" in this case
+    would be the value that allows outputting the first buffer without
+    waiting on the clock. This is useful for example to feed a non-live
+    input into an already-running pipeline.
+
+-   compositor:
+
+    -   multi-threaded input conversion and compositing. Set the
+        "max-threads" property to activate this.
+    -   new "sizing-policy" property to support display aspect ratio
+        (DAR)-aware scaling. By default the image is scaled to fill the
+        configured destination rectangle without padding and without
+        keeping the aspect ratio. With sizing-policy=keep-aspect-ratio
+        the input image is scaled to fit the destination rectangle
+        specified by GstCompositorPad:{xpos, ypos, width, height}
+        properties preserving the aspect ratio. As a result, the image
+        will be centered in the destination rectangle with padding if
+        necessary.
+    -   new "zero-size-is-unscaled" property on input pads. By default
+        pad width=0 or pad height=0 mean that the stream should not be
+        scaled in that dimension. But if the "zero-size-is-unscaled"
+        property is set to FALSE a width or height of 0 is instead
+        interpreted to mean that the input image on that pad should not
+        be composited, which is useful when creating animations where an
+        input image is made smaller and smaller until it disappears.
+    -   improved handling of new inputs at runtime via
+        "ignore-inactive-pads"property (see above for details)
+    -   allow output format with alpha even if none of the inputs have
+        alpha (also glvideomixer and other GstVideoAggregator
+        subclasses)
+
+-   dashsink: add h265 codec support and signals for allowing custom
+    playlist/fragment output
+
+-   decodebin3:
+
+    -   improved decoder selection, especially for hardware decoders
+    -   make input activation “atomic” when adding inputs dynamically
+    -   better interleave handling: take into account decoder latency
+        for interleave size
+
+-   decklink:
+
+    -   Updated DeckLink SDK to 11.2 to support DeckLink 8K Pro
+    -   decklinkvideosrc:
+        -   More accurate and stable capture timestamps: use the
+            hardware reference clock time when the frame was finished
+            being captured instead of a clock time much further down the
+            road.
+        -   Automatically detect widescreen vs. normal NTSC/PAL
+
+-   encodebin:
+
+    -   add “smart encoding” support for H.265, VP8 and VP9 (i.e. only
+        re-encode where needed and otherwise pass through encoded video
+        as-is).
+    -   H264/H265 smart encoding improvements: respect user-specified
+        stream-format, but if not specified default to avc3/hvc1 with
+        in-band SPS/PPS/VPS signalling for more flexibility.
+    -   new encodebin2 element with dynamic/sometimes source pads in
+        order to support the option of doing the muxing outside of
+        encodebin, e.g. in combination with splitmuxsink.
+    -   add APIs to set element properties on encoding profiles (see
+        below)
+
+-   errorignore: new "ignore-eos" property to also ignore FLOW_EOS from
+    downstream elements
+
+-   giosrc: add support for growing source files: applications can
+    specify that the underlying file being read is growing by setting
+    the "is-growing" property. If set, the source won’t EOS when it
+    reaches the end of the file, but will instead start monitoring it
+    and will start reading data again whenever a change is detected. The
+    new "waiting-data" and "done-waiting-data" signals keep the
+    application informed about the current state.
+
+-   gtksink, gtkglsink:
+
+    -   scroll event support: forwarded as navigation events into the
+        pipeline
+    -   "video-aspect-ratio-override" property to force a specific
+        aspect ratio
+    -   "rotate-method" property and support automatic rotation based on
+        image tags
+
+-   identity: new "stats" property allows applications to retrieve the
+    number of bytes and buffers that have passed through so far.
+
+-   interlace: add support for more formats, esp 10-bit, 12-bit and
+    16-bit ones
+
+-   jack: new "low-latency" property for automatic latency-optimized
+    setting and "port-names" property to select ports explicitly
+
+-   jpegdec: support output conversion to RGB using libjpeg-turbo (for
+    certain input files)
+
+-   line21dec:
+
+    -   "mode" property to control whether and how detected closed
+        captions should be inserted in the list of existing close
+        caption metas on the input frame (if any): add, drop, or
+        replace.
+    -   "ntsc-only" property to only look for captions if video has NTSC
+        resolution
+
+-   line21enc: new "remove-caption-meta" to remove metas from output
+    buffers after encoding the captions into the video data; support for
+    CDP closed captions
+
+-   matroskademux, matroskamux: Add support for ffv1, a lossless
+    intra-frame video coding format.
+
+-   matroskamux: accept in-band SPS/PPS/VPS for H264 and H265
+    (i.e. stream-format avc3 and hev1) which allows on-the-fly
+    profile/level/resolution changes.
+
+-   matroskamux: new "cluster-timestamp-offset" property, useful for use
+    cases where the container timestamps should map to some absolute
+    wall clock time, for example.
+
+-   rtpsrc: add "caps" property to allow explicit setting of the caps
+    where needed
+
+-   mpegts: support SCTE-35 passthrough via new "send-scte35-events"
+    property on MPEG-TS demuxer tsdemux. When enabled, SCTE 35 sections
+    (eg ad placement opportunities) are forwarded as events donwstream
+    where they can be picked up again by mpegtsmux. This required a
+    semantic change in the SCTE-35 section API: timestamps are now in
+    running time instead of muxer pts.
+
+-   tsdemux: Handle PCR-less MPEG-TS streams; more robust timestamp
+    handling in certain corner cases and for poorly muxed streams.
+
+-   mpegtsmux:
+
+    -   More conformance improvements to make MPEG-TS analyzers happy:
+        -   PCR timing accuracy: Improvements to the way mpegtsmux
+            outputs PCR observations in CBR mode, so that a PCR
+            observation is always inserted when needed, so that we never
+            miss the configured pcr-interval, as that triggers various
+            MPEG-TS analyser errors.
+        -   Improved PCR/SI scheduling
+    -   Don’t write PCR until PAT/PMT are output to make sure streams
+        start cleanly with a PAT/PMT.
+    -   Allow overriding the automatic PMT PID selection via
+        application-supplied PMT_%d fields in the prog-map
+        structure/property.
+
+-   mp4mux:
+
+    -   new "first-moov-then-finalise" mode for fragmented output where
+        the output will start with a self-contained moov atom for the
+        first fragment, and then produce regular fragments. Then at the
+        end when the file is finalised, the initial moov is invalidated
+        and a new moov is written covering the entire file. This way the
+        file is a “fragmented mp4” file while it is still being written
+        out, and remains playable at all times, but at the end it is
+        turned into a regular mp4 file (with former fragment headers
+        remaining as unused junk data in the file).
+    -   support H.264 avc3 and H.265 hvc1 stream formats as input where
+        the codec data is signalled in-band inside the bitstream instead
+        of caps/file headers.
+    -   support profile/level/resolution changes for H264/H265 input
+        streams (i.e. codec data changing on the fly). Each codec_data
+        is put into its own SampleTableEntry inside the stsd, unless the
+        input is in avc3 stream format in which case it’s written
+        in-band an not in the headers.
+
+-   multifilesink: new ""min-keyframe-distance"" property to make
+    minimum distance between keyframes in next-file=key-frame mode
+    configurable instead of hard-coding it to 10 seconds.
+
+-   mxfdemux has seen a big refactoring to support non-frame wrappings
+    and more accurate timestamp/seek handling for some formats
+
+-   msdk plugin for hardware-accelerated video encoding and decoding
+    using the Intel Media SDK:
+
+    -   oneVPL support (Intel oneAPI Video Processing Library)
+    -   AV1 decoding support
+    -   H.264 decoder now supports constrained-high and progressive-high
+        profiles
+    -   H.264 encoder:
+        -   more configuration options (properties):
+            "intra-refresh-type", "min-qp" , "max-qp", "p-pyramid",
+            "dblk-idc"
+    -   H.265 encoder:
+        -   can output main-still-picture profile
+        -   now inserts HDR SEIs (mastering display colour volume and
+            content light level)
+        -   more configuration options (properties):
+            "intra-refresh-type", "min-qp" , "max-qp", "p-pyramid",
+            "b-pyramid", "dblk-idc", "transform-skip"
+        -   support for RGB 10bit format
+    -   External bitrate control in encoders
+    -   Video post proc element msdkvpp gained support for 12-bit pixel
+        formats P012_LE, Y212_LE and Y412_LE
+
+-   nvh264sldec: interlaced stream support
+
+-   openh264enc: support main, high, constrained-high and
+    progressive-high profiles
+
+-   openjpeg: support for multithreaded decoding and encoding
+
+-   rtspsrc: now supports IPv6 also for tunneled mode (RTSP-over-HTTP);
+    new "ignore-x-server-reply" property to ignore the
+    x-server-ip-address server header reply in case of HTTP tunneling,
+    as it is often broken.
+
+-   souphttpsrc: Runtime compatibility support for libsoup2 and
+    libsoup3. libsoup3 is the latest major version of libsoup, but
+    libsoup2 and libsoup3 can’t co-exist in the same process because
+    there is no namespacing or versioning for GObject types. As a
+    result, it would be awkward if the GStreamer souphttpsrc plugin
+    linked to a specific version of libsoup, because it would only work
+    with applications that use the same version of libsoup. To make this
+    work, the soup plugin now tries to determine the libsoup version
+    used by the application (and its other dependencies) at runtime on
+    systems where GStreamer is linked dynamically. libsoup3 support is
+    still considered somewhat experimental at this point.
+
+-   srtsrc, srtsink: add signals for the application to accept/reject
+    incoming connections
+
+-   timeoverlay: new elapsed-running-time time mode which shows the
+    running time since the first running time (and each flush-stop).
+
+-   udpsrc: new timestamping mode to retrieve packet receive timestamps
+    from the kernel via socket control messages (SO_TIMESTAMPNS) on
+    supported platforms
+
+-   uritranscodebin: new setup-source and element-setup signals for
+    applications to configure elements used
+
+-   v4l2codecs plugin gained support for 4x4 and 32x32 tile formats
+    enabling some platforms or direct renders. Important memory usage
+    improvement.
+
+-   v4l2slh264dec now implements the final Linux uAPI as shipped on
+    Linux 5.11 and later.
+
+-   valve: add "drop-mode" property and provide two new modes of
+    operation: in drop-mode=forward-sticky-events sticky events
+    (stream-start, segment, tags, caps, etc.) are forwarded downstream
+    even when dropping is enabled; drop-mode=transform-to-gap will in
+    addition also convert buffers into gap events when dropping is
+    enabled, which lets downstream elements know that time is advancing
+    and might allow for preroll in many scenarios. By default all events
+    and all buffers are dropped when dropping is enabled, which can
+    cause problems with caps negotiation not progressing or branches not
+    prerolling when dropping is enabled.
+
+-   videocrop: support for many more pixel formats, e.g. planar YUV
+    formats with > 8bits and GBR* video formats; can now also accept
+    video not backed by system memory as long as downstream supports the
+    GstCropMeta
+
+-   videotestsrc: new smpte-rp-219 pattern for SMPTE75 RP-219 conformant
+    color bars
+
+-   vp8enc: finish support for temporal scalability: two new properties
+    ("temporal-scalability-layer-flags",
+    "temporal-scalability-layer-sync-flags") and a unit change on the
+    "temporal-scalability-target-bitrate" property (now expects bps);
+    also make temporal scalability details available to RTP payloaders
+    as buffer metadata.
+
+-   vp9enc: new properties to tweak encoder performance:
+
+    -   "aq-mode" to configure adaptive quantization modes
+    -   "frame-parallel-decoding" to configure whether to create a
+        bitstream that reduces decoding dependencies between frames
+        which allows staged parallel processing of more than one video
+        frames in the decoder. (Defaults to TRUE)
+    -   "row-mt", "tile-columns" and "tile-rows" so multithreading can
+        be enabled on a per-tile basis, instead of on a per tile-column
+        basis. In combination with the new "tile-rows" property, this
+        allows the encoder to make much better use of the available CPU
+        power.
+
+-   vp9dec, vp9enc: add support for 10-bit 4:2:0 and 4:2:2 YUV, as well
+    as 8-bit 4:4:4
+
+-   vp8enc, vp9enc now default to “good quality” for the deadline
+    property rather then “best quality”. Having the deadline set to best
+    quality causes the encoder to be absurdly slow, most real-life users
+    will prefer good-enough quality with better performance instead.
+
+-   wpesrc:
+
+    -   implement audio support: a new sometimes source pad will be
+        created for each audio stream created by the web engine.
+    -   move wpesrc to wpevideosrc and add a wrapper bin wpesrc to also
+        support audio
+    -   also handles web:// URIs now (same as cefsrc)
+    -   post messages with the estimated load progress on the bus
 
-Plugin and library moves
+-   x265enc: add negative DTS support, which means timestamps are now
+    offset by 1h same as with x264enc
+
+RTP Payloaders and Depayloaders
+
+-   rtpisacpay, rtpisacdepay: new RTP payloader and depayloader for iSAC
+    audio codec
+
+-   rtph264depay:
+
+    -   new "request-keyframe" property to make the depayloader
+        automatically request a new keyframe from the sender on packet
+        loss, consistent with the new property on rtpvp8depay.
+    -   new "wait-for-keyframe" property to make depayloader wait for a
+        new keyframe at the beginning and after packet loss (only
+        effective if the depayloader outputs AUs), consistent with the
+        existing property on rtpvp8depay.
+
+-   rtpopuspay, rtpopusdepay: support libwebrtc-compatible multichannel
+    audio in addition to the previously supported multichannel audio
+    modes
+
+-   rtpopuspay: add DTX (Discontinuous Transmission) support
+
+-   rtpvp8depay: new "request-keyframe" property to make the depayloader
+    automatically request a new keyframe from the sender on packet loss.
+
+-   rtpvp8pay: temporal scaling support
 
--   this section will be filled in in due course
+-   rtpvp9depay: Improved SVC handling (aggregate all layers)
+
+RTP Infrastructure
+
+-   rtpst2022-1-fecdec, rtpst2022-1-fecenc: new elements providing SMPTE
+    2022-1 2-D Forward Error Correction. More details in Mathieu’s blog
+    post.
+
+-   rtpreddec: BUNDLE support
+
+-   rtpredenc, rtpulpfecenc: add support for Transport-wide Congestion
+    Control (TWCC)
+
+-   rtpsession: new "twcc-feedback-interval" property to allow RTCP TWCC
+    reports to be scheduled on a timer instead of per marker-bit.
+
+Plugin and library moves
 
 -   There were no plugin moves or library moves in this cycle.
 
@@ -56,8 +998,6 @@ Plugin removals
 
 The following elements or plugins have been removed:
 
--   this section will be filled in in due course
-
 -   The ofa audio fingerprinting plugin has been removed. The MusicIP
     database has been defunct for years so this plugin is likely neither
     useful nor used by anyone.
@@ -71,135 +1011,575 @@ The following elements or plugins have been removed:
 
 Miscellaneous API additions
 
--   this section will be filled in in due course
+Core
+
+-   gst_buffer_new_memdup() is a convenience function for the
+    widely-used gst_buffer_new_wrapped(g_memdup(data,size),size)
+    pattern.
+
+-   gst_caps_features_new_single() creates a new single GstCapsFeatures,
+    avoiding the need to use the vararg function with NULL terminator
+    for simple cases.
+
+-   gst_element_type_set_skip_documentation() can be used by plugins to
+    signal that certain elements should not be included in the GStreamer
+    plugin documentation. This is useful for plugins where elements are
+    registered dynamically based on hardware capabilities and/or where
+    the available plugins and properties vary from system to system.
+    This is used in the d3d11 plugin for example to ensure that only the
+    list of default elements is advertised in the documentation.
+
+-   gst_type_find_suggest_empty_simple() is a new convenience function
+    for typefinders for cases where there’s only a media type and no
+    other fields.
+
+-   New API to create elements and set properties at construction time,
+    which is not only convenient, but also allows GStreamer elements to
+    have construct-only properties: gst_element_factory_make_full(),
+    gst_element_factory_make_valist(),
+    gst_element_factory_make_with_properties(),
+    gst_element_factory_create_full(),
+    gst_element_factory_create_valist(),
+    gst_element_factory_create_with_properties().
+
+-   GstSharedTaskPool: new “shared” task pool subclass with slightly
+    different default behaviour than the existing GstTaskPool which
+    would create unlimited number of threads for new tasks. The shared
+    taskpool creates up to N threads (default: 1) and then distributes
+    pending tasks to those threads round-robin style, and blocks if no
+    thread is available. It is possible to join tasks. This can be used
+    by plugins to implement simple multi-threaded processing and is used
+    for the new multi-threaded video conversion and compositing done in
+    GstVideoAggregator, videoconverter and compositor.
+
+Plugins Base Utils library
+
+-   GstDiscoverer:
+
+    -   gst_discoverer_container_info_get_tags() was added to retrieve
+        global/container tags (vs. per-stream tags). Per-Stream tags can
+        be retrieved via the existing
+        gst_discoverer_stream_info_get_tags().
+        gst_discoverer_info_get_tags(), which for many files returns a
+        confusing mix of stream and container tags, has been deprecated
+        in favour of the container/stream-specific functions.
+    -   gst_discoverer_stream_info_get_stream_number() returns a unique
+        integer identifier for a given stream within the given
+        GstDiscoverer context. (If this matches the stream number inside
+        the container bitstream that’s by coincidence and not by
+        design.)
+
+-   gst_pb_utils_get_caps_description_flags() can be used to query
+    whether certain caps represent a container, audio, video, image,
+    subtitles, tags, or something else. This only works for formats
+    known to GStreamer.
+
+-   gst_pb_utils_get_file_extension_from_caps() returns a possible file
+    extension for given caps.
+
+-   gst_codec_utils_h264_get_profile_flags_level(): Parses profile,
+    flags, and level from H264 AvcC codec_data. The format of H264 AVCC
+    extradata/sequence_header is documented in the ITU-T H.264
+    specification section 7.3.2.1.1 as well as in ISO/IEC 14496-15
+    section 5.3.3.1.2.
+
+-   gst_codec_utils_caps_get_mime_codec() to convert caps to a RFC 6381
+    compatible MIME codec string codec. Useful for providing the codecs
+    field inside the Content-Type HTTP header for containerized formats,
+    such as mp4 or matroska.
+
+GStreamer OpenGL integration library and plugins
+
+-   glcolorconvert: added suppport for converting the video formats
+    A420, AV12, BGR, BGRA, RGBP and BGRP.
+
+-   Added support to GstGLBuffer for persistent buffer mappings where a
+    Pixel Buffer Object (PBO) can be mapped by both the CPU and the GPU.
+    This removes a memcpy() when uploading textures or vertices
+    particularly when software decoders (e.g. libav) are direct
+    rendering into our memory. Improves transfer performance
+    significantly. Requires OpenGL 4.4, GL_ARB_buffer_storage or
+    GL_EXT_buffer_storage
+
+-   Added various helper functions for handling 4x4 matrices of affine
+    transformations as used by GstVideoAffineTransformationMeta.
+
+-   Add support to GstGLContext for allowing the application to control
+    the config (EGLConfig, GLXConfig, etc) used when creating the OpenGL
+    context. This allows the ability to choose between RGB16 or RGB10A2
+    or RGBA8 back/front buffer configurations that were previously
+    hardcoded. GstGLContext also supports retrieving the configuration
+    it was created with or from an externally provide OpenGL context
+    handle. This infrastructure is also used to create a compatible
+    config from an application/externally provided OpenGL context in
+    order to improve compatibility with other OpenGL frameworks and GUI
+    toolkits. A new environment variable GST_GL_CONFIG was also added to
+    be able to request a specific configuration from the command line.
+    Note: different platforms will have different functionality
+    available.
+
+-   Add support for choosing between EGL and WGL at runtime when running
+    on Windows. Previously this was a build-time switch. Allows use in
+    e.g. Gtk applications on Windows that target EGL/ANGLE without
+    recompiling GStreamer. gst_gl_display_new_with_type() can be used by
+    applications to choose a specific display type to use.
+
+-   Build fixes to explicitly check for Broadcom-specific libraries on
+    older versions of the Raspberry Pi platform. The Broadcom OpenGL ES
+    and EGL libraries have different filenames. Using the vc4 Mesa
+    driver on the Raspberry Pi is not affected.
+
+-   Added support to glupload and gldownload for transferring RGBA
+    buffers using the memory:NVMM available on the Nvidia Tegra family
+    of embedded devices.
+
+-   Added support for choosing libOpenGL and libGLX as used in a GLVND
+    environment on unix-based platforms. This allows using desktop
+    OpenGL and EGL without pulling in any GLX symbols as would be
+    required with libGL.
+
+Video library
+
+-   New raw video formats:
+
+    -   AV12 (NV12 with alpha plane)
+    -   RGBP and BGRP (planar RGB formats)
+    -   ARGB64 variants with specified endianness instead of host
+        endianness:
+        -   ARGB64_LE, ARGB64_BE
+        -   RGBA64_BE, RGBA64_LE
+        -   BGRA64_BE, BGRA64_LE
+        -   ABGR64_BE, ABGR64_LE
+
+-   gst_video_orientation_from_tag() is new convenience API to parse the
+    image orientation from a GstTagList.
+
+-   GstVideoDecoder subframe support (see below)
+
+-   GstVideoCodecState now also carries some HDR metadata
+
+-   Ancillary video data: implement transform functions for AFD/Bar
+    metas, so they will be forwarded in more cases
+
+MPEG-TS library
+
+This library only handles section parsing and such, see above for
+changes to the actual mpegtsmux and mpegtsdemux elements.
+
+-   many additions and improvements to SCTE-35 section parsing
+-   new API for fetching extended descriptors:
+    gst_mpegts_find_descriptor_with_extension()
+-   add support for SIT sections (Selection Information Tables)
+-   expose event-from-section constructor gst_event_new_mpegts_section()
+-   parse Audio Preselection Descriptor needed for Dolby AC-4
+
+GstWebRTC library + webrtcbin
+
+-   Change the way in which sink pads and transceivers are matched
+    together to support easier usage. If a pad is created without a
+    specific index (i.e. using sink_%u as the pad template), then an
+    available compatible transceiver will be searched for. If a specific
+    index is requested (i.e. sink_1) then if a transceiver for that
+    m-line already exists, that transceiver must match the new sink pad
+    request. If there is no transceiver available in either scenario, a
+    new transceiver is created. If a mixture of both sink_1 and sink_%u
+    requests result in an impossible situation, an error will be
+    produced at pad request time or from create offer/answer.
+
+-   webrtcbin now uses regular ICE nomination instead of libnice’s
+    default of aggressive ICE nomination. Regular ICE nomination is the
+    default recommended by various relevant standards and improves
+    connectivity in specific network scenarios.
+
+-   Add support for limiting the port range used for RTP with the
+    addition of the min-rtp-port and max-rtp-port properties on the ICE
+    object.
+
+-   Expose the SCTP transport as a property on webrtcbin to more closely
+    match the WebRTC specification.
+
+-   Added support for taking into account the data channel transport
+    state when determining the value of the "connection-state" property.
+    Previous versions of the WebRTC spec did not include the data
+    channel state when computing this value.
+
+-   Add configuration for choosing the size of the underlying sockets
+    used for transporting media data
+
+-   Always advertise support for the transport-cc RTCP feedback protocol
+    as rtpbin supports it. For full support, the configured caps (input
+    or through codec-preferences) need to include the relevant RTP
+    header extension.
+
+-   Numerous fixes to caps and media handling to fail-fast when an
+    incompatible situation is detected.
+
+-   Improved support for attaching the required media after a remote
+    offer has been set.
+
+-   Add support for dynamically changing the amount of FEC used for a
+    particular stream.
+
+-   webrtcbin now stops further SDP processing at the first error it
+    encounters.
+
+-   Completed support for either local or the remote closing a data
+    channel.
+
+-   Various fixes when performing BUNDLEing of the media streams in
+    relation to RTX and FEC usage.
+
+-   Add support for writing out QoS DSCP marking on outgoing packets to
+    improve reliability in some network scenarios.
+
+-   Improvements to the statistics returned by the get-stats signal
+    including the addition of the raw statistics from the internal
+    RTPSource, the TWCC stats when available.
+
+-   The webrtc library does not expose any objects anymore with public
+    fields. Instead properties have been added to replace that
+    functionality. If you are accessing such fields in your application,
+    switch to the corresponding properties.
+
+GstCodecs and Video Parsers
+
+-   Support for render delays to improve throughput across all CODECs
+    (used with NVDEC and V4L2).
+-   lots of improvements to parsers and the codec parsing decoder base
+    classes (H264, H265, VP8, VP9, AV1, MPEG-2) used for various
+    hardware-accelerate decoder APIs.
+
+Bindings support
+
+-   gst_allocation_params_new() allocates a GstAllocationParams struct
+    on the heap. This should only be used by bindings (and freed via
+    gst_allocation_params_free() then). In C code you would allocate
+    this on the stack and only init it in place.
+
+-   gst_debug_log_literal() can be used to log a string to the debug log
+    without going through any printf format expansion and associated
+    overhead. This is mostly useful for bindings such as the Rust
+    bindings which may have done their own formatting already .
+
+-   Provide non-inlined versions of refcounting APIs for various
+    GStreamer mini objects, so that they can be consumed by bindings
+    (e.g. gstreamer-sharp): gst_buffer_ref, gst_buffer_unref,
+    gst_clear_buffer, gst_buffer_copy, gst_buffer_replace,
+    gst_buffer_list_ref, gst_buffer_list_unref, gst_clear_buffer_list,
+    gst_buffer_list_copy, gst_buffer_list_replace, gst_buffer_list_take,
+    gst_caps_ref, gst_caps_unref, gst_clear_caps, gst_caps_replace,
+    gst_caps_take, gst_context_ref, gst_context_unref, gst_context_copy,
+    gst_context_replace, gst_event_replace, gst_event_steal,
+    gst_event_take, gst_event_ref, gst_event_unref, gst_clear_event,
+    gst_event_copy, gst_memory_ref, gst_memory_unref, gst_message_ref,
+    gst_message_unref, gst_clear_message, gst_message_copy,
+    gst_message_replace, gst_message_take, gst_promise_ref,
+    gst_promise_unref, gst_query_ref, gst_query_unref, gst_clear_query,
+    gst_query_copy, gst_query_replace, gst_query_take, gst_sample_ref,
+    gst_sample_unref, gst_sample_copy, gst_tag_list_ref,
+    gst_tag_list_unref, gst_clear_tag_list, gst_tag_list_replace,
+    gst_tag_list_take, gst_uri_copy, gst_uri_ref, gst_uri_unref,
+    gst_clear_uri.
+
+-   expose a GType for GstMiniObject
+
+-   gst_device_provider_probe() now returns non-floating device object
+
+API Deprecations
+
+-   gst_element_get_request_pad() has been deprecated in favour of the
+    newly-added gst_element_request_pad_simple() which does the exact
+    same thing but has a less confusing name that hopefully makes clear
+    that the function request a new pad rather than just retrieves an
+    already-existing request pad.
+
+-   gst_discoverer_info_get_tags(), which for many files returns a
+    confusing mix of stream and container tags, has been deprecated in
+    favour of the container-specific and stream-specific functions,
+    gst_discoverer_container_info_get_tags() and
+    gst_discoverer_stream_info_get_tags().
+
+-   gst_video_sink_center_rect() was deprecated in favour of the more
+    generic newly-added gst_video_center_rect().
+
+-   The GST_MEMORY_FLAG_NO_SHARE flag has been deprecated, as it tends
+    to cause problems and prevents sub-buffering. If pooling or lifetime
+    tracking is required, memories should be allocated through a custom
+    GstAllocator instead of relying on the lifetime of the buffers the
+    memories were originally attached to, which is fragile anyway.
+
+-   The GstPlayer high-level playback library is being replaced with the
+    new GstPlay library (see above). GstPlayer should be considered
+    deprecated at this point and will be marked as such in the next
+    development cycle. Applications should be ported to GstPlay.
+
+-   Gstreamer Editing Services: ges_video_transition_set_border(),
+    ges_video_transition_get_border()
+    ges_video_transition_set_inverted()
+    ges_video_transition_is_inverted() have been deprecated, use
+    ges_timeline_element_set_children_properties() instead.
 
 Miscellaneous performance, latency and memory optimisations
 
--   this section will be filled in in due course
+More video conversion fast paths
 
-Miscellaneous other changes and enhancements
+-   v210 ↔ I420, YV12, Y42B, UYVY and YUY2
+-   A420 → RGB
+
+Less jitter when waiting on the system clock
+
+-   Better system clock wait accuracy, less jitter: where available,
+    clock_nanosleep is used for higher accuracy for waits below 500
+    usecs, and waits below 2ms will first use the regular waiting system
+    and then clock_nanosleep for the remainder. The various wait
+    implementation have a latency ranging from 50 to 500+ microseconds.
+    While this is not a major issue when dealing with a low number of
+    waits per second (for ex: video), it does introduce a non-negligible
+    jitter for synchronisation of higher packet rate systems.
+
+Video decoder subframe support
 
--   this section will be filled in in due course
+-   The GstVideoDecoder base class gained API to process input at the
+    sub-frame level. That way video decoders can start decoding slices
+    before they have received the full input frame in its entirety (to
+    the extent this is supported by the codec, of course). This helps
+    with CPU utilisation and reduces latency.
 
-Tracing framework and debugging improvements
+-   This functionality is now being used in the OpenJPEG JPEG 2000
+    decoder, the FFmpeg H.264 decoder (in case of NAL-aligned input) and
+    the OpenMAX H.264/H.265 decoders (in case of NAL-aligned input).
 
--   this section will be filled in in due course
+Miscellaneous other changes and enhancements
+
+-   GstDeviceMonitor no longer fails to start just because one of the
+    device providers failed to start. That could happen for example on
+    systems where the pulseaudio device provider is installed, but
+    pulseaudio isn’t actually running but ALSA is used for audio
+    instead. In the same vein the device monitor now keeps track of
+    which providers have been started (via the new
+    gst_device_provider_is_started()) and only stops actually running
+    device providers when stopping the device monitor.
+
+-   On embedded systems it can be useful to create a registry that can
+    be shared and read by multiple processes running as different users.
+    It is now possible to set the new GST_REGISTRY_MODE environment
+    variable to specify the file mode for the registry file, which by
+    default is set to be only user readable/writable.
+
+-   GstNetClientClock will signal lost sync in case the remote time
+    resets (e.g. because device power cycles), by emitting the “synced”
+    signal with synced=FALSE parameter, so applications can take action.
+
+-   gst_value_deserialize_with_pspec() allows deserialization with a
+    hint for what the target GType should be. This allows for example
+    passing arrays of flags through the command line or
+    gst_util_set_object_arg(), eg: foo="<bar,bar+baz>".
+
+-   It’s now allowed to create an empty GstVideoOverlayComposition
+    without any rectangles by passing a NULL rectangle to
+    gst_video_overlay_composition_new(). This is useful for bindings and
+    simplifies application code in some places.
+
+Tracing framework, debugging and testing improvements
+
+-   New factories tracer to list loaded elements (and other plugin
+    features). This can be useful to collect a list of elements needed
+    for an application, which then in turn can be used to create a
+    tailored minimal GStreamer build that contains just the elements
+    needed and nothing else.
+-   New plugin-feature-loaded tracing hook for use by tracers like the
+    new factories tracer
+
+-   GstHarness: Add gst_harness_set_live() so that harnesses can be set
+    to non-live and return is-live=false in latency queries if needed.
+    Default behaviour is to always return is-live=true in latency
+    queries.
+
+-   navseek: new "hold-eos" property. When enabled, the element will
+    hold back an EOS event until the next keystroke (via navigation
+    events). This can be used to keep a video sink showing the last
+    frame of a video pipeline until a key is pressed instead of tearing
+    it down immediately on EOS.
+
+-   New fakeaudiosink element: mimics an audio sink and can be used for
+    testing and CI pipelines on systems where no audio system is
+    installed or running. It differs from fakesink in that it only
+    support audio caps and syncs to the clock by default like a normal
+    audio sink. It also implements the GstStreamVolume interface like
+    most audio sinks do.
+
+-   New videocodectestsink element for video codec conformance testing:
+    Calculates MD5 checksums for video frames and skips any padding
+    whilst doing so. Can optionally also write back the video data with
+    padding removed into a file for easy byte-by-byte comparison with
+    reference data.
 
 Tools
 
--   this section will be filled in in due course
+gst-inspect-1.0
+
+-   Can sort the list of plugins by passing --sort=name as command line
+    option
+
+gst-launch-1.0
+
+-   will now error out on top-level properties that don’t exist and
+    which were silently ignored before
+-   On Windows the high-resolution clock is enabled now, which provides
+    better clock and timer performance on Windows (see Windows section
+    below for more details).
+
+gst-play-1.0
+
+-   New --start-position command line argument to start playback from
+    the specified position
+-   Audio can be muted/unmuted in interactive mode by pressing the m
+    key.
+-   On Windows the high-resolution clock is enabled now (see Windows
+    section below for more details)
+
+gst-device-monitor-1.0
+
+-   New --include-hidden command line argument to also show “hidden”
+    device providers
+
+ges-launch-1.0
+
+-   New interactive mode that allows seeking and such. Can be disabled
+    by passing the --no-interactive argument on the command line.
+-   Option to forward tags
+-   Allow using an existing clip to determine the rendering format (both
+    topology and profile) via new --profile-from command line argument.
 
 GStreamer RTSP server
 
--   this section will be filled in in due course
+-   GstRTSPMediaFactory gained API to disable RTCP
+    (gst_rtsp_media_factory_set_enable_rtcp(), "enable-rtcp" property).
+    Previously RTCP was always allowed for all RTSP medias. With this
+    change it is possible to disable RTCP completely, no matter if the
+    client wants to do RTCP or not.
+
+-   Make a mount point of / work correctly. While not allowed by the
+    RTSP 2 spec, the RTSP 1 spec is silent on this and it is used in the
+    wild. It is now possible to use / as a mount path in
+    gst-rtsp-server, e.g. rtsp://example.com/ would work with this now.
+    Note that query/fragment parts of the URI are not necessarily
+    correctly handled, and behaviour will differ between various
+    client/server implementations; so use it if you must but don’t bug
+    us if it doesn’t work with third party clients as you’d hoped.
+
+-   multithreading fixes (races, refcounting issues, deadlocks)
+
+-   ONVIF audio backchannel fixes
+
+-   ONVIF trick mode optimisations
+
+-   rtspclientsink: new "update-sdp" signal that allows updating the SDP
+    before sending it to the server via ANNOUNCE. This can be used to
+    add additional metadata to the SDP, for example. The order and
+    number of medias must not be changed, however.
 
 GStreamer VAAPI
 
--   this section will be filled in in due course
+-   new AV1 decoder element (vaapiav1dec)
+
+-   H264 decoder: handle stereoscopic 3D video with frame packing
+    arrangement SEI messages
+
+-   H265 encoder: added Screen Content Coding extensions support
+
+-   H265 decoder: gained MAIN_444_12 profile support (decoded to
+    Y412_LE), and 4:2:2 12-bits support (decoded to Y212_LE)
+
+-   vaapipostproc: gained BT2020 color standard support
+
+-   vaapidecode: now generates caps templates dynamically at runtime in
+    order to advertise actually supported caps instead of all
+    theoretically supported caps.
+
+-   GST_VAAPI_DRM_DEVICE environment variable to force a specified DRM
+    device when a DRM display is used. It is ignored when other types of
+    displays are used. By default /dev/dri/renderD128 is used for DRM
+    display.
 
 GStreamer OMX
 
--   this section will be filled in in due course
+-   subframe support in H.264/H.265 decoders
 
 GStreamer Editing Services and NLE
 
--   this section will be filled in in due course
+-   framepositioner: new "operator" property to access blending modes in
+    the compositor
+-   timeline: Implement snapping to markers
+-   smart-mixer: Add support for d3d11compositor and glvideomixer
+-   titleclip: add "draw-shadow" child property
+-   ges:// URI support to define a timeline from a description.
+-   command-line-formatter
+    -   Add track management to timeline description
+    -   Add keyframe support
+-   ges-launch-1.0:
+    -   Add an interactive mode where we can seek etc…
+    -   Add option to forward tags
+    -   Allow using an existing clip to determine the rendering format
+        (both topology and profile) via new --profile-from command line
+        argument.
+-   Fix static build
 
 GStreamer validate
 
--   this section will be filled in in due course
+-   report: Add a way to force backtraces on reports even if not a
+    critical issue (GST_VALIDATE_ISSUE_FLAGS_FORCE_BACKTRACE)
+-   Add a flag to gst_validate_replace_variables_in_string() allow
+    defining how to resolve variables in structs
+-   Add gst_validate_bin_monitor_get_scenario() to get the bin monitor
+    scenario, which is useful for applications that use Validate
+    directly.
+-   Add an expected-values parameter to wait, message-type=XX allowing
+    more precise filtering of the message we are waiting for.
+-   Add config file support: each test can now use a config file for the
+    given media file used to test.
+-   Add support to check properties of object properties
+-   scenario: Add an "action-done" signal to signal when an action is
+    done
+-   scenario: Add a "run-command" action type
+-   scenario: Allow forcing running action on idle from scenario file
+-   scenario: Allow iterating over arrays in foreach
+-   scenario: Rename ‘interlaced’ action to ‘non-blocking’
+-   scenario: Add a non-blocking flag to the wait signal
 
 GStreamer Python Bindings
 
--   this section will be filled in in due course
+-   Fixes for Python 3.10
+-   Various build fixes
+-   at least one known breaking change caused by g-i annotation changes
+    (see below)
 
 GStreamer C# Bindings
 
--   this section will be filled in in due course
+-   Fix GstDebugGraphDetails enum
+-   Updated to latests GtkSharp
+-   Updated to include GStreamer 1.20 API
 
 GStreamer Rust Bindings and Rust Plugins
 
-The GStreamer Rust bindings are released separately with a different
-release cadence that’s tied to gtk-rs, but the latest release has
-already been updated for the upcoming new GStreamer 1.20 API.
-
-gst-plugins-rs, the module containing GStreamer plugins written in Rust,
-has also seen lots of activity with many new elements and plugins.
-
-What follows is a list of elements and plugins available in
-gst-plugins-rs, so people don’t miss out on all those potentially useful
-elements that have no C equivalent.
-
--   FIXME: add new elements
-
-Rust audio plugins
-
--   audiornnoise: New element for audio denoising which implements the
-    noise removal algorithm of the Xiph RNNoise library, in Rust
--   rsaudioecho: Port of the audioecho element from gst-plugins-good
-    rsaudioloudnorm: Live audio loudness normalization element based on
-    the FFmpeg af_loudnorm filter
--   claxondec: FLAC lossless audio codec decoder element based on the
-    pure-Rust claxon implementation
--   csoundfilter: Audio filter that can use any filter defined via the
-    Csound audio programming language
--   lewtondec: Vorbis audio decoder element based on the pure-Rust
-    lewton implementation
-
-Rust video plugins
-
--   cdgdec/cdgparse: Decoder and parser for the CD+G video codec based
-    on a pure-Rust CD+G implementation, used for example by karaoke CDs
--   cea608overlay: CEA-608 Closed Captions overlay element
--   cea608tott: CEA-608 Closed Captions to timed-text (e.g. VTT or SRT
-    subtitles) converter
--   tttocea608: CEA-608 Closed Captions from timed-text converter
--   mccenc/mccparse: MacCaption Closed Caption format encoder and parser
--   sccenc/sccparse: Scenarist Closed Caption format encoder and parser
--   dav1dec: AV1 video decoder based on the dav1d decoder implementation
-    by the VLC project
--   rav1enc: AV1 video encoder based on the fast and pure-Rust rav1e
-    encoder implementation
--   rsflvdemux: Alternative to the flvdemux FLV demuxer element from
-    gst-plugins-good, not feature-equivalent yet
--   rsgifenc/rspngenc: GIF/PNG encoder elements based on the pure-Rust
-    implementations by the image-rs project
-
-Rust text plugins
-
--   textwrap: Element for line-wrapping timed text (e.g. subtitles) for
-    better screen-fitting, including hyphenation support for some
-    languages
-
-Rust network plugins
-
--   reqwesthttpsrc: HTTP(S) source element based on the Rust
-    reqwest/hyper HTTP implementations and almost feature-equivalent
-    with the main GStreamer HTTP source souphttpsrc
--   s3src/s3sink: Source/sink element for the Amazon S3 cloud storage
--   awstranscriber: Live audio to timed text transcription element using
-    the Amazon AWS Transcribe API
-
-Generic Rust plugins
-
--   sodiumencrypter/sodiumdecrypter: Encryption/decryption element based
-    on libsodium/NaCl
--   togglerecord: Recording element that allows to pause/resume
-    recordings easily and considers keyframe boundaries
--   fallbackswitch/fallbacksrc: Elements for handling potentially
-    failing (network) sources, restarting them on errors/timeout and
-    showing a fallback stream instead
--   threadshare: Set of elements that provide alternatives for various
-    existing GStreamer elements but allow to share the streaming threads
-    between each other to reduce the number of threads
--   rsfilesrc/rsfilesink: File source/sink elements as replacements for
-    the existing filesrc/filesink elements
+-   The GStreamer Rust bindings are released separately with a different
+    release cadence that’s tied to gtk-rs, but the latest release has
+    already been updated for the upcoming new GStreamer 1.20 API (v1_20
+    feature).
+
+-   gst-plugins-rs, the module containing GStreamer plugins written in
+    Rust, has also seen lots of activity with many new elements and
+    plugins. See the New Elements section above for a list of new Rust
+    elements.
 
 Build and Dependencies
 
 -   Meson 0.59 or newer is required to build GStreamer now.
 
--   FIXME: this section will be filled in in due course
+-   The GLib requirement has been bumped to GLib 2.56 or newer (from
+    March 2018).
+
+-   The wpe plugin now requires wpe >= 2.28 and wpebackend-fdo >= 1.8
 
 Explicit opt-in required for build of certain plugins with (A)GPL dependencies
 
@@ -211,11 +1591,9 @@ even if the required dependencies are available.
 See Building plugins with (A)GPL-licensed dependencies for more details
 and a non-exhaustive list of plugins affected.
 
-gst-build: replaced by Monorepo
+gst-build: replaced by mono repository
 
--   this section will be filled in in due course
-
--   FIXME: describe + link to Monorepo FAQ
+See mono repository section above and the GStreamer mono repository FAQ.
 
 Cerbero
 
@@ -223,76 +1601,254 @@ Cerbero is a meta build system used to build GStreamer plus dependencies
 on platforms where dependencies are not readily available, such as
 Windows, Android, iOS and macOS.
 
-General improvements
+General Cerbero improvements
 
--   this section will be filled in in due course
+-   Plugin removed: libvisual
+-   New plugins: rtpmanagerbad and rist
 
-macOS / iOS
+macOS / iOS specific Cerbero improvements
 
--   this section will be filled in in due course
+-   XCode 12 support
+-   macOS OS release support is now future-proof, similar to iOS
+-   macOS Apple Silicon (ARM64) cross-compile support has been added
+-   macOS Apple Silicon (ARM64) native support is currently experimental
 
-Windows
+Windows specific Cerbero improvements
 
--   this section will be filled in in due course
+-   Visual Studio 2022 support has been added
+-   bootstrap is faster since it requires building fewer build-tools
+    recipes on Windows
+-   package is faster due to better scheduling of recipe stages and
+    elimination of unnecessary autotools regeneration
+-   The following plugins are no longer built on Windows:
+    -   a52dec (another decoder is still available in libav)
+    -   dvdread
+    -   resindvd
 
 Windows MSI installer
 
--   this section will be filled in in due course
+-   no major changes
 
-Linux
+Linux specific Cerbero improvements
 
--   this section will be filled in in due course
+-   Fedora, Debian OS release support is now more future-proof
+-   Amazon Linux 2 support has been added
 
-Android
+Android specific Cerbero improvements
 
--   this section will be filled in in due course
+-   no major changes
 
 Platform-specific changes and improvements
 
 Android
 
--   this section will be filled in in due course
+-   No major changes
 
 macOS and iOS
 
--   this section will be filled in in due course
+-   applemedia: add ProRes support to vtenc and vtdec
 
 Windows
 
--   this section will be filled in in due course
+-   On Windows the high-resolution clock is enabled now in the
+    gst-launch-1.0 and gst-play-1.0 command line tools, which provides
+    better clock and timer performance on Windows, at the cost of higher
+    power consumption. By default, without the high-resolution clock
+    enabled, the timer precision on Windows is system-dependent and may
+    be as bad as 15ms which is not good enough for many multimedia
+    applications. Developers may want to do the same in their Windows
+    applications if they think it’s a good idea for their application
+    use case, and depending on the Windows version they target. This is
+    not done automatically by GStreamer because on older Windows
+    versions (pre-Windows 10) this affects a global Windows setting and
+    also there’s a power consumption vs. performance trade-off that may
+    differ from application to application.
+
+-   dxgiscreencapsrc now supports resolution changes
+
+-   The wasapi2 audio plugin was rewritten and now has a higher rank
+    than the old wasapi plugin since it has a number of additional
+    features such as automatic stream routing, and no
+    known-but-hard-to-fix issues. The plugin is always built if the
+    Windows 10 SDK is available now.
+
+-   The wasapi device providers now detect and notify dynamic device
+    additions/removals
+
+-   d3d11screencapturesrc: new desktop capture element, including
+    GstDeviceProvider implementation to enumerate/select target monitors
+    for capture.
+
+-   Direct3D11/DXVA decoder now supports AV1 and MPEG2 codecs
+    (d3d11av1dec, d3d11mpeg2dec)
+
+-   VP9 decoding got more reliable and stable thanks to a newly written
+    codec parser
+
+-   Support for decoding interlaced H.264/AVC streams
+
+-   Hardware-accelerated video deinterlacing (d3d11deinterlace) and
+    video mixing (d3d11compositor)
+
+-   Video mixing with the Direct3D11 API (d3d11compositor)
+
+-   MediaFoundation API based hardware encoders gained the ability to
+    receive Direct3D11 textures as an input
+
+-   Seungha’s blog post “GStreamer ❤ Windows: A primer on the cool stuff
+    you’ll find in the 1.20 release” describes many of the
+    Windows-related improvements in more detail
 
 Linux
 
--   this section will be filled in in due course
+-   bluez: LDAC Bluetooth audio codec support in a2dpsink and avdtpsink,
+    as well as an LDAC RTP payloader (rtpldacpay) and an LDAC audio
+    encoder (ldacenc)
+
+-   kmssink: gained support for NV24, NV61, RGB16/BGR16 formats;
+    auto-detect NVIDIA Tegra driver
 
 Documentation improvements
 
--   this section will be filled in in due course
+-   hardware-accelerated GPU plugins will now no longer always list all
+    the element variants for all available GPUs, since those are
+    system-dependent and it’s confusing for users to see those in the
+    documentation just because the GStreamer developer who generated the
+    docs had multiple GPUs to play with at the time. Instead just show
+    the default elements.
 
-Possibly Breaking Changes
+Possibly Breaking and Other Noteworthy Behavioural Changes
+
+-   gst_parse_launch(), gst_parse_bin_from_description() and friends
+    will now error out when setting properties that don’t exist on
+    top-level bins. They were silently ignored before.
+
+-   The GstWebRTC library does not expose any objects anymore with
+    public fields. Instead properties have been added to replace that
+    functionality. If you are accessing such fields in your application,
+    switch to the corresponding properties.
 
--   FIXME: this section will be filled in in due course
--   MPEG-TS SCTE-35 API changes (FIXME: flesh out)
--   gst_parse_launch() and friends now error out on non-existing
-    properties on top-level bins where they would silently fail and
-    ignore those before.
 -   playbin and uridecodebin now emit the source-setup signal before the
     element is added to the bin and linked so that the source element is
     already configured before any scheduling query comes in, which is
-    useful for elements such as appsrc or giostreamsrc. (Merge Request)
+    useful for elements such as appsrc or giostreamsrc.
+
+-   The source element inside urisourcebin (used inside uridecodebin3
+    which is used inside playbin3) is no longer called "source". This
+    shouldn’t affect anyone hopefully, because there’s a "setup-source"
+    signal to configure the source element and no one should rely on
+    names of internal elements anyway.
+
+-   The vp8enc element now expects bps (bits per second) for the
+    "temporal-scalability-target-bitrate" property, which is consistent
+    with the "target-bitrate" property. Since additional configuration
+    is required with modern libvpx to make temporal scaling work anyway,
+    chances are that very few people will have been using this property
+
+-   vp8enc and vp9enc now default to “good quality” for the "deadline"
+    property rather then “best quality”. Having the deadline set to best
+    quality causes the encoder to be absurdly slow, most real-life users
+    will want the good quality tradeoff instead.
+
+-   The experimental GstTranscoder library API in gst-plugins-bad was
+    changed from a GObject signal-based notification mechanism to a
+    GstBus/message-based mechanism akin to GstPlayer/GstPlay.
+
+-   MPEG-TS SCTE-35 API: semantic change for SCTE-35 splice commands:
+    timestamps passed by the application should be in running time now,
+    since users of the API can’t really be expected to predict the local
+    PTS of the muxer.
+
+-   The GstContext used by souphttpsrc to share the session between
+    multiple element instances has changed. Previously it provided
+    direct access to the internal SoupSession object, now it only
+    provides access to an opaque, internal type. This change is
+    necessary because SoupSession is not thread-safe at all and can’t be
+    shared safely between arbitrary external code and souphttpsrc.
+
+-   Python bindings: GObject-introspection related Annotation fixes have
+    led to a case of a GstVideo.VideoInfo-related function signature
+    changing in the Python bindings (possibly one or two other cases
+    too). This is for a function that should never have been exposed in
+    the first place though, so the bindings are being updated to throw
+    an exception in that case, and the correct replacement API has been
+    added in form of an override.
 
 Known Issues
 
--   this section will be filled in in due course
-
--   There are a couple of known WebRTC-related regressions/blockers:
-
-    -   webrtc: DTLS setup with Chrome is broken
-    -   webrtcbin: First keyframe is usually lost
+-   nothing in particular at this point (but also see possibly breaking
+    changes section above)
 
 Contributors
 
--   this section will be filled in in due course
+Aaron Boxer, Adam Leppky, Adam Williamson, Alba Mendez, Alejandro
+González, Aleksandr Slobodeniuk, Alexander Vandenbulcke, Alex Ashley,
+Alicia Boya García, Andika Triwidada, Andoni Morales Alastruey, Andrew
+Wesie, Andrey Moiseev, Antonio Ospite, Antonio Rojas, Arthur Crippa
+Búrigo, Arun Raghavan, Ashley Brighthope, Axel Kellermann, Baek, Bastien
+Nocera, Bastien Reboulet, Benjamin Gaignard, Bing Song, Binh Truong,
+Biswapriyo Nath, Brad Hards, Brad Smith, Brady J. Garvin, Branko
+Subasic, Camilo Celis Guzman, Chris Bass, ChrisDuncanAnyvision, Chris
+White, Corentin Damman, Daniel Almeida, Daniel Knobe, Daniel Stone,
+david, David Fernandez, David Keijser, David Phung, Devarsh Thakkar,
+Dinesh Manajipet, Dmitry Samoylov, Dmitry Shusharin, Dominique Martinet,
+Doug Nazar, Ederson de Souza, Edward Hervey, Emmanuel Gil Peyrot,
+Enrique Ocaña González, Ezequiel Garcia, Fabian Orccon, Fabrice
+Fontaine, Fernando Jimenez Moreno, Florian Karydes, Francisco Javier
+Velázquez-García, François Laignel, Frederich Munch, Fredrik Pålsson,
+George Kiagiadakis, Georg Lippitsch, Göran Jönsson, Guido Günther,
+Guillaume Desmottes, Guiqin Zou, Haakon Sporsheim, Haelwenn (lanodan)
+Monnier, Haihao Xiang, Haihua Hu, Havard Graff, He Junyan, Helmut
+Januschka, Henry Wilkes, Hosang Lee, Hou Qi, Ignacio Casal Quinteiro,
+Igor Kovalenko, Ilya Kreymer, Imanol Fernandez, Jacek Tomaszewski, Jade
+Macho, Jakub Adam, Jakub Janků, Jan Alexander Steffens (heftig), Jan
+Schmidt, Jason Carrete, Jason Pereira, Jay Douglass, Jeongki Kim, Jérôme
+Laheurte, Jimmi Holst Christensen, Johan Sternerup, John Hassell, John
+Lindgren, John-Mark Bell, Jonathan Matthew, Jordan Petridis, Jose
+Quaresma, Julian Bouzas, Julien, Kai Uwe Broulik, Kasper Steensig
+Jensen, Kellermann Axel, Kevin Song, Khem Raj, Knut Inge Hvidsten, Knut
+Saastad, Kristofer Björkström, Lars Lundqvist, Lawrence Troup, Lim Siew
+Hoon, Lucas Stach, Ludvig Rappe, Luis Paulo Fernandes de Barros, Luke
+Yelavich, Mads Buvik Sandvei, Marc Leeman, Marco Felsch, Marek Vasut,
+Marian Cichy, Marijn Suijten, Marius Vlad, Markus Ebner, Mart Raudsepp,
+Matej Knopp, Mathieu Duponchelle, Matthew Waters, Matthieu De Beule,
+Mengkejiergeli Ba, Michael de Gans, Michael Olbrich, Michael Tretter,
+Michal Dzik, Miguel Paris, Mikhail Fludkov, mkba, Nazar Mokrynskyi,
+Nicholas Jackson, Nicola Murino, Nicolas Dufresne, Niklas Hambüchen,
+Nikolay Sivov, Nirbheek Chauhan, Olivier Blin, Olivier Crete, Olivier
+Crête, Paul Goulpié, Per Förlin, Peter Boba, P H, Philippe Normand,
+Philipp Zabel, Pieter Willem Jordaan, Piotrek Brzeziński, Rafał
+Dzięgiel, Rafostar, raghavendra, Raghavendra, Raju Babannavar, Raleigh
+Littles III, Randy Li, Randy Li (ayaka), Ratchanan Srirattanamet, Raul
+Tambre, reed.lawrence, Ricky Tang, Robert Rosengren, Robert Swain, Robin
+Burchell, Roman Sivriver, R S Nikhil Krishna, Ruben Gonzalez, Ruslan
+Khamidullin, Sanchayan Maity, Scott Moreau, Sebastian Dröge, Sergei
+Kovalev, Seungha Yang, Sid Sethupathi, sohwan.park, Sonny Piers, Staz M,
+Stefan Brüns, Stéphane Cerveau, Stephan Hesse, Stian Selnes, Stirling
+Westrup, Théo MAILLART, Thibault Saunier, Tim, Timo Wischer, Tim-Philipp
+Müller, Tim Schneider, Tobias Ronge, Tom Schoonjans, Tulio Beloqui,
+tyler-aicradle, U. Artie Eoff, Ung, Val Doroshchuk, VaL Doroshchuk,
+Víctor Manuel Jáquez Leal, Vivek R, Vivia Nikolaidou, Vivienne
+Watermeier, Vladimir Menshakov, Will Miller, Wim Taymans, Xabier
+Rodriguez Calvar, Xavier Claessens, Xℹ Ruoyao, Yacine Bandou, Yinhang
+Liu, youngh.lee, youngsoo.lee, yychao, Zebediah Figura, Zhang yuankun,
+Zhang Yuankun, Zhao, Zhao Zhili, , Aleksandar Topic, Antonio Ospite,
+Bastien Nocera, Benjamin Gaignard, Brad Hards, Carlos Falgueras García,
+Célestin Marot, Corentin Damman, Corentin Noël, Daniel Almeida, Daniel
+Knobe, Danny Smith, Dave Piché, Dmitry Osipenko, Fabrice Fontaine,
+fjmax, Florian Zwoch, Guillaume Desmottes, Haihua Hu, Heinrich Kruger,
+He Junyan, Jakub Adam, James Cowgill, Jan Alexander Steffens (heftig),
+Jean Felder, Jeongki Kim, Jiri Uncovsky, Joe Todd, Jordan Petridis,
+Krystian Wojtas, Marc-André Lureau, Marcin Kolny, Marc Leeman, Mark
+Nauwelaerts, Martin Reboredo, Mathieu Duponchelle, Matthew Waters,
+Mengkejiergeli Ba, Michael Gruner, Nicolas Dufresne, Nirbheek Chauhan,
+Olivier Crête, Philippe Normand, Rafał Dzięgiel, Ralf Sippl, Robert
+Mader, Sanchayan Maity, Sangchul Lee, Sebastian Dröge, Seungha Yang,
+Stéphane Cerveau, Teh Yule Kim, Thibault Saunier, Thomas Klausner, Timo
+Wischer, Tim-Philipp Müller, Tobias Reineke, Tomasz Andrzejak, Trung Do,
+Tyler Compton, Ung, Víctor Manuel Jáquez Leal, Vivia Nikolaidou, Wim
+Taymans, wngecn, Wonchul Lee, wuchang li, Xavier Claessens, Xi Ruoyao,
+Yoshiharu Hirose, Zhao,
 
 … and many others who have contributed bug reports, translations, sent
 suggestions or helped testing.
@@ -307,7 +1863,7 @@ the git 1.20 branch, which will be a stable branch.
 
 1.20.0
 
-1.20.0 is scheduled to be released around October/November 2021.
+1.20.0 is scheduled to be released around early February 2022.
 
 Schedule for 1.22
 
@@ -315,7 +1871,9 @@ Our next major feature release will be 1.22, and 1.21 will be the
 unstable development version leading up to the stable 1.22 release. The
 development of 1.21/1.22 will happen in the git main branch.
 
-The plan for the 1.22 development cycle is yet to be confirmed.
+The plan for the 1.22 development cycle is yet to be confirmed. Assuming
+no major project-wide reorganisations in the 1.22 cycle we might try and
+aim for a release around August 2022.
 
 1.22 will be backwards-compatible to the stable 1.20, 1.18, 1.16, 1.14,
 1.12, 1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series.
@@ -323,6 +1881,7 @@ The plan for the 1.22 development cycle is yet to be confirmed.
 ------------------------------------------------------------------------
 
 These release notes have been prepared by Tim-Philipp Müller with
-contributions from …
+contributions from Matthew Waters, Nicolas Dufresne, Nirbheek Chauhan,
+Sebastian Dröge and Seungha Yang.
 
 License: CC BY-SA 4.0
diff --git a/subprojects/gst-plugins-bad/RELEASE b/subprojects/gst-plugins-bad/RELEASE
index 198b702bde..a22cdf4adc 100644
--- a/subprojects/gst-plugins-bad/RELEASE
+++ b/subprojects/gst-plugins-bad/RELEASE
@@ -1,4 +1,4 @@
-This is GStreamer gst-plugins-bad 1.19.3.
+This is GStreamer gst-plugins-bad 1.19.90.
 
 GStreamer 1.19 is the development branch leading up to the next major
 stable version which will be 1.20.
diff --git a/subprojects/gst-plugins-bad/gst-plugins-bad.doap b/subprojects/gst-plugins-bad/gst-plugins-bad.doap
index afad4aefc8..3c65c70e96 100644
--- a/subprojects/gst-plugins-bad/gst-plugins-bad.doap
+++ b/subprojects/gst-plugins-bad/gst-plugins-bad.doap
@@ -35,6 +35,16 @@ real live maintainer, or some actual wide use.
 
  <release>
   <Version>
+   <revision>1.19.90</revision>
+   <branch>main</branch>
+   <name></name>
+   <created>2022-01-28</created>
+   <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-plugins-bad/gst-plugins-bad-1.19.90.tar.xz" />
+  </Version>
+ </release>
+
+ <release>
+  <Version>
    <revision>1.19.3</revision>
    <branch>main</branch>
    <name></name>
diff --git a/subprojects/gst-plugins-bad/meson.build b/subprojects/gst-plugins-bad/meson.build
index cbb9e65638..d79cb64bcf 100644
--- a/subprojects/gst-plugins-bad/meson.build
+++ b/subprojects/gst-plugins-bad/meson.build
@@ -1,5 +1,5 @@
 project('gst-plugins-bad', 'c', 'cpp',
-  version : '1.19.3.1',
+  version : '1.19.90',
   meson_version : '>= 0.59',
   default_options : [ 'warning_level=1',
                       'buildtype=debugoptimized' ])
diff --git a/subprojects/gst-plugins-base/ChangeLog b/subprojects/gst-plugins-base/ChangeLog
index 6c2406a3c6..5098601790 100644
--- a/subprojects/gst-plugins-base/ChangeLog
+++ b/subprojects/gst-plugins-base/ChangeLog
@@ -1,3 +1,18 @@
+=== release 1.19.90 ===
+
+2022-01-28 14:28:35 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* NEWS:
+	* RELEASE:
+	* gst-plugins-base.doap:
+	* meson.build:
+	  Release 1.19.90
+
+2022-01-28 14:28:28 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* ChangeLog:
+	  Update ChangeLogs for 1.19.90
+
 2022-01-27 14:22:26 +0200  Sebastian Dröge <sebastian@centricular.com>
 
 	* gst-libs/gst/rtp/gstrtphdrext.c:
diff --git a/subprojects/gst-plugins-base/NEWS b/subprojects/gst-plugins-base/NEWS
index e9f5227c15..1a512c1848 100644
--- a/subprojects/gst-plugins-base/NEWS
+++ b/subprojects/gst-plugins-base/NEWS
@@ -1,23 +1,19 @@
 GStreamer 1.20 Release Notes
 
 GStreamer 1.20 has not been released yet. It is scheduled for release in
-November 2021.
+late January / early February 2022.
 
 1.19.x is the unstable development version that is being developed in
-the git main branch and which will eventually result in 1.20, and 1.19.3
-is the current development release in that series
-
-It is expected that feature freeze will be in early October 2021,
-followed by one or two 1.19.9x pre-releases and the new 1.20 stable
-release around October/November 2021.
+the git main branch and which will eventually result in 1.20, and
+1.19.90 is the first release candidate in that series (1.20rc1).
 
 1.20 will be backwards-compatible to the stable 1.18, 1.16, 1.14, 1.12,
-1.10, 1.8, 1.6,, 1.4, 1.2 and 1.0 release series.
+1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series.
 
 See https://gstreamer.freedesktop.org/releases/1.20/ for the latest
 version of this document.
 
-Last updated: Monday 1 November 2021, 01:00 UTC (log)
+Last updated: Wednesday 26 January 2022, 01:00 UTC (log)
 
 Introduction
 
@@ -30,25 +26,971 @@ fixes and other improvements.
 
 Highlights
 
--   this section will be completed in due course
+-   Development in GitLab was switched to a single git repository
+    containing all the modules
+-   GstPlay: new high-level playback library, replaces GstPlayer
+-   WebM Alpha decoding support
+-   Encoding profiles can now be tweaked with additional
+    application-specified element properties
+-   Compositor: multi-threaded video conversion and mixing
+-   RTP header extensions: unified support in RTP depayloader and
+    payloader base classes
+-   SMPTE 2022-1 2-D Forward Error Correction support
+-   Smart encoding (passthrough) support for VP8, VP9, H.265 in
+    encodebin and transcodebin
+-   Runtime compatibility support for libsoup2 and libsoup3 (libsoup3
+    support experimental)
+-   Video decoder subframe support
+-   Video decoder automatic packet-loss, data corruption, and keyframe
+    request handling for RTP / WebRTC / RTSP
+-   MP4 and Matroska muxers now support profile/level/resolution changes
+    for H264/H265 input streams (i.e. codec data changing on the fly)
+-   MP4 muxing mode that initially creates a fragmented mp4 which is
+    converted to a regular mp4 on EOS
+-   Audio support for the WebKit Port for Embedded (WPE) web page source
+    element
+-   CUDA based video color space convert and rescale elements and
+    upload/download elements
+-   NVIDIA memory:NVMM support for OpenGL glupload and gldownload
+    elements
+-   Many WebRTC improvements
+-   The new VA-API plugin implemention fleshed out with more decoders
+    and new postproc elements
+-   AppSink API to retrieve events in addition to buffers and buffer
+    lists
+-   AppSrc gained more configuration options for the internal queue
+    (leakiness, limits in buffers and time, getters to read current
+    levels)
+-   Updated Rust bindings and many new Rust plugins
+-   Improved support for custom minimal GStreamer builds
+-   Support build against FFmpeg 5.0
+-   Linux Stateless CODEC support gained MPEG2 and VP9
+-   Windows Direct3D11/DXVA decoder gained AV1 and MPEG2 support
+-   Lots of new plugins, features, performance improvements and bug
+    fixes
 
 Major new features and changes
 
 Noteworthy new features and API
 
--   this section will be filled in in due course
+-   gst_element_get_request_pad() has been deprecated in favour of the
+    newly-added gst_element_request_pad_simple() which does the exact
+    same thing but has a less confusing name that hopefully makes clear
+    that the function request a new pad rather than just retrieves an
+    already-existing request pad.
+
+Development in GitLab was switched to a single git repository containing all the modules
+
+The GStreamer multimedia framework is a set of libraries and plugins
+split into a number of distinct modules which are released independently
+and which have so far been developed in separate git repositories in
+freedesktop.org GitLab.
+
+In addition to these separate git repositories there was a gst-build
+module that would use the Meson build systems’s subproject feature to
+download each individual module and then build everything in one go. It
+would also provide an uninstalled development environment that made it
+easy to work on GStreamer and use or test versions other than the
+system-installed GStreamer version.
+
+All of these modules have now (as of 28 September 2021) been merged into
+a single git repository (“Mono repository” or “monorepo”) which should
+simplify development workflows and continuous integration, especially
+where changes need to be made to multiple modules at once.
+
+This mono repository merge will primarily affect GStreamer developers
+and contributors and anyone who has workflows based on the GStreamer git
+repositories.
+
+The Rust bindings and Rust plugins modules have not been merged into the
+mono repository at this time because they follow a different release
+cycle.
+
+The mono repository lives in the existing GStreamer core git repository
+in GitLab in the new main branch and all future development will happen
+on this branch.
+
+Modules will continue to be released as separate tarballs.
+
+For more details, please see the GStreamer mono repository FAQ.
+
+GstPlay: new high-level playback library replacing GstPlayer
+
+-   GstPlay is a new high-level playback library that replaces the older
+    GstPlayer API. It is basically the same API as GstPlayer but
+    refactored to use bus messages for application notifications instead
+    of GObject signals. There is still a signal adapter object for those
+    who prefer signals. Since the existing GstPlayer API is already in
+    use in various applications, it didn’t seem like a good idea to
+    break it entirely. Instead a new API was added, and it is expected
+    that this new GstPlay API will be moved to gst-plugins-base in
+    future.
+
+-   The existing GstPlayer API is scheduled for deprecation and will be
+    removed at some point in the future (e.g. in GStreamer 1.24), so
+    application developers are urged to migrate to the new GstPlay API
+    at their earliest convenience.
+
+WebM alpha decoding
+
+-   Implement WebM alpha decoding (VP8/VP9 with alpha), which required
+    support and additions in various places. This is supported both with
+    software decoders and hardware-accelerated decoders.
+
+-   VP8/VP9 don’t support alpha components natively in the codec, so the
+    way this is implemented in WebM is by encoding the alpha plane with
+    transparency data as a separate VP8/VP9 stream. Inside the WebM
+    container (a variant of Matroska) this is coded as a single video
+    track with the “normal” VP8/VP9 video data making up the main video
+    data and each frame of video having an encoded alpha frame attached
+    to it as extra data ("BlockAdditional").
+
+-   matroskademux has been extended extract this per-frame alpha side
+    data and attach it in form of a GstVideoCodecAlphaMeta to the
+    regular video buffers. Note that this new meta is specific to this
+    VP8/VP9 alpha support and can’t be used to just add alpha support to
+    other codecs that don’t support it. Lastly, matroskademux also
+    advertises the fact that the streams contain alpha in the caps.
+
+-   The new codecalpha plugin contains various bits of infrastructure to
+    support autoplugging and debugging:
+
+    -   codecalphademux splits out the alpha stream from the metas on
+        the regular VP8/VP9 buffers
+    -   alphacombine takes two decoded raw video streams (one alpha, one
+        the regular video) and combines it into a video stream with
+        alpha
+    -   vp8alphadecodebin + vp9alphadecodebin are wrapper bins that use
+        the regular vp8dec and vp9dec software decoders to decode
+        regular and alpha streams and combine them again. To decodebin
+        these look like regular decoders which ju
+    -   The V4L2 CODEC plugin has stateless VP8/VP9 decoders that can
+        decode both alpha and non-alpha stream with a single decoder
+        instance
+
+-   A new AV12 video format was added which is basically NV12 with an
+    alpha plane, which is more convenient for many hardware-accelerated
+    decoders.
+
+-   Watch Nicolas Dufresne’s LCA 2022 talk “Bringing WebM Alpha support
+    to GStreamer” for all the details and a demo.
+
+RTP Header Extensions Base Class and Automatic Header Extension Handling in RTP Payloaders and Depayloaders
+
+-   RTP Header Extensions are specified in RFC 5285 and provide a way to
+    add small pieces of data to RTP packets in between the RTP header
+    and the RTP payload. This is often used for per-frame metadata,
+    extended timestamps or other application-specific extra data. There
+    are several commonly-used extensions specified in various RFCs, but
+    senders are free to put any kind of data in there, as long as sender
+    and receiver both know what that data is. Receivers that don’t know
+    about the header extensions will just skip the extra data without
+    ever looking at it. These header extensions can often be combined
+    with any kind of payload format, so may need to be supported by many
+    RTP payloader and depayloader elements.
+
+-   Inserting and extracting RTP header extension data has so far been a
+    bit inconvenient in GStreamer: There are functions to add and
+    retrieve RTP header extension data from RTP packets, but nothing
+    works automatically, even for common extensions. People would have
+    to do the insertion/extraction either in custom elements
+    before/after the RTP payloader/depayloader, or inside pad probes,
+    which isn’t very nice.
+
+-   This release adds various pieces of new infrastructure for generic
+    RTP header extension handling, as well as some implementations for
+    common extensions:
+
+    -   GstRTPHeaderExtension is a new helper base class for reading and
+        writing RTP header extensions. Nominally this subclasses
+        GstElement, but only so these extensions are stored in the
+        registry where they can be looked up by URI or name. They don’t
+        have pads and don’t get added to the pipeline graph as an
+        element.
+
+    -   "add-extension" and "clear-extension" action signals on RTP
+        payloaders and depayloaders for manual extension management
+
+    -   The "request-extension" signal will be emitted if an extension
+        is encountered that requires explicit mapping by the application
+
+    -   new "auto-header-extension" property on RTP payloaders and
+        depayloaders for automatic handling of known header extensions.
+        This is enabled by default. The extensions must be signalled via
+        caps / SDP.
+
+    -   RTP header extension implementations:
+
+        -   rtphdrextclientaudiolevel: Client-to-Mixer Audio Level
+            Indication (RFC 6464) (also see below)
+        -   rtphdrextcolorspace: Color Space extension, extends RTP
+            packets with color space and high dynamic range (HDR)
+            information
+        -   rtphdrexttwcc: Transport Wide Congestion Control support
+
+-   gst_rtp_buffer_remove_extension_data() is a new helper function to
+    remove an RTP header extension from an RTP buffer
+
+-   The existing gst_rtp_buffer_set_extension_data() now also supports
+    shrinking the extension data in size
+
+AppSink and AppSrc improvements
+
+-   appsink: new API to pull events out of appsink in addition to
+    buffers and buffer lists.
+
+    There was previously no way for users to receive incoming events
+    from appsink properly serialised with the data flow, even if they
+    are serialised events. The reason for that is that the only way to
+    intercept events was via a pad probe on the appsink sink pad, but
+    there is also internal queuing inside of appsink, so it’s difficult
+    to ascertain the right order of everything in all cases.
+
+    There is now a new "new-serialized-event" signal which will be
+    emitted when there’s a new event pending (just like the existing
+    "new-sample" signal). The "emit-signals" property must be set to
+    TRUE in order to activate this (but it’s also fine to just pull from
+    the application thread without using the signals).
+
+    gst_app_sink_pull_object() and gst_app_sink_try_pull_object() can be
+    used to pull out either an event or a new sample carrying a buffer
+    or buffer list, whatever is next in the queue.
+
+    EOS events will be filtered and will not be returned. EOS handling
+    can be done the ususal way, same as with _pull_sample().
+
+-   appsrc: allow configuration of internal queue limits in time and
+    buffers and add leaky mode.
+
+    There is internal queuing inside appsrc so the application thread
+    can push data into the element which will then be picked up by the
+    source element’s streaming thread and pushed into the pipeline from
+    that streaming thread. This queue is unlimited by default and until
+    now it was only possible to set a maximum size limit in bytes. When
+    that byte limit is reached, the pushing thread (application thread)
+    would be blocked until more space becomes available.
+
+    A limit in bytes is not particularly useful for many use cases, so
+    now it is possible to also configure limits in time and buffers
+    using the new "max-time" and "max-buffers" properties. Of course
+    there are also matching new read-only"current-level-buffers" and
+    "current-level-time properties" properties to query the current fill
+    level of the internal queue in time and buffers.
+
+    And as if that wasn’t enough the internal queue can also be
+    configured as leaky using the new "leaky-type" property. That way
+    when the queue is full the application thread won’t be blocked when
+    it tries to push in more data, but instead either the new buffer
+    will be dropped or the oldest data in the queue will be dropped.
+
+Better string serialization of nested GstCaps and GstStructures
+
+-   New string serialisation format for structs and caps that can handle
+    nested structs and caps properly by using brackets to delimit nested
+    items (e.g. some-struct, some-field=[nested-struct, nested=true]).
+    Unlike the default format the new variant can also support more than
+    one level of nesting. For backwards-compatibility reasons the old
+    format is still output by default when serialising caps and structs
+    using the existing API. The new functions gst_caps_serialize() and
+    gst_structure_serialize() can be used to output strings in the new
+    format.
+
+Convenience API for custom GstMetas
+
+-   New convenience API to register and create custom GstMetas:
+    gst_meta_register_custom() and gst_buffer_add_custom_meta(). Such
+    custom meta is backed by a GstStructure and does not require that
+    users of the API expose their GstMeta implementation as public API
+    for other components to make use of it. In addition, it provides a
+    simpler interface by ignoring the impl vs. api distinction that the
+    regular API exposes. This new API is meant to be the meta
+    counterpart to custom events and messages, and to be more convenient
+    than the lower-level API when the absolute best performance isn’t a
+    requirement. The reason it’s less performant than a “proper” meta is
+    that a proper meta is just a C struct in the end whereas this goes
+    through the GstStructure API which has a bit more overhead, which
+    for most scenarios is negligible however. This new API is useful for
+    experimentation or proprietary metas, but also has some limitations:
+    it can only be used if there’s a single producer of these metas;
+    it’s not allowed to register the same custom meta multiple times or
+    from multiple places.
+
+Additional Element Properties on Encoding Profiles
+
+-   GstEncodingProfile: The new "element-properties" and
+    gst_encoding_profile_set_element_properties() API allows
+    applications to set additional element properties on encoding
+    profiles to configure muxers and encoders. So far the encoding
+    profile template was the only place where this could be specified,
+    but often what applications want to do is take a ready-made encoding
+    profile shipped by GStreamer or the application and then tweak the
+    settings on top of that, which is now possible with this API. Since
+    applications can’t always know in advance what encoder element will
+    be used in the end, it’s even possible to specify properties on a
+    per-element basis.
+
+    Encoding Profiles are used in the encodebin, transcodebin and
+    camerabin elements and APIs to configure output formats (containers
+    and elementary streams).
+
+Audio Level Indication Meta for RFC 6464
+
+-   New GstAudioLevelMeta containing Audio Level Indication as per RFC
+    6464
+
+-   The level element has been updated to add GstAudioLevelMeta on
+    buffers if the "audio-level-meta" property is set to TRUE. This can
+    then in turn be picked up by RTP payloaders to signal the audio
+    level to receivers through RTP header extensions (see above).
+
+-   New Client-to-Mixer Audio Level Indication (RFC6464) RTP Header
+    Extension which should be automatically created and used by RTP
+    payloaders and depayloaders if their "auto-header-extension"
+    property is enabled and if the extension is part of the RTP caps.
+
+Automatic packet loss, data corruption and keyframe request handling for video decoders
+
+-   The GstVideoDecoder base class has gained various new APIs to
+    automatically handle packet loss and data corruption better by
+    default, especially in RTP, RTSP and WebRTC streaming scenarios, and
+    to give subclasses more control about how they want to handle
+    missing data:
+
+    -   Video decoder subclasses can mark output frames as corrupted via
+        the new GST_VIDEO_CODEC_FRAME_FLAG_CORRUPTED flag
+
+    -   A new "discard-corrupted-frames" property allows applications to
+        configure decoders so that corrupted frames are directly
+        discarded instead of being forwarded inside the pipeline. This
+        is a replacement for the "output-corrupt" property of the FFmpeg
+        decoders.
+
+    -   RTP depayloaders can now signal to decoders that data is missing
+        when sending GAP events for lost packets. GAP events can be sent
+        for various reason in a GStreamer pipeline. Often they are just
+        used to let downstream elements know that there isn’t a buffer
+        available at the moment, so downstream elements can move on
+        instead of waiting for one. They are also sent by RTP
+        depayloaders in the case that packets are missing, however, and
+        so far a decoder was not able to differentiate the two cases.
+        This has been remedied now: GAP events can be decorated with
+        gst_event_set_gap_flags() and GST_GAP_FLAG_MISSING_DATA to let
+        decoders now what happened, and decoders can then use that in
+        some cases to handle missing data better.
+
+    -   The GstVideoDecoder::handle_missing_data vfunc was added to
+        inform subclasses about packet loss or missing data and let them
+        handle it in their own way if they like.
+
+    -   gst_video_decoder_set_needs_sync_point() lets subclasses signal
+        that they need the stream to start with a sync point. If
+        enabled, the base class will discard all non-sync point frames
+        in the beginning and after a flush and does not pass them to the
+        subclass. Furthermore, if the first frame is not a sync point,
+        the base class will try and request a sync frame from upstream
+        by sending a force-key-unit event (see next items).
+
+    -   New "automatic-request-sync-points" and
+        "automatic-request-sync-point-flags" properties to automatically
+        request sync points when needed, e.g. on packet loss or if the
+        first frame is not a keyframe. Applications may want to enable
+        this on decoders operating in e.g. RTP/WebRTC/RTSP receiver
+        pipelines.
+
+    -   The new "min-force-key-unit-interval" property can be used to
+        ensure there’s a minimal interval between keyframe requests to
+        upstream (and/or the sender) and we’re not flooding the sender
+        with key unit requests.
+
+    -   gst_video_decoder_request_sync_point() allows subclasses to
+        request a new sync point (e.g. if they choose to do their own
+        missing data handling). This will still honour the
+        "min-force-key-unit-interval" property if set.
+
+Improved support for custom minimal GStreamer builds
+
+-   Element registration and registration of other plugin features
+    inside plugin init functions has been improved in order to
+    facilitate minimal custom GStreamer builds.
+
+-   A number of new macros have been added to declare and create
+    per-element and per-pluginfeature register functions in all plugins,
+    and then call those from the per-plugin plugin_init functions:
+
+    -   GST_ELEMENT_REGISTER_DEFINE,
+        GST_DEVICE_PROVIDER_REGISTER_DEFINE,
+        GST_DYNAMIC_TYPE_REGISTER_DEFINE, GST_TYPE_FIND_REGISTER_DEFINE
+        for the actual registration call with GStreamer
+    -   GST_ELEMENT_REGISTER, GST_DEVICE_PROVIDER_REGISTER,
+        GST_DYNAMIC_TYPE_REGISTER, GST_PLUGIN_STATIC_REGISTER,
+        GST_TYPE_FIND_REGISTER to call the registration function defined
+        by the REGISTER_DEFINE macro
+    -   GST_ELEMENT_REGISTER_DECLARE,
+        GST_DEVICE_PROVIDER_REGISTER_DECLARE,
+        GST_DYNAMIC_TYPE_REGISTER_DECLARE,
+        GST_TYPE_FIND_REGISTER_DECLARE to declare the registration
+        function defined by the REGISTER_DEFINE macro
+    -   and various variants for advanced use cases.
+
+-   This means that applications can call the per-element and
+    per-pluginfeature registration functions for only the elements they
+    need instead of registering plugins as a whole with all kinds of
+    elements that may not be required (e.g. encoder and decoder instead
+    of just decoder). In case of static linking all unused functions and
+    their dependencies would be removed in this case by the linker,
+    which helps minimise binary size for custom builds.
+
+-   gst_init() will automatically call a gst_init_static_plugins()
+    function if one exists.
+
+-   See the GStreamer static build documentation and Stéphane’s blog
+    post Generate a minimal GStreamer build, tailored to your needs for
+    more details.
 
 New elements
 
--   this section will be filled in in due course
+-   New aesdec and aesenc elements for AES encryption and decryption in
+    a custom format.
+
+-   New encodebin2 element with dynamic/sometimes source pads in order
+    to support the option of doing the muxing outside of encodebin,
+    e.g. in combination with a splitmuxsink.
+
+-   New fakeaudiosink and videocodectestsink elements for testing and
+    debugging (see below for more details)
+
+-   rtpisacpay, rtpisacdepay: new RTP payloader and depayloader for iSAC
+    audio codec
+
+-   rtpst2022-1-fecdec, rtpst2022-1-fecenc: new elements providing SMPTE
+    2022-1 2-D Forward Error Correction. More details in Mathieu’s blog
+    post.
+
+-   isac: new plugin wrapping the Internet Speech Audio Codec reference
+    encoder and decoder from the WebRTC project.
+
+-   asio: plugin for Steinberg ASIO (Audio Streaming Input/Output) API
+
+-   gssrc, gssink: add source and sink for Google Cloud Storage
+
+-   onnx: new plugin to apply ONNX neural network models to video
+
+-   openaptx: aptX and aptX-HD codecs using libopenaptx (v0.2.0)
+
+-   qroverlay, debugqroverlay: new elements that allows overlaying data
+    on top of video in form of a QR code
+
+-   cvtracker: new OpenCV-based tracker element
+
+-   av1parse, vp9parse: new parsers for AV1 and VP9 video
+
+-   va: work on the new VA-API plugin implementation for
+    hardware-accelerated video decoding and encoding has continued at
+    pace, with various new decoders and filters having joined the
+    initial vah264dec:
+
+    -   vah265dec: VA-API H.265 decoder
+    -   vavp8dec: VA-API VP8 decoder
+    -   vavp9dec: VA-API VP9 decoder
+    -   vaav1dec: VA-API AV1 decoder
+    -   vampeg2dec: VA-API MPEG-2 decoder
+    -   vadeinterlace: : VA-API deinterlace filter
+    -   vapostproc: : VA-API postproc filter (color conversion,
+        resizing, cropping, color balance, video rotation, skin tone
+        enhancement, denoise, sharpen)
+
+    See Víctor’s blog post “GstVA in GStreamer 1.20” for more details
+    and what’s coming up next.
+
+-   vaapiav1dec: new AV1 decoder element (in gstreamer-vaapi)
+
+-   msdkav1dec: hardware-accelerated AV1 decoder using the Intel Media
+    SDK / oneVPL
+
+-   nvcodec plugin for NVIDIA NVCODEC API for hardware-accelerated video
+    encoding and decoding:
+
+    -   cudaconvert, cudascale: new CUDA based video color space convert
+        and rescale elements
+    -   cudaupload, cudadownload: new helper elements for memory
+        transfer between CUDA and system memory spaces
+    -   nvvp8sldec, nvvp9sldec: new GstCodecs-based VP8/VP9 decoders
+
+-   Various new hardware-accelerated elements for Windows:
+
+    -   d3d11screencapturesrc: new desktop capture element, including a
+        GstDeviceProvider implementation to enumerate/select target
+        monitors for capture.
+    -   d3d11av1dec and d3d11mpeg2dec: AV1 and MPEG-2 decoders
+    -   d3d11deinterlace: deinterlacing filter
+    -   d3d11compositor: video composing element
+    -   see Windows section below for more details
+
+-   new Rust plugins:
+
+    -   audiornnoise: Removes noise from an audio stream
+    -   awstranscribeparse: Parses AWS audio transcripts into timed text
+        buffers
+    -   ccdetect: Detects if valid closed captions are present in a
+        closed captions stream
+    -   cea608tojson: Converts CEA-608 Closed Captions to a JSON
+        representation
+    -   cmafmux: CMAF fragmented MP4 muxer
+    -   dashmp4mux: DASH fragmented MP4 muxer
+    -   isofmp4mux: ISO fragmented MP4 muxer
+    -   ebur128level: EBU R128 Loudness Level Measurement
+    -   ffv1dec: FFV1 video decoder
+    -   gtk4paintablesink: GTK4 video sink, which provides a
+        GdkPaintable that can be rendered in various widgets
+    -   hlssink3: HTTP Live Streaming sink
+    -   hrtfrender: Head-Related Transfer Function (HRTF) renderer
+    -   hsvdetector: HSV colorspace detector
+    -   hsvfilter: HSV colorspace filter
+    -   jsongstenc: Wraps buffers containing any valid top-level JSON
+        structures into higher level JSON objects, and outputs those as
+        ndjson
+    -   jsongstparse: Parses ndjson as output by jsongstenc
+    -   jsontovtt: converts JSON to WebVTT subtitles
+    -   regex: Applies regular expression operations on text
+    -   roundedcorners: Adds rounded corners to video
+    -   spotifyaudiosrc: Spotify source
+    -   textahead: Display upcoming text buffers ahead (e.g. for
+        Karaoke)
+    -   transcriberbin: passthrough bin that transcribes raw audio to
+        closed captions using awstranscriber and puts the captions as
+        metas onto the video
+    -   tttojson: Converts timed text to a JSON representation
+    -   uriplaylistbin: Playlist source bin
+    -   webpdec-rs: WebP image decoder with animation support
+
+-   New plugin codecalpha with elements to assist with WebM Alpha
+    decoding
+
+    -   codecalphademux: Split stream with GstVideoCodecAlphaMeta into
+        two streams
+    -   alphacombine: Combine two raw video stream (I420 or NV12) as one
+        stream with alpha channel (A420 or AV12)
+    -   vp8alphadecodebin: A bin to handle software decoding of VP8 with
+        alpha
+    -   vp9alphadecodebin: A bin to handle software decoding of VP9 with
+        alpha
+
+-   New hardware accelerated elements for Linux:
+
+    -   v4l2slmpeg2dec: Support for Linux Stateless MPEG2 decoders
+    -   v4l2slvp9dec: Support for Linux Stateless VP9 decoders
+    -   v4l2slvp8alphadecodebin: Support HW accelerated VP8 with alpha
+        layer decoding
+    -   v4l2slvp9alphadecodebin: Support HW accelerated VP9 with alpha
+        layer decoding
 
 New element features and additions
 
--   this section will be filled in in due course
+-   assrender: handle more font mime types; better interaction with
+    matroskademux for embedded fonts
+
+-   audiobuffersplit: Add support for specifying output buffer size in
+    bytes (not just duration)
+
+-   audiolatency: new "samplesperbuffer" property so users can configure
+    the number of samples per buffer. The default value is 240 samples
+    which is equivalent to 5ms latency with a sample rate of 48000,
+    which might be larger than actual buffer size of audio capture
+    device.
+
+-   audiomixer, audiointerleave, GstAudioAggregator: now keep a count of
+    samples that are dropped or processed as statistic and can be made
+    to post QoS messages on the bus whenever samples are dropped by
+    setting the "qos-messages" property on input pads.
+
+-   audiomixer, compositor: improved handling of new inputs added at
+    runtime. New API was added to the GstAggregator base class to allow
+    subclasses to opt into an aggregation mode where inactive pads are
+    ignored when processing input buffers
+    (gst_aggregator_set_ignore_inactive_pads(),
+    gst_aggregator_pad_is_inactive()). An “inactive pad” in this context
+    is a pad which, in live mode, hasn’t yet received a first buffer,
+    but has been waited on at least once. What would happen usually in
+    this case is that the aggregator would wait for data on this pad
+    every time, up to the maximum configured latency. This would
+    inadvertently push mixer elements in live mode to the configured
+    latency envelope and delay processing when new inputs are added at
+    runtime until these inputs have actually produced data. This is
+    usually undesirable. With this new API, new inputs can be added
+    (requested) and configured and they won’t delay the data processing.
+    Applications can opt into this new behaviour by setting the
+    "ignore-inactive-pads" property on compositor, audiomixer or other
+    GstAudioAggregator-based elements.
+
+-   cccombiner: implement “scheduling” of captions. So far cccombiner’s
+    behaviour was essentially that of a funnel: it strictly looked at
+    input timestamps to associate together video and caption buffers.
+    Now it will try to smoothly schedule caption buffers in order to
+    have exactly one per output video buffer. This might involve
+    rewriting input captions, for example when the input is CDP then
+    sequence counters are rewritten, time codes are dropped and
+    potentially re-injected if the input video frame had a time code
+    meta. This can also lead to the input drifting from synchronization,
+    when there isn’t enough padding in the input stream to catch up. In
+    that case the element will start dropping old caption buffers once
+    the number of buffers in its internal queue reaches a certain limit
+    (configurable via the "max-scheduled" property). The new original
+    funnel-like behaviour can be restored by setting the "scheduling"
+    property to FALSE.
+
+-   ccconverter: new "cdp-mode" property to specify which sections to
+    include in CDP packets (timecode, CC data, service info). Various
+    software, including ffmpeg’s Decklink support, fails parsing CDP
+    packets that contain anything but CC data in the CDP packets.
+
+-   clocksync: new "sync-to-first" property for automatic timestamp
+    offset setup: if set clocksync will set up the "ts-offset" value
+    based on the first buffer and the pipeline’s running time when the
+    first buffer arrived. The newly configured "ts-offset" in this case
+    would be the value that allows outputting the first buffer without
+    waiting on the clock. This is useful for example to feed a non-live
+    input into an already-running pipeline.
+
+-   compositor:
+
+    -   multi-threaded input conversion and compositing. Set the
+        "max-threads" property to activate this.
+    -   new "sizing-policy" property to support display aspect ratio
+        (DAR)-aware scaling. By default the image is scaled to fill the
+        configured destination rectangle without padding and without
+        keeping the aspect ratio. With sizing-policy=keep-aspect-ratio
+        the input image is scaled to fit the destination rectangle
+        specified by GstCompositorPad:{xpos, ypos, width, height}
+        properties preserving the aspect ratio. As a result, the image
+        will be centered in the destination rectangle with padding if
+        necessary.
+    -   new "zero-size-is-unscaled" property on input pads. By default
+        pad width=0 or pad height=0 mean that the stream should not be
+        scaled in that dimension. But if the "zero-size-is-unscaled"
+        property is set to FALSE a width or height of 0 is instead
+        interpreted to mean that the input image on that pad should not
+        be composited, which is useful when creating animations where an
+        input image is made smaller and smaller until it disappears.
+    -   improved handling of new inputs at runtime via
+        "ignore-inactive-pads"property (see above for details)
+    -   allow output format with alpha even if none of the inputs have
+        alpha (also glvideomixer and other GstVideoAggregator
+        subclasses)
+
+-   dashsink: add h265 codec support and signals for allowing custom
+    playlist/fragment output
+
+-   decodebin3:
+
+    -   improved decoder selection, especially for hardware decoders
+    -   make input activation “atomic” when adding inputs dynamically
+    -   better interleave handling: take into account decoder latency
+        for interleave size
+
+-   decklink:
+
+    -   Updated DeckLink SDK to 11.2 to support DeckLink 8K Pro
+    -   decklinkvideosrc:
+        -   More accurate and stable capture timestamps: use the
+            hardware reference clock time when the frame was finished
+            being captured instead of a clock time much further down the
+            road.
+        -   Automatically detect widescreen vs. normal NTSC/PAL
+
+-   encodebin:
+
+    -   add “smart encoding” support for H.265, VP8 and VP9 (i.e. only
+        re-encode where needed and otherwise pass through encoded video
+        as-is).
+    -   H264/H265 smart encoding improvements: respect user-specified
+        stream-format, but if not specified default to avc3/hvc1 with
+        in-band SPS/PPS/VPS signalling for more flexibility.
+    -   new encodebin2 element with dynamic/sometimes source pads in
+        order to support the option of doing the muxing outside of
+        encodebin, e.g. in combination with splitmuxsink.
+    -   add APIs to set element properties on encoding profiles (see
+        below)
+
+-   errorignore: new "ignore-eos" property to also ignore FLOW_EOS from
+    downstream elements
+
+-   giosrc: add support for growing source files: applications can
+    specify that the underlying file being read is growing by setting
+    the "is-growing" property. If set, the source won’t EOS when it
+    reaches the end of the file, but will instead start monitoring it
+    and will start reading data again whenever a change is detected. The
+    new "waiting-data" and "done-waiting-data" signals keep the
+    application informed about the current state.
+
+-   gtksink, gtkglsink:
+
+    -   scroll event support: forwarded as navigation events into the
+        pipeline
+    -   "video-aspect-ratio-override" property to force a specific
+        aspect ratio
+    -   "rotate-method" property and support automatic rotation based on
+        image tags
+
+-   identity: new "stats" property allows applications to retrieve the
+    number of bytes and buffers that have passed through so far.
+
+-   interlace: add support for more formats, esp 10-bit, 12-bit and
+    16-bit ones
+
+-   jack: new "low-latency" property for automatic latency-optimized
+    setting and "port-names" property to select ports explicitly
+
+-   jpegdec: support output conversion to RGB using libjpeg-turbo (for
+    certain input files)
+
+-   line21dec:
+
+    -   "mode" property to control whether and how detected closed
+        captions should be inserted in the list of existing close
+        caption metas on the input frame (if any): add, drop, or
+        replace.
+    -   "ntsc-only" property to only look for captions if video has NTSC
+        resolution
+
+-   line21enc: new "remove-caption-meta" to remove metas from output
+    buffers after encoding the captions into the video data; support for
+    CDP closed captions
+
+-   matroskademux, matroskamux: Add support for ffv1, a lossless
+    intra-frame video coding format.
+
+-   matroskamux: accept in-band SPS/PPS/VPS for H264 and H265
+    (i.e. stream-format avc3 and hev1) which allows on-the-fly
+    profile/level/resolution changes.
+
+-   matroskamux: new "cluster-timestamp-offset" property, useful for use
+    cases where the container timestamps should map to some absolute
+    wall clock time, for example.
+
+-   rtpsrc: add "caps" property to allow explicit setting of the caps
+    where needed
+
+-   mpegts: support SCTE-35 passthrough via new "send-scte35-events"
+    property on MPEG-TS demuxer tsdemux. When enabled, SCTE 35 sections
+    (eg ad placement opportunities) are forwarded as events donwstream
+    where they can be picked up again by mpegtsmux. This required a
+    semantic change in the SCTE-35 section API: timestamps are now in
+    running time instead of muxer pts.
+
+-   tsdemux: Handle PCR-less MPEG-TS streams; more robust timestamp
+    handling in certain corner cases and for poorly muxed streams.
+
+-   mpegtsmux:
+
+    -   More conformance improvements to make MPEG-TS analyzers happy:
+        -   PCR timing accuracy: Improvements to the way mpegtsmux
+            outputs PCR observations in CBR mode, so that a PCR
+            observation is always inserted when needed, so that we never
+            miss the configured pcr-interval, as that triggers various
+            MPEG-TS analyser errors.
+        -   Improved PCR/SI scheduling
+    -   Don’t write PCR until PAT/PMT are output to make sure streams
+        start cleanly with a PAT/PMT.
+    -   Allow overriding the automatic PMT PID selection via
+        application-supplied PMT_%d fields in the prog-map
+        structure/property.
+
+-   mp4mux:
+
+    -   new "first-moov-then-finalise" mode for fragmented output where
+        the output will start with a self-contained moov atom for the
+        first fragment, and then produce regular fragments. Then at the
+        end when the file is finalised, the initial moov is invalidated
+        and a new moov is written covering the entire file. This way the
+        file is a “fragmented mp4” file while it is still being written
+        out, and remains playable at all times, but at the end it is
+        turned into a regular mp4 file (with former fragment headers
+        remaining as unused junk data in the file).
+    -   support H.264 avc3 and H.265 hvc1 stream formats as input where
+        the codec data is signalled in-band inside the bitstream instead
+        of caps/file headers.
+    -   support profile/level/resolution changes for H264/H265 input
+        streams (i.e. codec data changing on the fly). Each codec_data
+        is put into its own SampleTableEntry inside the stsd, unless the
+        input is in avc3 stream format in which case it’s written
+        in-band an not in the headers.
+
+-   multifilesink: new ""min-keyframe-distance"" property to make
+    minimum distance between keyframes in next-file=key-frame mode
+    configurable instead of hard-coding it to 10 seconds.
+
+-   mxfdemux has seen a big refactoring to support non-frame wrappings
+    and more accurate timestamp/seek handling for some formats
+
+-   msdk plugin for hardware-accelerated video encoding and decoding
+    using the Intel Media SDK:
+
+    -   oneVPL support (Intel oneAPI Video Processing Library)
+    -   AV1 decoding support
+    -   H.264 decoder now supports constrained-high and progressive-high
+        profiles
+    -   H.264 encoder:
+        -   more configuration options (properties):
+            "intra-refresh-type", "min-qp" , "max-qp", "p-pyramid",
+            "dblk-idc"
+    -   H.265 encoder:
+        -   can output main-still-picture profile
+        -   now inserts HDR SEIs (mastering display colour volume and
+            content light level)
+        -   more configuration options (properties):
+            "intra-refresh-type", "min-qp" , "max-qp", "p-pyramid",
+            "b-pyramid", "dblk-idc", "transform-skip"
+        -   support for RGB 10bit format
+    -   External bitrate control in encoders
+    -   Video post proc element msdkvpp gained support for 12-bit pixel
+        formats P012_LE, Y212_LE and Y412_LE
+
+-   nvh264sldec: interlaced stream support
+
+-   openh264enc: support main, high, constrained-high and
+    progressive-high profiles
+
+-   openjpeg: support for multithreaded decoding and encoding
+
+-   rtspsrc: now supports IPv6 also for tunneled mode (RTSP-over-HTTP);
+    new "ignore-x-server-reply" property to ignore the
+    x-server-ip-address server header reply in case of HTTP tunneling,
+    as it is often broken.
+
+-   souphttpsrc: Runtime compatibility support for libsoup2 and
+    libsoup3. libsoup3 is the latest major version of libsoup, but
+    libsoup2 and libsoup3 can’t co-exist in the same process because
+    there is no namespacing or versioning for GObject types. As a
+    result, it would be awkward if the GStreamer souphttpsrc plugin
+    linked to a specific version of libsoup, because it would only work
+    with applications that use the same version of libsoup. To make this
+    work, the soup plugin now tries to determine the libsoup version
+    used by the application (and its other dependencies) at runtime on
+    systems where GStreamer is linked dynamically. libsoup3 support is
+    still considered somewhat experimental at this point.
+
+-   srtsrc, srtsink: add signals for the application to accept/reject
+    incoming connections
+
+-   timeoverlay: new elapsed-running-time time mode which shows the
+    running time since the first running time (and each flush-stop).
+
+-   udpsrc: new timestamping mode to retrieve packet receive timestamps
+    from the kernel via socket control messages (SO_TIMESTAMPNS) on
+    supported platforms
+
+-   uritranscodebin: new setup-source and element-setup signals for
+    applications to configure elements used
+
+-   v4l2codecs plugin gained support for 4x4 and 32x32 tile formats
+    enabling some platforms or direct renders. Important memory usage
+    improvement.
+
+-   v4l2slh264dec now implements the final Linux uAPI as shipped on
+    Linux 5.11 and later.
+
+-   valve: add "drop-mode" property and provide two new modes of
+    operation: in drop-mode=forward-sticky-events sticky events
+    (stream-start, segment, tags, caps, etc.) are forwarded downstream
+    even when dropping is enabled; drop-mode=transform-to-gap will in
+    addition also convert buffers into gap events when dropping is
+    enabled, which lets downstream elements know that time is advancing
+    and might allow for preroll in many scenarios. By default all events
+    and all buffers are dropped when dropping is enabled, which can
+    cause problems with caps negotiation not progressing or branches not
+    prerolling when dropping is enabled.
+
+-   videocrop: support for many more pixel formats, e.g. planar YUV
+    formats with > 8bits and GBR* video formats; can now also accept
+    video not backed by system memory as long as downstream supports the
+    GstCropMeta
+
+-   videotestsrc: new smpte-rp-219 pattern for SMPTE75 RP-219 conformant
+    color bars
+
+-   vp8enc: finish support for temporal scalability: two new properties
+    ("temporal-scalability-layer-flags",
+    "temporal-scalability-layer-sync-flags") and a unit change on the
+    "temporal-scalability-target-bitrate" property (now expects bps);
+    also make temporal scalability details available to RTP payloaders
+    as buffer metadata.
+
+-   vp9enc: new properties to tweak encoder performance:
+
+    -   "aq-mode" to configure adaptive quantization modes
+    -   "frame-parallel-decoding" to configure whether to create a
+        bitstream that reduces decoding dependencies between frames
+        which allows staged parallel processing of more than one video
+        frames in the decoder. (Defaults to TRUE)
+    -   "row-mt", "tile-columns" and "tile-rows" so multithreading can
+        be enabled on a per-tile basis, instead of on a per tile-column
+        basis. In combination with the new "tile-rows" property, this
+        allows the encoder to make much better use of the available CPU
+        power.
+
+-   vp9dec, vp9enc: add support for 10-bit 4:2:0 and 4:2:2 YUV, as well
+    as 8-bit 4:4:4
+
+-   vp8enc, vp9enc now default to “good quality” for the deadline
+    property rather then “best quality”. Having the deadline set to best
+    quality causes the encoder to be absurdly slow, most real-life users
+    will prefer good-enough quality with better performance instead.
+
+-   wpesrc:
+
+    -   implement audio support: a new sometimes source pad will be
+        created for each audio stream created by the web engine.
+    -   move wpesrc to wpevideosrc and add a wrapper bin wpesrc to also
+        support audio
+    -   also handles web:// URIs now (same as cefsrc)
+    -   post messages with the estimated load progress on the bus
 
-Plugin and library moves
+-   x265enc: add negative DTS support, which means timestamps are now
+    offset by 1h same as with x264enc
+
+RTP Payloaders and Depayloaders
+
+-   rtpisacpay, rtpisacdepay: new RTP payloader and depayloader for iSAC
+    audio codec
+
+-   rtph264depay:
+
+    -   new "request-keyframe" property to make the depayloader
+        automatically request a new keyframe from the sender on packet
+        loss, consistent with the new property on rtpvp8depay.
+    -   new "wait-for-keyframe" property to make depayloader wait for a
+        new keyframe at the beginning and after packet loss (only
+        effective if the depayloader outputs AUs), consistent with the
+        existing property on rtpvp8depay.
+
+-   rtpopuspay, rtpopusdepay: support libwebrtc-compatible multichannel
+    audio in addition to the previously supported multichannel audio
+    modes
+
+-   rtpopuspay: add DTX (Discontinuous Transmission) support
+
+-   rtpvp8depay: new "request-keyframe" property to make the depayloader
+    automatically request a new keyframe from the sender on packet loss.
+
+-   rtpvp8pay: temporal scaling support
 
--   this section will be filled in in due course
+-   rtpvp9depay: Improved SVC handling (aggregate all layers)
+
+RTP Infrastructure
+
+-   rtpst2022-1-fecdec, rtpst2022-1-fecenc: new elements providing SMPTE
+    2022-1 2-D Forward Error Correction. More details in Mathieu’s blog
+    post.
+
+-   rtpreddec: BUNDLE support
+
+-   rtpredenc, rtpulpfecenc: add support for Transport-wide Congestion
+    Control (TWCC)
+
+-   rtpsession: new "twcc-feedback-interval" property to allow RTCP TWCC
+    reports to be scheduled on a timer instead of per marker-bit.
+
+Plugin and library moves
 
 -   There were no plugin moves or library moves in this cycle.
 
@@ -56,8 +998,6 @@ Plugin removals
 
 The following elements or plugins have been removed:
 
--   this section will be filled in in due course
-
 -   The ofa audio fingerprinting plugin has been removed. The MusicIP
     database has been defunct for years so this plugin is likely neither
     useful nor used by anyone.
@@ -71,135 +1011,575 @@ The following elements or plugins have been removed:
 
 Miscellaneous API additions
 
--   this section will be filled in in due course
+Core
+
+-   gst_buffer_new_memdup() is a convenience function for the
+    widely-used gst_buffer_new_wrapped(g_memdup(data,size),size)
+    pattern.
+
+-   gst_caps_features_new_single() creates a new single GstCapsFeatures,
+    avoiding the need to use the vararg function with NULL terminator
+    for simple cases.
+
+-   gst_element_type_set_skip_documentation() can be used by plugins to
+    signal that certain elements should not be included in the GStreamer
+    plugin documentation. This is useful for plugins where elements are
+    registered dynamically based on hardware capabilities and/or where
+    the available plugins and properties vary from system to system.
+    This is used in the d3d11 plugin for example to ensure that only the
+    list of default elements is advertised in the documentation.
+
+-   gst_type_find_suggest_empty_simple() is a new convenience function
+    for typefinders for cases where there’s only a media type and no
+    other fields.
+
+-   New API to create elements and set properties at construction time,
+    which is not only convenient, but also allows GStreamer elements to
+    have construct-only properties: gst_element_factory_make_full(),
+    gst_element_factory_make_valist(),
+    gst_element_factory_make_with_properties(),
+    gst_element_factory_create_full(),
+    gst_element_factory_create_valist(),
+    gst_element_factory_create_with_properties().
+
+-   GstSharedTaskPool: new “shared” task pool subclass with slightly
+    different default behaviour than the existing GstTaskPool which
+    would create unlimited number of threads for new tasks. The shared
+    taskpool creates up to N threads (default: 1) and then distributes
+    pending tasks to those threads round-robin style, and blocks if no
+    thread is available. It is possible to join tasks. This can be used
+    by plugins to implement simple multi-threaded processing and is used
+    for the new multi-threaded video conversion and compositing done in
+    GstVideoAggregator, videoconverter and compositor.
+
+Plugins Base Utils library
+
+-   GstDiscoverer:
+
+    -   gst_discoverer_container_info_get_tags() was added to retrieve
+        global/container tags (vs. per-stream tags). Per-Stream tags can
+        be retrieved via the existing
+        gst_discoverer_stream_info_get_tags().
+        gst_discoverer_info_get_tags(), which for many files returns a
+        confusing mix of stream and container tags, has been deprecated
+        in favour of the container/stream-specific functions.
+    -   gst_discoverer_stream_info_get_stream_number() returns a unique
+        integer identifier for a given stream within the given
+        GstDiscoverer context. (If this matches the stream number inside
+        the container bitstream that’s by coincidence and not by
+        design.)
+
+-   gst_pb_utils_get_caps_description_flags() can be used to query
+    whether certain caps represent a container, audio, video, image,
+    subtitles, tags, or something else. This only works for formats
+    known to GStreamer.
+
+-   gst_pb_utils_get_file_extension_from_caps() returns a possible file
+    extension for given caps.
+
+-   gst_codec_utils_h264_get_profile_flags_level(): Parses profile,
+    flags, and level from H264 AvcC codec_data. The format of H264 AVCC
+    extradata/sequence_header is documented in the ITU-T H.264
+    specification section 7.3.2.1.1 as well as in ISO/IEC 14496-15
+    section 5.3.3.1.2.
+
+-   gst_codec_utils_caps_get_mime_codec() to convert caps to a RFC 6381
+    compatible MIME codec string codec. Useful for providing the codecs
+    field inside the Content-Type HTTP header for containerized formats,
+    such as mp4 or matroska.
+
+GStreamer OpenGL integration library and plugins
+
+-   glcolorconvert: added suppport for converting the video formats
+    A420, AV12, BGR, BGRA, RGBP and BGRP.
+
+-   Added support to GstGLBuffer for persistent buffer mappings where a
+    Pixel Buffer Object (PBO) can be mapped by both the CPU and the GPU.
+    This removes a memcpy() when uploading textures or vertices
+    particularly when software decoders (e.g. libav) are direct
+    rendering into our memory. Improves transfer performance
+    significantly. Requires OpenGL 4.4, GL_ARB_buffer_storage or
+    GL_EXT_buffer_storage
+
+-   Added various helper functions for handling 4x4 matrices of affine
+    transformations as used by GstVideoAffineTransformationMeta.
+
+-   Add support to GstGLContext for allowing the application to control
+    the config (EGLConfig, GLXConfig, etc) used when creating the OpenGL
+    context. This allows the ability to choose between RGB16 or RGB10A2
+    or RGBA8 back/front buffer configurations that were previously
+    hardcoded. GstGLContext also supports retrieving the configuration
+    it was created with or from an externally provide OpenGL context
+    handle. This infrastructure is also used to create a compatible
+    config from an application/externally provided OpenGL context in
+    order to improve compatibility with other OpenGL frameworks and GUI
+    toolkits. A new environment variable GST_GL_CONFIG was also added to
+    be able to request a specific configuration from the command line.
+    Note: different platforms will have different functionality
+    available.
+
+-   Add support for choosing between EGL and WGL at runtime when running
+    on Windows. Previously this was a build-time switch. Allows use in
+    e.g. Gtk applications on Windows that target EGL/ANGLE without
+    recompiling GStreamer. gst_gl_display_new_with_type() can be used by
+    applications to choose a specific display type to use.
+
+-   Build fixes to explicitly check for Broadcom-specific libraries on
+    older versions of the Raspberry Pi platform. The Broadcom OpenGL ES
+    and EGL libraries have different filenames. Using the vc4 Mesa
+    driver on the Raspberry Pi is not affected.
+
+-   Added support to glupload and gldownload for transferring RGBA
+    buffers using the memory:NVMM available on the Nvidia Tegra family
+    of embedded devices.
+
+-   Added support for choosing libOpenGL and libGLX as used in a GLVND
+    environment on unix-based platforms. This allows using desktop
+    OpenGL and EGL without pulling in any GLX symbols as would be
+    required with libGL.
+
+Video library
+
+-   New raw video formats:
+
+    -   AV12 (NV12 with alpha plane)
+    -   RGBP and BGRP (planar RGB formats)
+    -   ARGB64 variants with specified endianness instead of host
+        endianness:
+        -   ARGB64_LE, ARGB64_BE
+        -   RGBA64_BE, RGBA64_LE
+        -   BGRA64_BE, BGRA64_LE
+        -   ABGR64_BE, ABGR64_LE
+
+-   gst_video_orientation_from_tag() is new convenience API to parse the
+    image orientation from a GstTagList.
+
+-   GstVideoDecoder subframe support (see below)
+
+-   GstVideoCodecState now also carries some HDR metadata
+
+-   Ancillary video data: implement transform functions for AFD/Bar
+    metas, so they will be forwarded in more cases
+
+MPEG-TS library
+
+This library only handles section parsing and such, see above for
+changes to the actual mpegtsmux and mpegtsdemux elements.
+
+-   many additions and improvements to SCTE-35 section parsing
+-   new API for fetching extended descriptors:
+    gst_mpegts_find_descriptor_with_extension()
+-   add support for SIT sections (Selection Information Tables)
+-   expose event-from-section constructor gst_event_new_mpegts_section()
+-   parse Audio Preselection Descriptor needed for Dolby AC-4
+
+GstWebRTC library + webrtcbin
+
+-   Change the way in which sink pads and transceivers are matched
+    together to support easier usage. If a pad is created without a
+    specific index (i.e. using sink_%u as the pad template), then an
+    available compatible transceiver will be searched for. If a specific
+    index is requested (i.e. sink_1) then if a transceiver for that
+    m-line already exists, that transceiver must match the new sink pad
+    request. If there is no transceiver available in either scenario, a
+    new transceiver is created. If a mixture of both sink_1 and sink_%u
+    requests result in an impossible situation, an error will be
+    produced at pad request time or from create offer/answer.
+
+-   webrtcbin now uses regular ICE nomination instead of libnice’s
+    default of aggressive ICE nomination. Regular ICE nomination is the
+    default recommended by various relevant standards and improves
+    connectivity in specific network scenarios.
+
+-   Add support for limiting the port range used for RTP with the
+    addition of the min-rtp-port and max-rtp-port properties on the ICE
+    object.
+
+-   Expose the SCTP transport as a property on webrtcbin to more closely
+    match the WebRTC specification.
+
+-   Added support for taking into account the data channel transport
+    state when determining the value of the "connection-state" property.
+    Previous versions of the WebRTC spec did not include the data
+    channel state when computing this value.
+
+-   Add configuration for choosing the size of the underlying sockets
+    used for transporting media data
+
+-   Always advertise support for the transport-cc RTCP feedback protocol
+    as rtpbin supports it. For full support, the configured caps (input
+    or through codec-preferences) need to include the relevant RTP
+    header extension.
+
+-   Numerous fixes to caps and media handling to fail-fast when an
+    incompatible situation is detected.
+
+-   Improved support for attaching the required media after a remote
+    offer has been set.
+
+-   Add support for dynamically changing the amount of FEC used for a
+    particular stream.
+
+-   webrtcbin now stops further SDP processing at the first error it
+    encounters.
+
+-   Completed support for either local or the remote closing a data
+    channel.
+
+-   Various fixes when performing BUNDLEing of the media streams in
+    relation to RTX and FEC usage.
+
+-   Add support for writing out QoS DSCP marking on outgoing packets to
+    improve reliability in some network scenarios.
+
+-   Improvements to the statistics returned by the get-stats signal
+    including the addition of the raw statistics from the internal
+    RTPSource, the TWCC stats when available.
+
+-   The webrtc library does not expose any objects anymore with public
+    fields. Instead properties have been added to replace that
+    functionality. If you are accessing such fields in your application,
+    switch to the corresponding properties.
+
+GstCodecs and Video Parsers
+
+-   Support for render delays to improve throughput across all CODECs
+    (used with NVDEC and V4L2).
+-   lots of improvements to parsers and the codec parsing decoder base
+    classes (H264, H265, VP8, VP9, AV1, MPEG-2) used for various
+    hardware-accelerate decoder APIs.
+
+Bindings support
+
+-   gst_allocation_params_new() allocates a GstAllocationParams struct
+    on the heap. This should only be used by bindings (and freed via
+    gst_allocation_params_free() then). In C code you would allocate
+    this on the stack and only init it in place.
+
+-   gst_debug_log_literal() can be used to log a string to the debug log
+    without going through any printf format expansion and associated
+    overhead. This is mostly useful for bindings such as the Rust
+    bindings which may have done their own formatting already .
+
+-   Provide non-inlined versions of refcounting APIs for various
+    GStreamer mini objects, so that they can be consumed by bindings
+    (e.g. gstreamer-sharp): gst_buffer_ref, gst_buffer_unref,
+    gst_clear_buffer, gst_buffer_copy, gst_buffer_replace,
+    gst_buffer_list_ref, gst_buffer_list_unref, gst_clear_buffer_list,
+    gst_buffer_list_copy, gst_buffer_list_replace, gst_buffer_list_take,
+    gst_caps_ref, gst_caps_unref, gst_clear_caps, gst_caps_replace,
+    gst_caps_take, gst_context_ref, gst_context_unref, gst_context_copy,
+    gst_context_replace, gst_event_replace, gst_event_steal,
+    gst_event_take, gst_event_ref, gst_event_unref, gst_clear_event,
+    gst_event_copy, gst_memory_ref, gst_memory_unref, gst_message_ref,
+    gst_message_unref, gst_clear_message, gst_message_copy,
+    gst_message_replace, gst_message_take, gst_promise_ref,
+    gst_promise_unref, gst_query_ref, gst_query_unref, gst_clear_query,
+    gst_query_copy, gst_query_replace, gst_query_take, gst_sample_ref,
+    gst_sample_unref, gst_sample_copy, gst_tag_list_ref,
+    gst_tag_list_unref, gst_clear_tag_list, gst_tag_list_replace,
+    gst_tag_list_take, gst_uri_copy, gst_uri_ref, gst_uri_unref,
+    gst_clear_uri.
+
+-   expose a GType for GstMiniObject
+
+-   gst_device_provider_probe() now returns non-floating device object
+
+API Deprecations
+
+-   gst_element_get_request_pad() has been deprecated in favour of the
+    newly-added gst_element_request_pad_simple() which does the exact
+    same thing but has a less confusing name that hopefully makes clear
+    that the function request a new pad rather than just retrieves an
+    already-existing request pad.
+
+-   gst_discoverer_info_get_tags(), which for many files returns a
+    confusing mix of stream and container tags, has been deprecated in
+    favour of the container-specific and stream-specific functions,
+    gst_discoverer_container_info_get_tags() and
+    gst_discoverer_stream_info_get_tags().
+
+-   gst_video_sink_center_rect() was deprecated in favour of the more
+    generic newly-added gst_video_center_rect().
+
+-   The GST_MEMORY_FLAG_NO_SHARE flag has been deprecated, as it tends
+    to cause problems and prevents sub-buffering. If pooling or lifetime
+    tracking is required, memories should be allocated through a custom
+    GstAllocator instead of relying on the lifetime of the buffers the
+    memories were originally attached to, which is fragile anyway.
+
+-   The GstPlayer high-level playback library is being replaced with the
+    new GstPlay library (see above). GstPlayer should be considered
+    deprecated at this point and will be marked as such in the next
+    development cycle. Applications should be ported to GstPlay.
+
+-   Gstreamer Editing Services: ges_video_transition_set_border(),
+    ges_video_transition_get_border()
+    ges_video_transition_set_inverted()
+    ges_video_transition_is_inverted() have been deprecated, use
+    ges_timeline_element_set_children_properties() instead.
 
 Miscellaneous performance, latency and memory optimisations
 
--   this section will be filled in in due course
+More video conversion fast paths
 
-Miscellaneous other changes and enhancements
+-   v210 ↔ I420, YV12, Y42B, UYVY and YUY2
+-   A420 → RGB
+
+Less jitter when waiting on the system clock
+
+-   Better system clock wait accuracy, less jitter: where available,
+    clock_nanosleep is used for higher accuracy for waits below 500
+    usecs, and waits below 2ms will first use the regular waiting system
+    and then clock_nanosleep for the remainder. The various wait
+    implementation have a latency ranging from 50 to 500+ microseconds.
+    While this is not a major issue when dealing with a low number of
+    waits per second (for ex: video), it does introduce a non-negligible
+    jitter for synchronisation of higher packet rate systems.
+
+Video decoder subframe support
 
--   this section will be filled in in due course
+-   The GstVideoDecoder base class gained API to process input at the
+    sub-frame level. That way video decoders can start decoding slices
+    before they have received the full input frame in its entirety (to
+    the extent this is supported by the codec, of course). This helps
+    with CPU utilisation and reduces latency.
 
-Tracing framework and debugging improvements
+-   This functionality is now being used in the OpenJPEG JPEG 2000
+    decoder, the FFmpeg H.264 decoder (in case of NAL-aligned input) and
+    the OpenMAX H.264/H.265 decoders (in case of NAL-aligned input).
 
--   this section will be filled in in due course
+Miscellaneous other changes and enhancements
+
+-   GstDeviceMonitor no longer fails to start just because one of the
+    device providers failed to start. That could happen for example on
+    systems where the pulseaudio device provider is installed, but
+    pulseaudio isn’t actually running but ALSA is used for audio
+    instead. In the same vein the device monitor now keeps track of
+    which providers have been started (via the new
+    gst_device_provider_is_started()) and only stops actually running
+    device providers when stopping the device monitor.
+
+-   On embedded systems it can be useful to create a registry that can
+    be shared and read by multiple processes running as different users.
+    It is now possible to set the new GST_REGISTRY_MODE environment
+    variable to specify the file mode for the registry file, which by
+    default is set to be only user readable/writable.
+
+-   GstNetClientClock will signal lost sync in case the remote time
+    resets (e.g. because device power cycles), by emitting the “synced”
+    signal with synced=FALSE parameter, so applications can take action.
+
+-   gst_value_deserialize_with_pspec() allows deserialization with a
+    hint for what the target GType should be. This allows for example
+    passing arrays of flags through the command line or
+    gst_util_set_object_arg(), eg: foo="<bar,bar+baz>".
+
+-   It’s now allowed to create an empty GstVideoOverlayComposition
+    without any rectangles by passing a NULL rectangle to
+    gst_video_overlay_composition_new(). This is useful for bindings and
+    simplifies application code in some places.
+
+Tracing framework, debugging and testing improvements
+
+-   New factories tracer to list loaded elements (and other plugin
+    features). This can be useful to collect a list of elements needed
+    for an application, which then in turn can be used to create a
+    tailored minimal GStreamer build that contains just the elements
+    needed and nothing else.
+-   New plugin-feature-loaded tracing hook for use by tracers like the
+    new factories tracer
+
+-   GstHarness: Add gst_harness_set_live() so that harnesses can be set
+    to non-live and return is-live=false in latency queries if needed.
+    Default behaviour is to always return is-live=true in latency
+    queries.
+
+-   navseek: new "hold-eos" property. When enabled, the element will
+    hold back an EOS event until the next keystroke (via navigation
+    events). This can be used to keep a video sink showing the last
+    frame of a video pipeline until a key is pressed instead of tearing
+    it down immediately on EOS.
+
+-   New fakeaudiosink element: mimics an audio sink and can be used for
+    testing and CI pipelines on systems where no audio system is
+    installed or running. It differs from fakesink in that it only
+    support audio caps and syncs to the clock by default like a normal
+    audio sink. It also implements the GstStreamVolume interface like
+    most audio sinks do.
+
+-   New videocodectestsink element for video codec conformance testing:
+    Calculates MD5 checksums for video frames and skips any padding
+    whilst doing so. Can optionally also write back the video data with
+    padding removed into a file for easy byte-by-byte comparison with
+    reference data.
 
 Tools
 
--   this section will be filled in in due course
+gst-inspect-1.0
+
+-   Can sort the list of plugins by passing --sort=name as command line
+    option
+
+gst-launch-1.0
+
+-   will now error out on top-level properties that don’t exist and
+    which were silently ignored before
+-   On Windows the high-resolution clock is enabled now, which provides
+    better clock and timer performance on Windows (see Windows section
+    below for more details).
+
+gst-play-1.0
+
+-   New --start-position command line argument to start playback from
+    the specified position
+-   Audio can be muted/unmuted in interactive mode by pressing the m
+    key.
+-   On Windows the high-resolution clock is enabled now (see Windows
+    section below for more details)
+
+gst-device-monitor-1.0
+
+-   New --include-hidden command line argument to also show “hidden”
+    device providers
+
+ges-launch-1.0
+
+-   New interactive mode that allows seeking and such. Can be disabled
+    by passing the --no-interactive argument on the command line.
+-   Option to forward tags
+-   Allow using an existing clip to determine the rendering format (both
+    topology and profile) via new --profile-from command line argument.
 
 GStreamer RTSP server
 
--   this section will be filled in in due course
+-   GstRTSPMediaFactory gained API to disable RTCP
+    (gst_rtsp_media_factory_set_enable_rtcp(), "enable-rtcp" property).
+    Previously RTCP was always allowed for all RTSP medias. With this
+    change it is possible to disable RTCP completely, no matter if the
+    client wants to do RTCP or not.
+
+-   Make a mount point of / work correctly. While not allowed by the
+    RTSP 2 spec, the RTSP 1 spec is silent on this and it is used in the
+    wild. It is now possible to use / as a mount path in
+    gst-rtsp-server, e.g. rtsp://example.com/ would work with this now.
+    Note that query/fragment parts of the URI are not necessarily
+    correctly handled, and behaviour will differ between various
+    client/server implementations; so use it if you must but don’t bug
+    us if it doesn’t work with third party clients as you’d hoped.
+
+-   multithreading fixes (races, refcounting issues, deadlocks)
+
+-   ONVIF audio backchannel fixes
+
+-   ONVIF trick mode optimisations
+
+-   rtspclientsink: new "update-sdp" signal that allows updating the SDP
+    before sending it to the server via ANNOUNCE. This can be used to
+    add additional metadata to the SDP, for example. The order and
+    number of medias must not be changed, however.
 
 GStreamer VAAPI
 
--   this section will be filled in in due course
+-   new AV1 decoder element (vaapiav1dec)
+
+-   H264 decoder: handle stereoscopic 3D video with frame packing
+    arrangement SEI messages
+
+-   H265 encoder: added Screen Content Coding extensions support
+
+-   H265 decoder: gained MAIN_444_12 profile support (decoded to
+    Y412_LE), and 4:2:2 12-bits support (decoded to Y212_LE)
+
+-   vaapipostproc: gained BT2020 color standard support
+
+-   vaapidecode: now generates caps templates dynamically at runtime in
+    order to advertise actually supported caps instead of all
+    theoretically supported caps.
+
+-   GST_VAAPI_DRM_DEVICE environment variable to force a specified DRM
+    device when a DRM display is used. It is ignored when other types of
+    displays are used. By default /dev/dri/renderD128 is used for DRM
+    display.
 
 GStreamer OMX
 
--   this section will be filled in in due course
+-   subframe support in H.264/H.265 decoders
 
 GStreamer Editing Services and NLE
 
--   this section will be filled in in due course
+-   framepositioner: new "operator" property to access blending modes in
+    the compositor
+-   timeline: Implement snapping to markers
+-   smart-mixer: Add support for d3d11compositor and glvideomixer
+-   titleclip: add "draw-shadow" child property
+-   ges:// URI support to define a timeline from a description.
+-   command-line-formatter
+    -   Add track management to timeline description
+    -   Add keyframe support
+-   ges-launch-1.0:
+    -   Add an interactive mode where we can seek etc…
+    -   Add option to forward tags
+    -   Allow using an existing clip to determine the rendering format
+        (both topology and profile) via new --profile-from command line
+        argument.
+-   Fix static build
 
 GStreamer validate
 
--   this section will be filled in in due course
+-   report: Add a way to force backtraces on reports even if not a
+    critical issue (GST_VALIDATE_ISSUE_FLAGS_FORCE_BACKTRACE)
+-   Add a flag to gst_validate_replace_variables_in_string() allow
+    defining how to resolve variables in structs
+-   Add gst_validate_bin_monitor_get_scenario() to get the bin monitor
+    scenario, which is useful for applications that use Validate
+    directly.
+-   Add an expected-values parameter to wait, message-type=XX allowing
+    more precise filtering of the message we are waiting for.
+-   Add config file support: each test can now use a config file for the
+    given media file used to test.
+-   Add support to check properties of object properties
+-   scenario: Add an "action-done" signal to signal when an action is
+    done
+-   scenario: Add a "run-command" action type
+-   scenario: Allow forcing running action on idle from scenario file
+-   scenario: Allow iterating over arrays in foreach
+-   scenario: Rename ‘interlaced’ action to ‘non-blocking’
+-   scenario: Add a non-blocking flag to the wait signal
 
 GStreamer Python Bindings
 
--   this section will be filled in in due course
+-   Fixes for Python 3.10
+-   Various build fixes
+-   at least one known breaking change caused by g-i annotation changes
+    (see below)
 
 GStreamer C# Bindings
 
--   this section will be filled in in due course
+-   Fix GstDebugGraphDetails enum
+-   Updated to latests GtkSharp
+-   Updated to include GStreamer 1.20 API
 
 GStreamer Rust Bindings and Rust Plugins
 
-The GStreamer Rust bindings are released separately with a different
-release cadence that’s tied to gtk-rs, but the latest release has
-already been updated for the upcoming new GStreamer 1.20 API.
-
-gst-plugins-rs, the module containing GStreamer plugins written in Rust,
-has also seen lots of activity with many new elements and plugins.
-
-What follows is a list of elements and plugins available in
-gst-plugins-rs, so people don’t miss out on all those potentially useful
-elements that have no C equivalent.
-
--   FIXME: add new elements
-
-Rust audio plugins
-
--   audiornnoise: New element for audio denoising which implements the
-    noise removal algorithm of the Xiph RNNoise library, in Rust
--   rsaudioecho: Port of the audioecho element from gst-plugins-good
-    rsaudioloudnorm: Live audio loudness normalization element based on
-    the FFmpeg af_loudnorm filter
--   claxondec: FLAC lossless audio codec decoder element based on the
-    pure-Rust claxon implementation
--   csoundfilter: Audio filter that can use any filter defined via the
-    Csound audio programming language
--   lewtondec: Vorbis audio decoder element based on the pure-Rust
-    lewton implementation
-
-Rust video plugins
-
--   cdgdec/cdgparse: Decoder and parser for the CD+G video codec based
-    on a pure-Rust CD+G implementation, used for example by karaoke CDs
--   cea608overlay: CEA-608 Closed Captions overlay element
--   cea608tott: CEA-608 Closed Captions to timed-text (e.g. VTT or SRT
-    subtitles) converter
--   tttocea608: CEA-608 Closed Captions from timed-text converter
--   mccenc/mccparse: MacCaption Closed Caption format encoder and parser
--   sccenc/sccparse: Scenarist Closed Caption format encoder and parser
--   dav1dec: AV1 video decoder based on the dav1d decoder implementation
-    by the VLC project
--   rav1enc: AV1 video encoder based on the fast and pure-Rust rav1e
-    encoder implementation
--   rsflvdemux: Alternative to the flvdemux FLV demuxer element from
-    gst-plugins-good, not feature-equivalent yet
--   rsgifenc/rspngenc: GIF/PNG encoder elements based on the pure-Rust
-    implementations by the image-rs project
-
-Rust text plugins
-
--   textwrap: Element for line-wrapping timed text (e.g. subtitles) for
-    better screen-fitting, including hyphenation support for some
-    languages
-
-Rust network plugins
-
--   reqwesthttpsrc: HTTP(S) source element based on the Rust
-    reqwest/hyper HTTP implementations and almost feature-equivalent
-    with the main GStreamer HTTP source souphttpsrc
--   s3src/s3sink: Source/sink element for the Amazon S3 cloud storage
--   awstranscriber: Live audio to timed text transcription element using
-    the Amazon AWS Transcribe API
-
-Generic Rust plugins
-
--   sodiumencrypter/sodiumdecrypter: Encryption/decryption element based
-    on libsodium/NaCl
--   togglerecord: Recording element that allows to pause/resume
-    recordings easily and considers keyframe boundaries
--   fallbackswitch/fallbacksrc: Elements for handling potentially
-    failing (network) sources, restarting them on errors/timeout and
-    showing a fallback stream instead
--   threadshare: Set of elements that provide alternatives for various
-    existing GStreamer elements but allow to share the streaming threads
-    between each other to reduce the number of threads
--   rsfilesrc/rsfilesink: File source/sink elements as replacements for
-    the existing filesrc/filesink elements
+-   The GStreamer Rust bindings are released separately with a different
+    release cadence that’s tied to gtk-rs, but the latest release has
+    already been updated for the upcoming new GStreamer 1.20 API (v1_20
+    feature).
+
+-   gst-plugins-rs, the module containing GStreamer plugins written in
+    Rust, has also seen lots of activity with many new elements and
+    plugins. See the New Elements section above for a list of new Rust
+    elements.
 
 Build and Dependencies
 
 -   Meson 0.59 or newer is required to build GStreamer now.
 
--   FIXME: this section will be filled in in due course
+-   The GLib requirement has been bumped to GLib 2.56 or newer (from
+    March 2018).
+
+-   The wpe plugin now requires wpe >= 2.28 and wpebackend-fdo >= 1.8
 
 Explicit opt-in required for build of certain plugins with (A)GPL dependencies
 
@@ -211,11 +1591,9 @@ even if the required dependencies are available.
 See Building plugins with (A)GPL-licensed dependencies for more details
 and a non-exhaustive list of plugins affected.
 
-gst-build: replaced by Monorepo
+gst-build: replaced by mono repository
 
--   this section will be filled in in due course
-
--   FIXME: describe + link to Monorepo FAQ
+See mono repository section above and the GStreamer mono repository FAQ.
 
 Cerbero
 
@@ -223,76 +1601,254 @@ Cerbero is a meta build system used to build GStreamer plus dependencies
 on platforms where dependencies are not readily available, such as
 Windows, Android, iOS and macOS.
 
-General improvements
+General Cerbero improvements
 
--   this section will be filled in in due course
+-   Plugin removed: libvisual
+-   New plugins: rtpmanagerbad and rist
 
-macOS / iOS
+macOS / iOS specific Cerbero improvements
 
--   this section will be filled in in due course
+-   XCode 12 support
+-   macOS OS release support is now future-proof, similar to iOS
+-   macOS Apple Silicon (ARM64) cross-compile support has been added
+-   macOS Apple Silicon (ARM64) native support is currently experimental
 
-Windows
+Windows specific Cerbero improvements
 
--   this section will be filled in in due course
+-   Visual Studio 2022 support has been added
+-   bootstrap is faster since it requires building fewer build-tools
+    recipes on Windows
+-   package is faster due to better scheduling of recipe stages and
+    elimination of unnecessary autotools regeneration
+-   The following plugins are no longer built on Windows:
+    -   a52dec (another decoder is still available in libav)
+    -   dvdread
+    -   resindvd
 
 Windows MSI installer
 
--   this section will be filled in in due course
+-   no major changes
 
-Linux
+Linux specific Cerbero improvements
 
--   this section will be filled in in due course
+-   Fedora, Debian OS release support is now more future-proof
+-   Amazon Linux 2 support has been added
 
-Android
+Android specific Cerbero improvements
 
--   this section will be filled in in due course
+-   no major changes
 
 Platform-specific changes and improvements
 
 Android
 
--   this section will be filled in in due course
+-   No major changes
 
 macOS and iOS
 
--   this section will be filled in in due course
+-   applemedia: add ProRes support to vtenc and vtdec
 
 Windows
 
--   this section will be filled in in due course
+-   On Windows the high-resolution clock is enabled now in the
+    gst-launch-1.0 and gst-play-1.0 command line tools, which provides
+    better clock and timer performance on Windows, at the cost of higher
+    power consumption. By default, without the high-resolution clock
+    enabled, the timer precision on Windows is system-dependent and may
+    be as bad as 15ms which is not good enough for many multimedia
+    applications. Developers may want to do the same in their Windows
+    applications if they think it’s a good idea for their application
+    use case, and depending on the Windows version they target. This is
+    not done automatically by GStreamer because on older Windows
+    versions (pre-Windows 10) this affects a global Windows setting and
+    also there’s a power consumption vs. performance trade-off that may
+    differ from application to application.
+
+-   dxgiscreencapsrc now supports resolution changes
+
+-   The wasapi2 audio plugin was rewritten and now has a higher rank
+    than the old wasapi plugin since it has a number of additional
+    features such as automatic stream routing, and no
+    known-but-hard-to-fix issues. The plugin is always built if the
+    Windows 10 SDK is available now.
+
+-   The wasapi device providers now detect and notify dynamic device
+    additions/removals
+
+-   d3d11screencapturesrc: new desktop capture element, including
+    GstDeviceProvider implementation to enumerate/select target monitors
+    for capture.
+
+-   Direct3D11/DXVA decoder now supports AV1 and MPEG2 codecs
+    (d3d11av1dec, d3d11mpeg2dec)
+
+-   VP9 decoding got more reliable and stable thanks to a newly written
+    codec parser
+
+-   Support for decoding interlaced H.264/AVC streams
+
+-   Hardware-accelerated video deinterlacing (d3d11deinterlace) and
+    video mixing (d3d11compositor)
+
+-   Video mixing with the Direct3D11 API (d3d11compositor)
+
+-   MediaFoundation API based hardware encoders gained the ability to
+    receive Direct3D11 textures as an input
+
+-   Seungha’s blog post “GStreamer ❤ Windows: A primer on the cool stuff
+    you’ll find in the 1.20 release” describes many of the
+    Windows-related improvements in more detail
 
 Linux
 
--   this section will be filled in in due course
+-   bluez: LDAC Bluetooth audio codec support in a2dpsink and avdtpsink,
+    as well as an LDAC RTP payloader (rtpldacpay) and an LDAC audio
+    encoder (ldacenc)
+
+-   kmssink: gained support for NV24, NV61, RGB16/BGR16 formats;
+    auto-detect NVIDIA Tegra driver
 
 Documentation improvements
 
--   this section will be filled in in due course
+-   hardware-accelerated GPU plugins will now no longer always list all
+    the element variants for all available GPUs, since those are
+    system-dependent and it’s confusing for users to see those in the
+    documentation just because the GStreamer developer who generated the
+    docs had multiple GPUs to play with at the time. Instead just show
+    the default elements.
 
-Possibly Breaking Changes
+Possibly Breaking and Other Noteworthy Behavioural Changes
+
+-   gst_parse_launch(), gst_parse_bin_from_description() and friends
+    will now error out when setting properties that don’t exist on
+    top-level bins. They were silently ignored before.
+
+-   The GstWebRTC library does not expose any objects anymore with
+    public fields. Instead properties have been added to replace that
+    functionality. If you are accessing such fields in your application,
+    switch to the corresponding properties.
 
--   FIXME: this section will be filled in in due course
--   MPEG-TS SCTE-35 API changes (FIXME: flesh out)
--   gst_parse_launch() and friends now error out on non-existing
-    properties on top-level bins where they would silently fail and
-    ignore those before.
 -   playbin and uridecodebin now emit the source-setup signal before the
     element is added to the bin and linked so that the source element is
     already configured before any scheduling query comes in, which is
-    useful for elements such as appsrc or giostreamsrc. (Merge Request)
+    useful for elements such as appsrc or giostreamsrc.
+
+-   The source element inside urisourcebin (used inside uridecodebin3
+    which is used inside playbin3) is no longer called "source". This
+    shouldn’t affect anyone hopefully, because there’s a "setup-source"
+    signal to configure the source element and no one should rely on
+    names of internal elements anyway.
+
+-   The vp8enc element now expects bps (bits per second) for the
+    "temporal-scalability-target-bitrate" property, which is consistent
+    with the "target-bitrate" property. Since additional configuration
+    is required with modern libvpx to make temporal scaling work anyway,
+    chances are that very few people will have been using this property
+
+-   vp8enc and vp9enc now default to “good quality” for the "deadline"
+    property rather then “best quality”. Having the deadline set to best
+    quality causes the encoder to be absurdly slow, most real-life users
+    will want the good quality tradeoff instead.
+
+-   The experimental GstTranscoder library API in gst-plugins-bad was
+    changed from a GObject signal-based notification mechanism to a
+    GstBus/message-based mechanism akin to GstPlayer/GstPlay.
+
+-   MPEG-TS SCTE-35 API: semantic change for SCTE-35 splice commands:
+    timestamps passed by the application should be in running time now,
+    since users of the API can’t really be expected to predict the local
+    PTS of the muxer.
+
+-   The GstContext used by souphttpsrc to share the session between
+    multiple element instances has changed. Previously it provided
+    direct access to the internal SoupSession object, now it only
+    provides access to an opaque, internal type. This change is
+    necessary because SoupSession is not thread-safe at all and can’t be
+    shared safely between arbitrary external code and souphttpsrc.
+
+-   Python bindings: GObject-introspection related Annotation fixes have
+    led to a case of a GstVideo.VideoInfo-related function signature
+    changing in the Python bindings (possibly one or two other cases
+    too). This is for a function that should never have been exposed in
+    the first place though, so the bindings are being updated to throw
+    an exception in that case, and the correct replacement API has been
+    added in form of an override.
 
 Known Issues
 
--   this section will be filled in in due course
-
--   There are a couple of known WebRTC-related regressions/blockers:
-
-    -   webrtc: DTLS setup with Chrome is broken
-    -   webrtcbin: First keyframe is usually lost
+-   nothing in particular at this point (but also see possibly breaking
+    changes section above)
 
 Contributors
 
--   this section will be filled in in due course
+Aaron Boxer, Adam Leppky, Adam Williamson, Alba Mendez, Alejandro
+González, Aleksandr Slobodeniuk, Alexander Vandenbulcke, Alex Ashley,
+Alicia Boya García, Andika Triwidada, Andoni Morales Alastruey, Andrew
+Wesie, Andrey Moiseev, Antonio Ospite, Antonio Rojas, Arthur Crippa
+Búrigo, Arun Raghavan, Ashley Brighthope, Axel Kellermann, Baek, Bastien
+Nocera, Bastien Reboulet, Benjamin Gaignard, Bing Song, Binh Truong,
+Biswapriyo Nath, Brad Hards, Brad Smith, Brady J. Garvin, Branko
+Subasic, Camilo Celis Guzman, Chris Bass, ChrisDuncanAnyvision, Chris
+White, Corentin Damman, Daniel Almeida, Daniel Knobe, Daniel Stone,
+david, David Fernandez, David Keijser, David Phung, Devarsh Thakkar,
+Dinesh Manajipet, Dmitry Samoylov, Dmitry Shusharin, Dominique Martinet,
+Doug Nazar, Ederson de Souza, Edward Hervey, Emmanuel Gil Peyrot,
+Enrique Ocaña González, Ezequiel Garcia, Fabian Orccon, Fabrice
+Fontaine, Fernando Jimenez Moreno, Florian Karydes, Francisco Javier
+Velázquez-García, François Laignel, Frederich Munch, Fredrik Pålsson,
+George Kiagiadakis, Georg Lippitsch, Göran Jönsson, Guido Günther,
+Guillaume Desmottes, Guiqin Zou, Haakon Sporsheim, Haelwenn (lanodan)
+Monnier, Haihao Xiang, Haihua Hu, Havard Graff, He Junyan, Helmut
+Januschka, Henry Wilkes, Hosang Lee, Hou Qi, Ignacio Casal Quinteiro,
+Igor Kovalenko, Ilya Kreymer, Imanol Fernandez, Jacek Tomaszewski, Jade
+Macho, Jakub Adam, Jakub Janků, Jan Alexander Steffens (heftig), Jan
+Schmidt, Jason Carrete, Jason Pereira, Jay Douglass, Jeongki Kim, Jérôme
+Laheurte, Jimmi Holst Christensen, Johan Sternerup, John Hassell, John
+Lindgren, John-Mark Bell, Jonathan Matthew, Jordan Petridis, Jose
+Quaresma, Julian Bouzas, Julien, Kai Uwe Broulik, Kasper Steensig
+Jensen, Kellermann Axel, Kevin Song, Khem Raj, Knut Inge Hvidsten, Knut
+Saastad, Kristofer Björkström, Lars Lundqvist, Lawrence Troup, Lim Siew
+Hoon, Lucas Stach, Ludvig Rappe, Luis Paulo Fernandes de Barros, Luke
+Yelavich, Mads Buvik Sandvei, Marc Leeman, Marco Felsch, Marek Vasut,
+Marian Cichy, Marijn Suijten, Marius Vlad, Markus Ebner, Mart Raudsepp,
+Matej Knopp, Mathieu Duponchelle, Matthew Waters, Matthieu De Beule,
+Mengkejiergeli Ba, Michael de Gans, Michael Olbrich, Michael Tretter,
+Michal Dzik, Miguel Paris, Mikhail Fludkov, mkba, Nazar Mokrynskyi,
+Nicholas Jackson, Nicola Murino, Nicolas Dufresne, Niklas Hambüchen,
+Nikolay Sivov, Nirbheek Chauhan, Olivier Blin, Olivier Crete, Olivier
+Crête, Paul Goulpié, Per Förlin, Peter Boba, P H, Philippe Normand,
+Philipp Zabel, Pieter Willem Jordaan, Piotrek Brzeziński, Rafał
+Dzięgiel, Rafostar, raghavendra, Raghavendra, Raju Babannavar, Raleigh
+Littles III, Randy Li, Randy Li (ayaka), Ratchanan Srirattanamet, Raul
+Tambre, reed.lawrence, Ricky Tang, Robert Rosengren, Robert Swain, Robin
+Burchell, Roman Sivriver, R S Nikhil Krishna, Ruben Gonzalez, Ruslan
+Khamidullin, Sanchayan Maity, Scott Moreau, Sebastian Dröge, Sergei
+Kovalev, Seungha Yang, Sid Sethupathi, sohwan.park, Sonny Piers, Staz M,
+Stefan Brüns, Stéphane Cerveau, Stephan Hesse, Stian Selnes, Stirling
+Westrup, Théo MAILLART, Thibault Saunier, Tim, Timo Wischer, Tim-Philipp
+Müller, Tim Schneider, Tobias Ronge, Tom Schoonjans, Tulio Beloqui,
+tyler-aicradle, U. Artie Eoff, Ung, Val Doroshchuk, VaL Doroshchuk,
+Víctor Manuel Jáquez Leal, Vivek R, Vivia Nikolaidou, Vivienne
+Watermeier, Vladimir Menshakov, Will Miller, Wim Taymans, Xabier
+Rodriguez Calvar, Xavier Claessens, Xℹ Ruoyao, Yacine Bandou, Yinhang
+Liu, youngh.lee, youngsoo.lee, yychao, Zebediah Figura, Zhang yuankun,
+Zhang Yuankun, Zhao, Zhao Zhili, , Aleksandar Topic, Antonio Ospite,
+Bastien Nocera, Benjamin Gaignard, Brad Hards, Carlos Falgueras García,
+Célestin Marot, Corentin Damman, Corentin Noël, Daniel Almeida, Daniel
+Knobe, Danny Smith, Dave Piché, Dmitry Osipenko, Fabrice Fontaine,
+fjmax, Florian Zwoch, Guillaume Desmottes, Haihua Hu, Heinrich Kruger,
+He Junyan, Jakub Adam, James Cowgill, Jan Alexander Steffens (heftig),
+Jean Felder, Jeongki Kim, Jiri Uncovsky, Joe Todd, Jordan Petridis,
+Krystian Wojtas, Marc-André Lureau, Marcin Kolny, Marc Leeman, Mark
+Nauwelaerts, Martin Reboredo, Mathieu Duponchelle, Matthew Waters,
+Mengkejiergeli Ba, Michael Gruner, Nicolas Dufresne, Nirbheek Chauhan,
+Olivier Crête, Philippe Normand, Rafał Dzięgiel, Ralf Sippl, Robert
+Mader, Sanchayan Maity, Sangchul Lee, Sebastian Dröge, Seungha Yang,
+Stéphane Cerveau, Teh Yule Kim, Thibault Saunier, Thomas Klausner, Timo
+Wischer, Tim-Philipp Müller, Tobias Reineke, Tomasz Andrzejak, Trung Do,
+Tyler Compton, Ung, Víctor Manuel Jáquez Leal, Vivia Nikolaidou, Wim
+Taymans, wngecn, Wonchul Lee, wuchang li, Xavier Claessens, Xi Ruoyao,
+Yoshiharu Hirose, Zhao,
 
 … and many others who have contributed bug reports, translations, sent
 suggestions or helped testing.
@@ -307,7 +1863,7 @@ the git 1.20 branch, which will be a stable branch.
 
 1.20.0
 
-1.20.0 is scheduled to be released around October/November 2021.
+1.20.0 is scheduled to be released around early February 2022.
 
 Schedule for 1.22
 
@@ -315,7 +1871,9 @@ Our next major feature release will be 1.22, and 1.21 will be the
 unstable development version leading up to the stable 1.22 release. The
 development of 1.21/1.22 will happen in the git main branch.
 
-The plan for the 1.22 development cycle is yet to be confirmed.
+The plan for the 1.22 development cycle is yet to be confirmed. Assuming
+no major project-wide reorganisations in the 1.22 cycle we might try and
+aim for a release around August 2022.
 
 1.22 will be backwards-compatible to the stable 1.20, 1.18, 1.16, 1.14,
 1.12, 1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series.
@@ -323,6 +1881,7 @@ The plan for the 1.22 development cycle is yet to be confirmed.
 ------------------------------------------------------------------------
 
 These release notes have been prepared by Tim-Philipp Müller with
-contributions from …
+contributions from Matthew Waters, Nicolas Dufresne, Nirbheek Chauhan,
+Sebastian Dröge and Seungha Yang.
 
 License: CC BY-SA 4.0
diff --git a/subprojects/gst-plugins-base/RELEASE b/subprojects/gst-plugins-base/RELEASE
index 30785f5f0e..2128ca5951 100644
--- a/subprojects/gst-plugins-base/RELEASE
+++ b/subprojects/gst-plugins-base/RELEASE
@@ -1,4 +1,4 @@
-This is GStreamer gst-plugins-base 1.19.3.
+This is GStreamer gst-plugins-base 1.19.90.
 
 GStreamer 1.19 is the development branch leading up to the next major
 stable version which will be 1.20.
diff --git a/subprojects/gst-plugins-base/gst-plugins-base.doap b/subprojects/gst-plugins-base/gst-plugins-base.doap
index 05255eef61..6fcb9be8ce 100644
--- a/subprojects/gst-plugins-base/gst-plugins-base.doap
+++ b/subprojects/gst-plugins-base/gst-plugins-base.doap
@@ -36,6 +36,16 @@ A wide range of video and audio decoders, encoders, and filters are included.
 
  <release>
   <Version>
+   <revision>1.19.90</revision>
+   <branch>main</branch>
+   <name></name>
+   <created>2022-01-28</created>
+   <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-plugins-base/gst-plugins-base-1.19.90.tar.xz" />
+  </Version>
+ </release>
+
+ <release>
+  <Version>
    <revision>1.19.3</revision>
    <branch>main</branch>
    <name></name>
diff --git a/subprojects/gst-plugins-base/meson.build b/subprojects/gst-plugins-base/meson.build
index 404c983f01..fc7746c4b7 100644
--- a/subprojects/gst-plugins-base/meson.build
+++ b/subprojects/gst-plugins-base/meson.build
@@ -1,5 +1,5 @@
 project('gst-plugins-base', 'c',
-  version : '1.19.3.1',
+  version : '1.19.90',
   meson_version : '>= 0.59',
   default_options : [ 'warning_level=1',
                       'buildtype=debugoptimized' ])
diff --git a/subprojects/gst-plugins-good/ChangeLog b/subprojects/gst-plugins-good/ChangeLog
index a9fc458b5d..cc23ba0f6d 100644
--- a/subprojects/gst-plugins-good/ChangeLog
+++ b/subprojects/gst-plugins-good/ChangeLog
@@ -1,3 +1,19 @@
+=== release 1.19.90 ===
+
+2022-01-28 14:28:35 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* NEWS:
+	* RELEASE:
+	* docs/gst_plugins_cache.json:
+	* gst-plugins-good.doap:
+	* meson.build:
+	  Release 1.19.90
+
+2022-01-28 14:28:28 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* ChangeLog:
+	  Update ChangeLogs for 1.19.90
+
 2022-01-28 15:47:44 +0200  Sebastian Dröge <sebastian@centricular.com>
 
 	* ext/soup/gstsouphttpclientsink.c:
diff --git a/subprojects/gst-plugins-good/NEWS b/subprojects/gst-plugins-good/NEWS
index e9f5227c15..1a512c1848 100644
--- a/subprojects/gst-plugins-good/NEWS
+++ b/subprojects/gst-plugins-good/NEWS
@@ -1,23 +1,19 @@
 GStreamer 1.20 Release Notes
 
 GStreamer 1.20 has not been released yet. It is scheduled for release in
-November 2021.
+late January / early February 2022.
 
 1.19.x is the unstable development version that is being developed in
-the git main branch and which will eventually result in 1.20, and 1.19.3
-is the current development release in that series
-
-It is expected that feature freeze will be in early October 2021,
-followed by one or two 1.19.9x pre-releases and the new 1.20 stable
-release around October/November 2021.
+the git main branch and which will eventually result in 1.20, and
+1.19.90 is the first release candidate in that series (1.20rc1).
 
 1.20 will be backwards-compatible to the stable 1.18, 1.16, 1.14, 1.12,
-1.10, 1.8, 1.6,, 1.4, 1.2 and 1.0 release series.
+1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series.
 
 See https://gstreamer.freedesktop.org/releases/1.20/ for the latest
 version of this document.
 
-Last updated: Monday 1 November 2021, 01:00 UTC (log)
+Last updated: Wednesday 26 January 2022, 01:00 UTC (log)
 
 Introduction
 
@@ -30,25 +26,971 @@ fixes and other improvements.
 
 Highlights
 
--   this section will be completed in due course
+-   Development in GitLab was switched to a single git repository
+    containing all the modules
+-   GstPlay: new high-level playback library, replaces GstPlayer
+-   WebM Alpha decoding support
+-   Encoding profiles can now be tweaked with additional
+    application-specified element properties
+-   Compositor: multi-threaded video conversion and mixing
+-   RTP header extensions: unified support in RTP depayloader and
+    payloader base classes
+-   SMPTE 2022-1 2-D Forward Error Correction support
+-   Smart encoding (passthrough) support for VP8, VP9, H.265 in
+    encodebin and transcodebin
+-   Runtime compatibility support for libsoup2 and libsoup3 (libsoup3
+    support experimental)
+-   Video decoder subframe support
+-   Video decoder automatic packet-loss, data corruption, and keyframe
+    request handling for RTP / WebRTC / RTSP
+-   MP4 and Matroska muxers now support profile/level/resolution changes
+    for H264/H265 input streams (i.e. codec data changing on the fly)
+-   MP4 muxing mode that initially creates a fragmented mp4 which is
+    converted to a regular mp4 on EOS
+-   Audio support for the WebKit Port for Embedded (WPE) web page source
+    element
+-   CUDA based video color space convert and rescale elements and
+    upload/download elements
+-   NVIDIA memory:NVMM support for OpenGL glupload and gldownload
+    elements
+-   Many WebRTC improvements
+-   The new VA-API plugin implemention fleshed out with more decoders
+    and new postproc elements
+-   AppSink API to retrieve events in addition to buffers and buffer
+    lists
+-   AppSrc gained more configuration options for the internal queue
+    (leakiness, limits in buffers and time, getters to read current
+    levels)
+-   Updated Rust bindings and many new Rust plugins
+-   Improved support for custom minimal GStreamer builds
+-   Support build against FFmpeg 5.0
+-   Linux Stateless CODEC support gained MPEG2 and VP9
+-   Windows Direct3D11/DXVA decoder gained AV1 and MPEG2 support
+-   Lots of new plugins, features, performance improvements and bug
+    fixes
 
 Major new features and changes
 
 Noteworthy new features and API
 
--   this section will be filled in in due course
+-   gst_element_get_request_pad() has been deprecated in favour of the
+    newly-added gst_element_request_pad_simple() which does the exact
+    same thing but has a less confusing name that hopefully makes clear
+    that the function request a new pad rather than just retrieves an
+    already-existing request pad.
+
+Development in GitLab was switched to a single git repository containing all the modules
+
+The GStreamer multimedia framework is a set of libraries and plugins
+split into a number of distinct modules which are released independently
+and which have so far been developed in separate git repositories in
+freedesktop.org GitLab.
+
+In addition to these separate git repositories there was a gst-build
+module that would use the Meson build systems’s subproject feature to
+download each individual module and then build everything in one go. It
+would also provide an uninstalled development environment that made it
+easy to work on GStreamer and use or test versions other than the
+system-installed GStreamer version.
+
+All of these modules have now (as of 28 September 2021) been merged into
+a single git repository (“Mono repository” or “monorepo”) which should
+simplify development workflows and continuous integration, especially
+where changes need to be made to multiple modules at once.
+
+This mono repository merge will primarily affect GStreamer developers
+and contributors and anyone who has workflows based on the GStreamer git
+repositories.
+
+The Rust bindings and Rust plugins modules have not been merged into the
+mono repository at this time because they follow a different release
+cycle.
+
+The mono repository lives in the existing GStreamer core git repository
+in GitLab in the new main branch and all future development will happen
+on this branch.
+
+Modules will continue to be released as separate tarballs.
+
+For more details, please see the GStreamer mono repository FAQ.
+
+GstPlay: new high-level playback library replacing GstPlayer
+
+-   GstPlay is a new high-level playback library that replaces the older
+    GstPlayer API. It is basically the same API as GstPlayer but
+    refactored to use bus messages for application notifications instead
+    of GObject signals. There is still a signal adapter object for those
+    who prefer signals. Since the existing GstPlayer API is already in
+    use in various applications, it didn’t seem like a good idea to
+    break it entirely. Instead a new API was added, and it is expected
+    that this new GstPlay API will be moved to gst-plugins-base in
+    future.
+
+-   The existing GstPlayer API is scheduled for deprecation and will be
+    removed at some point in the future (e.g. in GStreamer 1.24), so
+    application developers are urged to migrate to the new GstPlay API
+    at their earliest convenience.
+
+WebM alpha decoding
+
+-   Implement WebM alpha decoding (VP8/VP9 with alpha), which required
+    support and additions in various places. This is supported both with
+    software decoders and hardware-accelerated decoders.
+
+-   VP8/VP9 don’t support alpha components natively in the codec, so the
+    way this is implemented in WebM is by encoding the alpha plane with
+    transparency data as a separate VP8/VP9 stream. Inside the WebM
+    container (a variant of Matroska) this is coded as a single video
+    track with the “normal” VP8/VP9 video data making up the main video
+    data and each frame of video having an encoded alpha frame attached
+    to it as extra data ("BlockAdditional").
+
+-   matroskademux has been extended extract this per-frame alpha side
+    data and attach it in form of a GstVideoCodecAlphaMeta to the
+    regular video buffers. Note that this new meta is specific to this
+    VP8/VP9 alpha support and can’t be used to just add alpha support to
+    other codecs that don’t support it. Lastly, matroskademux also
+    advertises the fact that the streams contain alpha in the caps.
+
+-   The new codecalpha plugin contains various bits of infrastructure to
+    support autoplugging and debugging:
+
+    -   codecalphademux splits out the alpha stream from the metas on
+        the regular VP8/VP9 buffers
+    -   alphacombine takes two decoded raw video streams (one alpha, one
+        the regular video) and combines it into a video stream with
+        alpha
+    -   vp8alphadecodebin + vp9alphadecodebin are wrapper bins that use
+        the regular vp8dec and vp9dec software decoders to decode
+        regular and alpha streams and combine them again. To decodebin
+        these look like regular decoders which ju
+    -   The V4L2 CODEC plugin has stateless VP8/VP9 decoders that can
+        decode both alpha and non-alpha stream with a single decoder
+        instance
+
+-   A new AV12 video format was added which is basically NV12 with an
+    alpha plane, which is more convenient for many hardware-accelerated
+    decoders.
+
+-   Watch Nicolas Dufresne’s LCA 2022 talk “Bringing WebM Alpha support
+    to GStreamer” for all the details and a demo.
+
+RTP Header Extensions Base Class and Automatic Header Extension Handling in RTP Payloaders and Depayloaders
+
+-   RTP Header Extensions are specified in RFC 5285 and provide a way to
+    add small pieces of data to RTP packets in between the RTP header
+    and the RTP payload. This is often used for per-frame metadata,
+    extended timestamps or other application-specific extra data. There
+    are several commonly-used extensions specified in various RFCs, but
+    senders are free to put any kind of data in there, as long as sender
+    and receiver both know what that data is. Receivers that don’t know
+    about the header extensions will just skip the extra data without
+    ever looking at it. These header extensions can often be combined
+    with any kind of payload format, so may need to be supported by many
+    RTP payloader and depayloader elements.
+
+-   Inserting and extracting RTP header extension data has so far been a
+    bit inconvenient in GStreamer: There are functions to add and
+    retrieve RTP header extension data from RTP packets, but nothing
+    works automatically, even for common extensions. People would have
+    to do the insertion/extraction either in custom elements
+    before/after the RTP payloader/depayloader, or inside pad probes,
+    which isn’t very nice.
+
+-   This release adds various pieces of new infrastructure for generic
+    RTP header extension handling, as well as some implementations for
+    common extensions:
+
+    -   GstRTPHeaderExtension is a new helper base class for reading and
+        writing RTP header extensions. Nominally this subclasses
+        GstElement, but only so these extensions are stored in the
+        registry where they can be looked up by URI or name. They don’t
+        have pads and don’t get added to the pipeline graph as an
+        element.
+
+    -   "add-extension" and "clear-extension" action signals on RTP
+        payloaders and depayloaders for manual extension management
+
+    -   The "request-extension" signal will be emitted if an extension
+        is encountered that requires explicit mapping by the application
+
+    -   new "auto-header-extension" property on RTP payloaders and
+        depayloaders for automatic handling of known header extensions.
+        This is enabled by default. The extensions must be signalled via
+        caps / SDP.
+
+    -   RTP header extension implementations:
+
+        -   rtphdrextclientaudiolevel: Client-to-Mixer Audio Level
+            Indication (RFC 6464) (also see below)
+        -   rtphdrextcolorspace: Color Space extension, extends RTP
+            packets with color space and high dynamic range (HDR)
+            information
+        -   rtphdrexttwcc: Transport Wide Congestion Control support
+
+-   gst_rtp_buffer_remove_extension_data() is a new helper function to
+    remove an RTP header extension from an RTP buffer
+
+-   The existing gst_rtp_buffer_set_extension_data() now also supports
+    shrinking the extension data in size
+
+AppSink and AppSrc improvements
+
+-   appsink: new API to pull events out of appsink in addition to
+    buffers and buffer lists.
+
+    There was previously no way for users to receive incoming events
+    from appsink properly serialised with the data flow, even if they
+    are serialised events. The reason for that is that the only way to
+    intercept events was via a pad probe on the appsink sink pad, but
+    there is also internal queuing inside of appsink, so it’s difficult
+    to ascertain the right order of everything in all cases.
+
+    There is now a new "new-serialized-event" signal which will be
+    emitted when there’s a new event pending (just like the existing
+    "new-sample" signal). The "emit-signals" property must be set to
+    TRUE in order to activate this (but it’s also fine to just pull from
+    the application thread without using the signals).
+
+    gst_app_sink_pull_object() and gst_app_sink_try_pull_object() can be
+    used to pull out either an event or a new sample carrying a buffer
+    or buffer list, whatever is next in the queue.
+
+    EOS events will be filtered and will not be returned. EOS handling
+    can be done the ususal way, same as with _pull_sample().
+
+-   appsrc: allow configuration of internal queue limits in time and
+    buffers and add leaky mode.
+
+    There is internal queuing inside appsrc so the application thread
+    can push data into the element which will then be picked up by the
+    source element’s streaming thread and pushed into the pipeline from
+    that streaming thread. This queue is unlimited by default and until
+    now it was only possible to set a maximum size limit in bytes. When
+    that byte limit is reached, the pushing thread (application thread)
+    would be blocked until more space becomes available.
+
+    A limit in bytes is not particularly useful for many use cases, so
+    now it is possible to also configure limits in time and buffers
+    using the new "max-time" and "max-buffers" properties. Of course
+    there are also matching new read-only"current-level-buffers" and
+    "current-level-time properties" properties to query the current fill
+    level of the internal queue in time and buffers.
+
+    And as if that wasn’t enough the internal queue can also be
+    configured as leaky using the new "leaky-type" property. That way
+    when the queue is full the application thread won’t be blocked when
+    it tries to push in more data, but instead either the new buffer
+    will be dropped or the oldest data in the queue will be dropped.
+
+Better string serialization of nested GstCaps and GstStructures
+
+-   New string serialisation format for structs and caps that can handle
+    nested structs and caps properly by using brackets to delimit nested
+    items (e.g. some-struct, some-field=[nested-struct, nested=true]).
+    Unlike the default format the new variant can also support more than
+    one level of nesting. For backwards-compatibility reasons the old
+    format is still output by default when serialising caps and structs
+    using the existing API. The new functions gst_caps_serialize() and
+    gst_structure_serialize() can be used to output strings in the new
+    format.
+
+Convenience API for custom GstMetas
+
+-   New convenience API to register and create custom GstMetas:
+    gst_meta_register_custom() and gst_buffer_add_custom_meta(). Such
+    custom meta is backed by a GstStructure and does not require that
+    users of the API expose their GstMeta implementation as public API
+    for other components to make use of it. In addition, it provides a
+    simpler interface by ignoring the impl vs. api distinction that the
+    regular API exposes. This new API is meant to be the meta
+    counterpart to custom events and messages, and to be more convenient
+    than the lower-level API when the absolute best performance isn’t a
+    requirement. The reason it’s less performant than a “proper” meta is
+    that a proper meta is just a C struct in the end whereas this goes
+    through the GstStructure API which has a bit more overhead, which
+    for most scenarios is negligible however. This new API is useful for
+    experimentation or proprietary metas, but also has some limitations:
+    it can only be used if there’s a single producer of these metas;
+    it’s not allowed to register the same custom meta multiple times or
+    from multiple places.
+
+Additional Element Properties on Encoding Profiles
+
+-   GstEncodingProfile: The new "element-properties" and
+    gst_encoding_profile_set_element_properties() API allows
+    applications to set additional element properties on encoding
+    profiles to configure muxers and encoders. So far the encoding
+    profile template was the only place where this could be specified,
+    but often what applications want to do is take a ready-made encoding
+    profile shipped by GStreamer or the application and then tweak the
+    settings on top of that, which is now possible with this API. Since
+    applications can’t always know in advance what encoder element will
+    be used in the end, it’s even possible to specify properties on a
+    per-element basis.
+
+    Encoding Profiles are used in the encodebin, transcodebin and
+    camerabin elements and APIs to configure output formats (containers
+    and elementary streams).
+
+Audio Level Indication Meta for RFC 6464
+
+-   New GstAudioLevelMeta containing Audio Level Indication as per RFC
+    6464
+
+-   The level element has been updated to add GstAudioLevelMeta on
+    buffers if the "audio-level-meta" property is set to TRUE. This can
+    then in turn be picked up by RTP payloaders to signal the audio
+    level to receivers through RTP header extensions (see above).
+
+-   New Client-to-Mixer Audio Level Indication (RFC6464) RTP Header
+    Extension which should be automatically created and used by RTP
+    payloaders and depayloaders if their "auto-header-extension"
+    property is enabled and if the extension is part of the RTP caps.
+
+Automatic packet loss, data corruption and keyframe request handling for video decoders
+
+-   The GstVideoDecoder base class has gained various new APIs to
+    automatically handle packet loss and data corruption better by
+    default, especially in RTP, RTSP and WebRTC streaming scenarios, and
+    to give subclasses more control about how they want to handle
+    missing data:
+
+    -   Video decoder subclasses can mark output frames as corrupted via
+        the new GST_VIDEO_CODEC_FRAME_FLAG_CORRUPTED flag
+
+    -   A new "discard-corrupted-frames" property allows applications to
+        configure decoders so that corrupted frames are directly
+        discarded instead of being forwarded inside the pipeline. This
+        is a replacement for the "output-corrupt" property of the FFmpeg
+        decoders.
+
+    -   RTP depayloaders can now signal to decoders that data is missing
+        when sending GAP events for lost packets. GAP events can be sent
+        for various reason in a GStreamer pipeline. Often they are just
+        used to let downstream elements know that there isn’t a buffer
+        available at the moment, so downstream elements can move on
+        instead of waiting for one. They are also sent by RTP
+        depayloaders in the case that packets are missing, however, and
+        so far a decoder was not able to differentiate the two cases.
+        This has been remedied now: GAP events can be decorated with
+        gst_event_set_gap_flags() and GST_GAP_FLAG_MISSING_DATA to let
+        decoders now what happened, and decoders can then use that in
+        some cases to handle missing data better.
+
+    -   The GstVideoDecoder::handle_missing_data vfunc was added to
+        inform subclasses about packet loss or missing data and let them
+        handle it in their own way if they like.
+
+    -   gst_video_decoder_set_needs_sync_point() lets subclasses signal
+        that they need the stream to start with a sync point. If
+        enabled, the base class will discard all non-sync point frames
+        in the beginning and after a flush and does not pass them to the
+        subclass. Furthermore, if the first frame is not a sync point,
+        the base class will try and request a sync frame from upstream
+        by sending a force-key-unit event (see next items).
+
+    -   New "automatic-request-sync-points" and
+        "automatic-request-sync-point-flags" properties to automatically
+        request sync points when needed, e.g. on packet loss or if the
+        first frame is not a keyframe. Applications may want to enable
+        this on decoders operating in e.g. RTP/WebRTC/RTSP receiver
+        pipelines.
+
+    -   The new "min-force-key-unit-interval" property can be used to
+        ensure there’s a minimal interval between keyframe requests to
+        upstream (and/or the sender) and we’re not flooding the sender
+        with key unit requests.
+
+    -   gst_video_decoder_request_sync_point() allows subclasses to
+        request a new sync point (e.g. if they choose to do their own
+        missing data handling). This will still honour the
+        "min-force-key-unit-interval" property if set.
+
+Improved support for custom minimal GStreamer builds
+
+-   Element registration and registration of other plugin features
+    inside plugin init functions has been improved in order to
+    facilitate minimal custom GStreamer builds.
+
+-   A number of new macros have been added to declare and create
+    per-element and per-pluginfeature register functions in all plugins,
+    and then call those from the per-plugin plugin_init functions:
+
+    -   GST_ELEMENT_REGISTER_DEFINE,
+        GST_DEVICE_PROVIDER_REGISTER_DEFINE,
+        GST_DYNAMIC_TYPE_REGISTER_DEFINE, GST_TYPE_FIND_REGISTER_DEFINE
+        for the actual registration call with GStreamer
+    -   GST_ELEMENT_REGISTER, GST_DEVICE_PROVIDER_REGISTER,
+        GST_DYNAMIC_TYPE_REGISTER, GST_PLUGIN_STATIC_REGISTER,
+        GST_TYPE_FIND_REGISTER to call the registration function defined
+        by the REGISTER_DEFINE macro
+    -   GST_ELEMENT_REGISTER_DECLARE,
+        GST_DEVICE_PROVIDER_REGISTER_DECLARE,
+        GST_DYNAMIC_TYPE_REGISTER_DECLARE,
+        GST_TYPE_FIND_REGISTER_DECLARE to declare the registration
+        function defined by the REGISTER_DEFINE macro
+    -   and various variants for advanced use cases.
+
+-   This means that applications can call the per-element and
+    per-pluginfeature registration functions for only the elements they
+    need instead of registering plugins as a whole with all kinds of
+    elements that may not be required (e.g. encoder and decoder instead
+    of just decoder). In case of static linking all unused functions and
+    their dependencies would be removed in this case by the linker,
+    which helps minimise binary size for custom builds.
+
+-   gst_init() will automatically call a gst_init_static_plugins()
+    function if one exists.
+
+-   See the GStreamer static build documentation and Stéphane’s blog
+    post Generate a minimal GStreamer build, tailored to your needs for
+    more details.
 
 New elements
 
--   this section will be filled in in due course
+-   New aesdec and aesenc elements for AES encryption and decryption in
+    a custom format.
+
+-   New encodebin2 element with dynamic/sometimes source pads in order
+    to support the option of doing the muxing outside of encodebin,
+    e.g. in combination with a splitmuxsink.
+
+-   New fakeaudiosink and videocodectestsink elements for testing and
+    debugging (see below for more details)
+
+-   rtpisacpay, rtpisacdepay: new RTP payloader and depayloader for iSAC
+    audio codec
+
+-   rtpst2022-1-fecdec, rtpst2022-1-fecenc: new elements providing SMPTE
+    2022-1 2-D Forward Error Correction. More details in Mathieu’s blog
+    post.
+
+-   isac: new plugin wrapping the Internet Speech Audio Codec reference
+    encoder and decoder from the WebRTC project.
+
+-   asio: plugin for Steinberg ASIO (Audio Streaming Input/Output) API
+
+-   gssrc, gssink: add source and sink for Google Cloud Storage
+
+-   onnx: new plugin to apply ONNX neural network models to video
+
+-   openaptx: aptX and aptX-HD codecs using libopenaptx (v0.2.0)
+
+-   qroverlay, debugqroverlay: new elements that allows overlaying data
+    on top of video in form of a QR code
+
+-   cvtracker: new OpenCV-based tracker element
+
+-   av1parse, vp9parse: new parsers for AV1 and VP9 video
+
+-   va: work on the new VA-API plugin implementation for
+    hardware-accelerated video decoding and encoding has continued at
+    pace, with various new decoders and filters having joined the
+    initial vah264dec:
+
+    -   vah265dec: VA-API H.265 decoder
+    -   vavp8dec: VA-API VP8 decoder
+    -   vavp9dec: VA-API VP9 decoder
+    -   vaav1dec: VA-API AV1 decoder
+    -   vampeg2dec: VA-API MPEG-2 decoder
+    -   vadeinterlace: : VA-API deinterlace filter
+    -   vapostproc: : VA-API postproc filter (color conversion,
+        resizing, cropping, color balance, video rotation, skin tone
+        enhancement, denoise, sharpen)
+
+    See Víctor’s blog post “GstVA in GStreamer 1.20” for more details
+    and what’s coming up next.
+
+-   vaapiav1dec: new AV1 decoder element (in gstreamer-vaapi)
+
+-   msdkav1dec: hardware-accelerated AV1 decoder using the Intel Media
+    SDK / oneVPL
+
+-   nvcodec plugin for NVIDIA NVCODEC API for hardware-accelerated video
+    encoding and decoding:
+
+    -   cudaconvert, cudascale: new CUDA based video color space convert
+        and rescale elements
+    -   cudaupload, cudadownload: new helper elements for memory
+        transfer between CUDA and system memory spaces
+    -   nvvp8sldec, nvvp9sldec: new GstCodecs-based VP8/VP9 decoders
+
+-   Various new hardware-accelerated elements for Windows:
+
+    -   d3d11screencapturesrc: new desktop capture element, including a
+        GstDeviceProvider implementation to enumerate/select target
+        monitors for capture.
+    -   d3d11av1dec and d3d11mpeg2dec: AV1 and MPEG-2 decoders
+    -   d3d11deinterlace: deinterlacing filter
+    -   d3d11compositor: video composing element
+    -   see Windows section below for more details
+
+-   new Rust plugins:
+
+    -   audiornnoise: Removes noise from an audio stream
+    -   awstranscribeparse: Parses AWS audio transcripts into timed text
+        buffers
+    -   ccdetect: Detects if valid closed captions are present in a
+        closed captions stream
+    -   cea608tojson: Converts CEA-608 Closed Captions to a JSON
+        representation
+    -   cmafmux: CMAF fragmented MP4 muxer
+    -   dashmp4mux: DASH fragmented MP4 muxer
+    -   isofmp4mux: ISO fragmented MP4 muxer
+    -   ebur128level: EBU R128 Loudness Level Measurement
+    -   ffv1dec: FFV1 video decoder
+    -   gtk4paintablesink: GTK4 video sink, which provides a
+        GdkPaintable that can be rendered in various widgets
+    -   hlssink3: HTTP Live Streaming sink
+    -   hrtfrender: Head-Related Transfer Function (HRTF) renderer
+    -   hsvdetector: HSV colorspace detector
+    -   hsvfilter: HSV colorspace filter
+    -   jsongstenc: Wraps buffers containing any valid top-level JSON
+        structures into higher level JSON objects, and outputs those as
+        ndjson
+    -   jsongstparse: Parses ndjson as output by jsongstenc
+    -   jsontovtt: converts JSON to WebVTT subtitles
+    -   regex: Applies regular expression operations on text
+    -   roundedcorners: Adds rounded corners to video
+    -   spotifyaudiosrc: Spotify source
+    -   textahead: Display upcoming text buffers ahead (e.g. for
+        Karaoke)
+    -   transcriberbin: passthrough bin that transcribes raw audio to
+        closed captions using awstranscriber and puts the captions as
+        metas onto the video
+    -   tttojson: Converts timed text to a JSON representation
+    -   uriplaylistbin: Playlist source bin
+    -   webpdec-rs: WebP image decoder with animation support
+
+-   New plugin codecalpha with elements to assist with WebM Alpha
+    decoding
+
+    -   codecalphademux: Split stream with GstVideoCodecAlphaMeta into
+        two streams
+    -   alphacombine: Combine two raw video stream (I420 or NV12) as one
+        stream with alpha channel (A420 or AV12)
+    -   vp8alphadecodebin: A bin to handle software decoding of VP8 with
+        alpha
+    -   vp9alphadecodebin: A bin to handle software decoding of VP9 with
+        alpha
+
+-   New hardware accelerated elements for Linux:
+
+    -   v4l2slmpeg2dec: Support for Linux Stateless MPEG2 decoders
+    -   v4l2slvp9dec: Support for Linux Stateless VP9 decoders
+    -   v4l2slvp8alphadecodebin: Support HW accelerated VP8 with alpha
+        layer decoding
+    -   v4l2slvp9alphadecodebin: Support HW accelerated VP9 with alpha
+        layer decoding
 
 New element features and additions
 
--   this section will be filled in in due course
+-   assrender: handle more font mime types; better interaction with
+    matroskademux for embedded fonts
+
+-   audiobuffersplit: Add support for specifying output buffer size in
+    bytes (not just duration)
+
+-   audiolatency: new "samplesperbuffer" property so users can configure
+    the number of samples per buffer. The default value is 240 samples
+    which is equivalent to 5ms latency with a sample rate of 48000,
+    which might be larger than actual buffer size of audio capture
+    device.
+
+-   audiomixer, audiointerleave, GstAudioAggregator: now keep a count of
+    samples that are dropped or processed as statistic and can be made
+    to post QoS messages on the bus whenever samples are dropped by
+    setting the "qos-messages" property on input pads.
+
+-   audiomixer, compositor: improved handling of new inputs added at
+    runtime. New API was added to the GstAggregator base class to allow
+    subclasses to opt into an aggregation mode where inactive pads are
+    ignored when processing input buffers
+    (gst_aggregator_set_ignore_inactive_pads(),
+    gst_aggregator_pad_is_inactive()). An “inactive pad” in this context
+    is a pad which, in live mode, hasn’t yet received a first buffer,
+    but has been waited on at least once. What would happen usually in
+    this case is that the aggregator would wait for data on this pad
+    every time, up to the maximum configured latency. This would
+    inadvertently push mixer elements in live mode to the configured
+    latency envelope and delay processing when new inputs are added at
+    runtime until these inputs have actually produced data. This is
+    usually undesirable. With this new API, new inputs can be added
+    (requested) and configured and they won’t delay the data processing.
+    Applications can opt into this new behaviour by setting the
+    "ignore-inactive-pads" property on compositor, audiomixer or other
+    GstAudioAggregator-based elements.
+
+-   cccombiner: implement “scheduling” of captions. So far cccombiner’s
+    behaviour was essentially that of a funnel: it strictly looked at
+    input timestamps to associate together video and caption buffers.
+    Now it will try to smoothly schedule caption buffers in order to
+    have exactly one per output video buffer. This might involve
+    rewriting input captions, for example when the input is CDP then
+    sequence counters are rewritten, time codes are dropped and
+    potentially re-injected if the input video frame had a time code
+    meta. This can also lead to the input drifting from synchronization,
+    when there isn’t enough padding in the input stream to catch up. In
+    that case the element will start dropping old caption buffers once
+    the number of buffers in its internal queue reaches a certain limit
+    (configurable via the "max-scheduled" property). The new original
+    funnel-like behaviour can be restored by setting the "scheduling"
+    property to FALSE.
+
+-   ccconverter: new "cdp-mode" property to specify which sections to
+    include in CDP packets (timecode, CC data, service info). Various
+    software, including ffmpeg’s Decklink support, fails parsing CDP
+    packets that contain anything but CC data in the CDP packets.
+
+-   clocksync: new "sync-to-first" property for automatic timestamp
+    offset setup: if set clocksync will set up the "ts-offset" value
+    based on the first buffer and the pipeline’s running time when the
+    first buffer arrived. The newly configured "ts-offset" in this case
+    would be the value that allows outputting the first buffer without
+    waiting on the clock. This is useful for example to feed a non-live
+    input into an already-running pipeline.
+
+-   compositor:
+
+    -   multi-threaded input conversion and compositing. Set the
+        "max-threads" property to activate this.
+    -   new "sizing-policy" property to support display aspect ratio
+        (DAR)-aware scaling. By default the image is scaled to fill the
+        configured destination rectangle without padding and without
+        keeping the aspect ratio. With sizing-policy=keep-aspect-ratio
+        the input image is scaled to fit the destination rectangle
+        specified by GstCompositorPad:{xpos, ypos, width, height}
+        properties preserving the aspect ratio. As a result, the image
+        will be centered in the destination rectangle with padding if
+        necessary.
+    -   new "zero-size-is-unscaled" property on input pads. By default
+        pad width=0 or pad height=0 mean that the stream should not be
+        scaled in that dimension. But if the "zero-size-is-unscaled"
+        property is set to FALSE a width or height of 0 is instead
+        interpreted to mean that the input image on that pad should not
+        be composited, which is useful when creating animations where an
+        input image is made smaller and smaller until it disappears.
+    -   improved handling of new inputs at runtime via
+        "ignore-inactive-pads"property (see above for details)
+    -   allow output format with alpha even if none of the inputs have
+        alpha (also glvideomixer and other GstVideoAggregator
+        subclasses)
+
+-   dashsink: add h265 codec support and signals for allowing custom
+    playlist/fragment output
+
+-   decodebin3:
+
+    -   improved decoder selection, especially for hardware decoders
+    -   make input activation “atomic” when adding inputs dynamically
+    -   better interleave handling: take into account decoder latency
+        for interleave size
+
+-   decklink:
+
+    -   Updated DeckLink SDK to 11.2 to support DeckLink 8K Pro
+    -   decklinkvideosrc:
+        -   More accurate and stable capture timestamps: use the
+            hardware reference clock time when the frame was finished
+            being captured instead of a clock time much further down the
+            road.
+        -   Automatically detect widescreen vs. normal NTSC/PAL
+
+-   encodebin:
+
+    -   add “smart encoding” support for H.265, VP8 and VP9 (i.e. only
+        re-encode where needed and otherwise pass through encoded video
+        as-is).
+    -   H264/H265 smart encoding improvements: respect user-specified
+        stream-format, but if not specified default to avc3/hvc1 with
+        in-band SPS/PPS/VPS signalling for more flexibility.
+    -   new encodebin2 element with dynamic/sometimes source pads in
+        order to support the option of doing the muxing outside of
+        encodebin, e.g. in combination with splitmuxsink.
+    -   add APIs to set element properties on encoding profiles (see
+        below)
+
+-   errorignore: new "ignore-eos" property to also ignore FLOW_EOS from
+    downstream elements
+
+-   giosrc: add support for growing source files: applications can
+    specify that the underlying file being read is growing by setting
+    the "is-growing" property. If set, the source won’t EOS when it
+    reaches the end of the file, but will instead start monitoring it
+    and will start reading data again whenever a change is detected. The
+    new "waiting-data" and "done-waiting-data" signals keep the
+    application informed about the current state.
+
+-   gtksink, gtkglsink:
+
+    -   scroll event support: forwarded as navigation events into the
+        pipeline
+    -   "video-aspect-ratio-override" property to force a specific
+        aspect ratio
+    -   "rotate-method" property and support automatic rotation based on
+        image tags
+
+-   identity: new "stats" property allows applications to retrieve the
+    number of bytes and buffers that have passed through so far.
+
+-   interlace: add support for more formats, esp 10-bit, 12-bit and
+    16-bit ones
+
+-   jack: new "low-latency" property for automatic latency-optimized
+    setting and "port-names" property to select ports explicitly
+
+-   jpegdec: support output conversion to RGB using libjpeg-turbo (for
+    certain input files)
+
+-   line21dec:
+
+    -   "mode" property to control whether and how detected closed
+        captions should be inserted in the list of existing close
+        caption metas on the input frame (if any): add, drop, or
+        replace.
+    -   "ntsc-only" property to only look for captions if video has NTSC
+        resolution
+
+-   line21enc: new "remove-caption-meta" to remove metas from output
+    buffers after encoding the captions into the video data; support for
+    CDP closed captions
+
+-   matroskademux, matroskamux: Add support for ffv1, a lossless
+    intra-frame video coding format.
+
+-   matroskamux: accept in-band SPS/PPS/VPS for H264 and H265
+    (i.e. stream-format avc3 and hev1) which allows on-the-fly
+    profile/level/resolution changes.
+
+-   matroskamux: new "cluster-timestamp-offset" property, useful for use
+    cases where the container timestamps should map to some absolute
+    wall clock time, for example.
+
+-   rtpsrc: add "caps" property to allow explicit setting of the caps
+    where needed
+
+-   mpegts: support SCTE-35 passthrough via new "send-scte35-events"
+    property on MPEG-TS demuxer tsdemux. When enabled, SCTE 35 sections
+    (eg ad placement opportunities) are forwarded as events donwstream
+    where they can be picked up again by mpegtsmux. This required a
+    semantic change in the SCTE-35 section API: timestamps are now in
+    running time instead of muxer pts.
+
+-   tsdemux: Handle PCR-less MPEG-TS streams; more robust timestamp
+    handling in certain corner cases and for poorly muxed streams.
+
+-   mpegtsmux:
+
+    -   More conformance improvements to make MPEG-TS analyzers happy:
+        -   PCR timing accuracy: Improvements to the way mpegtsmux
+            outputs PCR observations in CBR mode, so that a PCR
+            observation is always inserted when needed, so that we never
+            miss the configured pcr-interval, as that triggers various
+            MPEG-TS analyser errors.
+        -   Improved PCR/SI scheduling
+    -   Don’t write PCR until PAT/PMT are output to make sure streams
+        start cleanly with a PAT/PMT.
+    -   Allow overriding the automatic PMT PID selection via
+        application-supplied PMT_%d fields in the prog-map
+        structure/property.
+
+-   mp4mux:
+
+    -   new "first-moov-then-finalise" mode for fragmented output where
+        the output will start with a self-contained moov atom for the
+        first fragment, and then produce regular fragments. Then at the
+        end when the file is finalised, the initial moov is invalidated
+        and a new moov is written covering the entire file. This way the
+        file is a “fragmented mp4” file while it is still being written
+        out, and remains playable at all times, but at the end it is
+        turned into a regular mp4 file (with former fragment headers
+        remaining as unused junk data in the file).
+    -   support H.264 avc3 and H.265 hvc1 stream formats as input where
+        the codec data is signalled in-band inside the bitstream instead
+        of caps/file headers.
+    -   support profile/level/resolution changes for H264/H265 input
+        streams (i.e. codec data changing on the fly). Each codec_data
+        is put into its own SampleTableEntry inside the stsd, unless the
+        input is in avc3 stream format in which case it’s written
+        in-band an not in the headers.
+
+-   multifilesink: new ""min-keyframe-distance"" property to make
+    minimum distance between keyframes in next-file=key-frame mode
+    configurable instead of hard-coding it to 10 seconds.
+
+-   mxfdemux has seen a big refactoring to support non-frame wrappings
+    and more accurate timestamp/seek handling for some formats
+
+-   msdk plugin for hardware-accelerated video encoding and decoding
+    using the Intel Media SDK:
+
+    -   oneVPL support (Intel oneAPI Video Processing Library)
+    -   AV1 decoding support
+    -   H.264 decoder now supports constrained-high and progressive-high
+        profiles
+    -   H.264 encoder:
+        -   more configuration options (properties):
+            "intra-refresh-type", "min-qp" , "max-qp", "p-pyramid",
+            "dblk-idc"
+    -   H.265 encoder:
+        -   can output main-still-picture profile
+        -   now inserts HDR SEIs (mastering display colour volume and
+            content light level)
+        -   more configuration options (properties):
+            "intra-refresh-type", "min-qp" , "max-qp", "p-pyramid",
+            "b-pyramid", "dblk-idc", "transform-skip"
+        -   support for RGB 10bit format
+    -   External bitrate control in encoders
+    -   Video post proc element msdkvpp gained support for 12-bit pixel
+        formats P012_LE, Y212_LE and Y412_LE
+
+-   nvh264sldec: interlaced stream support
+
+-   openh264enc: support main, high, constrained-high and
+    progressive-high profiles
+
+-   openjpeg: support for multithreaded decoding and encoding
+
+-   rtspsrc: now supports IPv6 also for tunneled mode (RTSP-over-HTTP);
+    new "ignore-x-server-reply" property to ignore the
+    x-server-ip-address server header reply in case of HTTP tunneling,
+    as it is often broken.
+
+-   souphttpsrc: Runtime compatibility support for libsoup2 and
+    libsoup3. libsoup3 is the latest major version of libsoup, but
+    libsoup2 and libsoup3 can’t co-exist in the same process because
+    there is no namespacing or versioning for GObject types. As a
+    result, it would be awkward if the GStreamer souphttpsrc plugin
+    linked to a specific version of libsoup, because it would only work
+    with applications that use the same version of libsoup. To make this
+    work, the soup plugin now tries to determine the libsoup version
+    used by the application (and its other dependencies) at runtime on
+    systems where GStreamer is linked dynamically. libsoup3 support is
+    still considered somewhat experimental at this point.
+
+-   srtsrc, srtsink: add signals for the application to accept/reject
+    incoming connections
+
+-   timeoverlay: new elapsed-running-time time mode which shows the
+    running time since the first running time (and each flush-stop).
+
+-   udpsrc: new timestamping mode to retrieve packet receive timestamps
+    from the kernel via socket control messages (SO_TIMESTAMPNS) on
+    supported platforms
+
+-   uritranscodebin: new setup-source and element-setup signals for
+    applications to configure elements used
+
+-   v4l2codecs plugin gained support for 4x4 and 32x32 tile formats
+    enabling some platforms or direct renders. Important memory usage
+    improvement.
+
+-   v4l2slh264dec now implements the final Linux uAPI as shipped on
+    Linux 5.11 and later.
+
+-   valve: add "drop-mode" property and provide two new modes of
+    operation: in drop-mode=forward-sticky-events sticky events
+    (stream-start, segment, tags, caps, etc.) are forwarded downstream
+    even when dropping is enabled; drop-mode=transform-to-gap will in
+    addition also convert buffers into gap events when dropping is
+    enabled, which lets downstream elements know that time is advancing
+    and might allow for preroll in many scenarios. By default all events
+    and all buffers are dropped when dropping is enabled, which can
+    cause problems with caps negotiation not progressing or branches not
+    prerolling when dropping is enabled.
+
+-   videocrop: support for many more pixel formats, e.g. planar YUV
+    formats with > 8bits and GBR* video formats; can now also accept
+    video not backed by system memory as long as downstream supports the
+    GstCropMeta
+
+-   videotestsrc: new smpte-rp-219 pattern for SMPTE75 RP-219 conformant
+    color bars
+
+-   vp8enc: finish support for temporal scalability: two new properties
+    ("temporal-scalability-layer-flags",
+    "temporal-scalability-layer-sync-flags") and a unit change on the
+    "temporal-scalability-target-bitrate" property (now expects bps);
+    also make temporal scalability details available to RTP payloaders
+    as buffer metadata.
+
+-   vp9enc: new properties to tweak encoder performance:
+
+    -   "aq-mode" to configure adaptive quantization modes
+    -   "frame-parallel-decoding" to configure whether to create a
+        bitstream that reduces decoding dependencies between frames
+        which allows staged parallel processing of more than one video
+        frames in the decoder. (Defaults to TRUE)
+    -   "row-mt", "tile-columns" and "tile-rows" so multithreading can
+        be enabled on a per-tile basis, instead of on a per tile-column
+        basis. In combination with the new "tile-rows" property, this
+        allows the encoder to make much better use of the available CPU
+        power.
+
+-   vp9dec, vp9enc: add support for 10-bit 4:2:0 and 4:2:2 YUV, as well
+    as 8-bit 4:4:4
+
+-   vp8enc, vp9enc now default to “good quality” for the deadline
+    property rather then “best quality”. Having the deadline set to best
+    quality causes the encoder to be absurdly slow, most real-life users
+    will prefer good-enough quality with better performance instead.
+
+-   wpesrc:
+
+    -   implement audio support: a new sometimes source pad will be
+        created for each audio stream created by the web engine.
+    -   move wpesrc to wpevideosrc and add a wrapper bin wpesrc to also
+        support audio
+    -   also handles web:// URIs now (same as cefsrc)
+    -   post messages with the estimated load progress on the bus
 
-Plugin and library moves
+-   x265enc: add negative DTS support, which means timestamps are now
+    offset by 1h same as with x264enc
+
+RTP Payloaders and Depayloaders
+
+-   rtpisacpay, rtpisacdepay: new RTP payloader and depayloader for iSAC
+    audio codec
+
+-   rtph264depay:
+
+    -   new "request-keyframe" property to make the depayloader
+        automatically request a new keyframe from the sender on packet
+        loss, consistent with the new property on rtpvp8depay.
+    -   new "wait-for-keyframe" property to make depayloader wait for a
+        new keyframe at the beginning and after packet loss (only
+        effective if the depayloader outputs AUs), consistent with the
+        existing property on rtpvp8depay.
+
+-   rtpopuspay, rtpopusdepay: support libwebrtc-compatible multichannel
+    audio in addition to the previously supported multichannel audio
+    modes
+
+-   rtpopuspay: add DTX (Discontinuous Transmission) support
+
+-   rtpvp8depay: new "request-keyframe" property to make the depayloader
+    automatically request a new keyframe from the sender on packet loss.
+
+-   rtpvp8pay: temporal scaling support
 
--   this section will be filled in in due course
+-   rtpvp9depay: Improved SVC handling (aggregate all layers)
+
+RTP Infrastructure
+
+-   rtpst2022-1-fecdec, rtpst2022-1-fecenc: new elements providing SMPTE
+    2022-1 2-D Forward Error Correction. More details in Mathieu’s blog
+    post.
+
+-   rtpreddec: BUNDLE support
+
+-   rtpredenc, rtpulpfecenc: add support for Transport-wide Congestion
+    Control (TWCC)
+
+-   rtpsession: new "twcc-feedback-interval" property to allow RTCP TWCC
+    reports to be scheduled on a timer instead of per marker-bit.
+
+Plugin and library moves
 
 -   There were no plugin moves or library moves in this cycle.
 
@@ -56,8 +998,6 @@ Plugin removals
 
 The following elements or plugins have been removed:
 
--   this section will be filled in in due course
-
 -   The ofa audio fingerprinting plugin has been removed. The MusicIP
     database has been defunct for years so this plugin is likely neither
     useful nor used by anyone.
@@ -71,135 +1011,575 @@ The following elements or plugins have been removed:
 
 Miscellaneous API additions
 
--   this section will be filled in in due course
+Core
+
+-   gst_buffer_new_memdup() is a convenience function for the
+    widely-used gst_buffer_new_wrapped(g_memdup(data,size),size)
+    pattern.
+
+-   gst_caps_features_new_single() creates a new single GstCapsFeatures,
+    avoiding the need to use the vararg function with NULL terminator
+    for simple cases.
+
+-   gst_element_type_set_skip_documentation() can be used by plugins to
+    signal that certain elements should not be included in the GStreamer
+    plugin documentation. This is useful for plugins where elements are
+    registered dynamically based on hardware capabilities and/or where
+    the available plugins and properties vary from system to system.
+    This is used in the d3d11 plugin for example to ensure that only the
+    list of default elements is advertised in the documentation.
+
+-   gst_type_find_suggest_empty_simple() is a new convenience function
+    for typefinders for cases where there’s only a media type and no
+    other fields.
+
+-   New API to create elements and set properties at construction time,
+    which is not only convenient, but also allows GStreamer elements to
+    have construct-only properties: gst_element_factory_make_full(),
+    gst_element_factory_make_valist(),
+    gst_element_factory_make_with_properties(),
+    gst_element_factory_create_full(),
+    gst_element_factory_create_valist(),
+    gst_element_factory_create_with_properties().
+
+-   GstSharedTaskPool: new “shared” task pool subclass with slightly
+    different default behaviour than the existing GstTaskPool which
+    would create unlimited number of threads for new tasks. The shared
+    taskpool creates up to N threads (default: 1) and then distributes
+    pending tasks to those threads round-robin style, and blocks if no
+    thread is available. It is possible to join tasks. This can be used
+    by plugins to implement simple multi-threaded processing and is used
+    for the new multi-threaded video conversion and compositing done in
+    GstVideoAggregator, videoconverter and compositor.
+
+Plugins Base Utils library
+
+-   GstDiscoverer:
+
+    -   gst_discoverer_container_info_get_tags() was added to retrieve
+        global/container tags (vs. per-stream tags). Per-Stream tags can
+        be retrieved via the existing
+        gst_discoverer_stream_info_get_tags().
+        gst_discoverer_info_get_tags(), which for many files returns a
+        confusing mix of stream and container tags, has been deprecated
+        in favour of the container/stream-specific functions.
+    -   gst_discoverer_stream_info_get_stream_number() returns a unique
+        integer identifier for a given stream within the given
+        GstDiscoverer context. (If this matches the stream number inside
+        the container bitstream that’s by coincidence and not by
+        design.)
+
+-   gst_pb_utils_get_caps_description_flags() can be used to query
+    whether certain caps represent a container, audio, video, image,
+    subtitles, tags, or something else. This only works for formats
+    known to GStreamer.
+
+-   gst_pb_utils_get_file_extension_from_caps() returns a possible file
+    extension for given caps.
+
+-   gst_codec_utils_h264_get_profile_flags_level(): Parses profile,
+    flags, and level from H264 AvcC codec_data. The format of H264 AVCC
+    extradata/sequence_header is documented in the ITU-T H.264
+    specification section 7.3.2.1.1 as well as in ISO/IEC 14496-15
+    section 5.3.3.1.2.
+
+-   gst_codec_utils_caps_get_mime_codec() to convert caps to a RFC 6381
+    compatible MIME codec string codec. Useful for providing the codecs
+    field inside the Content-Type HTTP header for containerized formats,
+    such as mp4 or matroska.
+
+GStreamer OpenGL integration library and plugins
+
+-   glcolorconvert: added suppport for converting the video formats
+    A420, AV12, BGR, BGRA, RGBP and BGRP.
+
+-   Added support to GstGLBuffer for persistent buffer mappings where a
+    Pixel Buffer Object (PBO) can be mapped by both the CPU and the GPU.
+    This removes a memcpy() when uploading textures or vertices
+    particularly when software decoders (e.g. libav) are direct
+    rendering into our memory. Improves transfer performance
+    significantly. Requires OpenGL 4.4, GL_ARB_buffer_storage or
+    GL_EXT_buffer_storage
+
+-   Added various helper functions for handling 4x4 matrices of affine
+    transformations as used by GstVideoAffineTransformationMeta.
+
+-   Add support to GstGLContext for allowing the application to control
+    the config (EGLConfig, GLXConfig, etc) used when creating the OpenGL
+    context. This allows the ability to choose between RGB16 or RGB10A2
+    or RGBA8 back/front buffer configurations that were previously
+    hardcoded. GstGLContext also supports retrieving the configuration
+    it was created with or from an externally provide OpenGL context
+    handle. This infrastructure is also used to create a compatible
+    config from an application/externally provided OpenGL context in
+    order to improve compatibility with other OpenGL frameworks and GUI
+    toolkits. A new environment variable GST_GL_CONFIG was also added to
+    be able to request a specific configuration from the command line.
+    Note: different platforms will have different functionality
+    available.
+
+-   Add support for choosing between EGL and WGL at runtime when running
+    on Windows. Previously this was a build-time switch. Allows use in
+    e.g. Gtk applications on Windows that target EGL/ANGLE without
+    recompiling GStreamer. gst_gl_display_new_with_type() can be used by
+    applications to choose a specific display type to use.
+
+-   Build fixes to explicitly check for Broadcom-specific libraries on
+    older versions of the Raspberry Pi platform. The Broadcom OpenGL ES
+    and EGL libraries have different filenames. Using the vc4 Mesa
+    driver on the Raspberry Pi is not affected.
+
+-   Added support to glupload and gldownload for transferring RGBA
+    buffers using the memory:NVMM available on the Nvidia Tegra family
+    of embedded devices.
+
+-   Added support for choosing libOpenGL and libGLX as used in a GLVND
+    environment on unix-based platforms. This allows using desktop
+    OpenGL and EGL without pulling in any GLX symbols as would be
+    required with libGL.
+
+Video library
+
+-   New raw video formats:
+
+    -   AV12 (NV12 with alpha plane)
+    -   RGBP and BGRP (planar RGB formats)
+    -   ARGB64 variants with specified endianness instead of host
+        endianness:
+        -   ARGB64_LE, ARGB64_BE
+        -   RGBA64_BE, RGBA64_LE
+        -   BGRA64_BE, BGRA64_LE
+        -   ABGR64_BE, ABGR64_LE
+
+-   gst_video_orientation_from_tag() is new convenience API to parse the
+    image orientation from a GstTagList.
+
+-   GstVideoDecoder subframe support (see below)
+
+-   GstVideoCodecState now also carries some HDR metadata
+
+-   Ancillary video data: implement transform functions for AFD/Bar
+    metas, so they will be forwarded in more cases
+
+MPEG-TS library
+
+This library only handles section parsing and such, see above for
+changes to the actual mpegtsmux and mpegtsdemux elements.
+
+-   many additions and improvements to SCTE-35 section parsing
+-   new API for fetching extended descriptors:
+    gst_mpegts_find_descriptor_with_extension()
+-   add support for SIT sections (Selection Information Tables)
+-   expose event-from-section constructor gst_event_new_mpegts_section()
+-   parse Audio Preselection Descriptor needed for Dolby AC-4
+
+GstWebRTC library + webrtcbin
+
+-   Change the way in which sink pads and transceivers are matched
+    together to support easier usage. If a pad is created without a
+    specific index (i.e. using sink_%u as the pad template), then an
+    available compatible transceiver will be searched for. If a specific
+    index is requested (i.e. sink_1) then if a transceiver for that
+    m-line already exists, that transceiver must match the new sink pad
+    request. If there is no transceiver available in either scenario, a
+    new transceiver is created. If a mixture of both sink_1 and sink_%u
+    requests result in an impossible situation, an error will be
+    produced at pad request time or from create offer/answer.
+
+-   webrtcbin now uses regular ICE nomination instead of libnice’s
+    default of aggressive ICE nomination. Regular ICE nomination is the
+    default recommended by various relevant standards and improves
+    connectivity in specific network scenarios.
+
+-   Add support for limiting the port range used for RTP with the
+    addition of the min-rtp-port and max-rtp-port properties on the ICE
+    object.
+
+-   Expose the SCTP transport as a property on webrtcbin to more closely
+    match the WebRTC specification.
+
+-   Added support for taking into account the data channel transport
+    state when determining the value of the "connection-state" property.
+    Previous versions of the WebRTC spec did not include the data
+    channel state when computing this value.
+
+-   Add configuration for choosing the size of the underlying sockets
+    used for transporting media data
+
+-   Always advertise support for the transport-cc RTCP feedback protocol
+    as rtpbin supports it. For full support, the configured caps (input
+    or through codec-preferences) need to include the relevant RTP
+    header extension.
+
+-   Numerous fixes to caps and media handling to fail-fast when an
+    incompatible situation is detected.
+
+-   Improved support for attaching the required media after a remote
+    offer has been set.
+
+-   Add support for dynamically changing the amount of FEC used for a
+    particular stream.
+
+-   webrtcbin now stops further SDP processing at the first error it
+    encounters.
+
+-   Completed support for either local or the remote closing a data
+    channel.
+
+-   Various fixes when performing BUNDLEing of the media streams in
+    relation to RTX and FEC usage.
+
+-   Add support for writing out QoS DSCP marking on outgoing packets to
+    improve reliability in some network scenarios.
+
+-   Improvements to the statistics returned by the get-stats signal
+    including the addition of the raw statistics from the internal
+    RTPSource, the TWCC stats when available.
+
+-   The webrtc library does not expose any objects anymore with public
+    fields. Instead properties have been added to replace that
+    functionality. If you are accessing such fields in your application,
+    switch to the corresponding properties.
+
+GstCodecs and Video Parsers
+
+-   Support for render delays to improve throughput across all CODECs
+    (used with NVDEC and V4L2).
+-   lots of improvements to parsers and the codec parsing decoder base
+    classes (H264, H265, VP8, VP9, AV1, MPEG-2) used for various
+    hardware-accelerate decoder APIs.
+
+Bindings support
+
+-   gst_allocation_params_new() allocates a GstAllocationParams struct
+    on the heap. This should only be used by bindings (and freed via
+    gst_allocation_params_free() then). In C code you would allocate
+    this on the stack and only init it in place.
+
+-   gst_debug_log_literal() can be used to log a string to the debug log
+    without going through any printf format expansion and associated
+    overhead. This is mostly useful for bindings such as the Rust
+    bindings which may have done their own formatting already .
+
+-   Provide non-inlined versions of refcounting APIs for various
+    GStreamer mini objects, so that they can be consumed by bindings
+    (e.g. gstreamer-sharp): gst_buffer_ref, gst_buffer_unref,
+    gst_clear_buffer, gst_buffer_copy, gst_buffer_replace,
+    gst_buffer_list_ref, gst_buffer_list_unref, gst_clear_buffer_list,
+    gst_buffer_list_copy, gst_buffer_list_replace, gst_buffer_list_take,
+    gst_caps_ref, gst_caps_unref, gst_clear_caps, gst_caps_replace,
+    gst_caps_take, gst_context_ref, gst_context_unref, gst_context_copy,
+    gst_context_replace, gst_event_replace, gst_event_steal,
+    gst_event_take, gst_event_ref, gst_event_unref, gst_clear_event,
+    gst_event_copy, gst_memory_ref, gst_memory_unref, gst_message_ref,
+    gst_message_unref, gst_clear_message, gst_message_copy,
+    gst_message_replace, gst_message_take, gst_promise_ref,
+    gst_promise_unref, gst_query_ref, gst_query_unref, gst_clear_query,
+    gst_query_copy, gst_query_replace, gst_query_take, gst_sample_ref,
+    gst_sample_unref, gst_sample_copy, gst_tag_list_ref,
+    gst_tag_list_unref, gst_clear_tag_list, gst_tag_list_replace,
+    gst_tag_list_take, gst_uri_copy, gst_uri_ref, gst_uri_unref,
+    gst_clear_uri.
+
+-   expose a GType for GstMiniObject
+
+-   gst_device_provider_probe() now returns non-floating device object
+
+API Deprecations
+
+-   gst_element_get_request_pad() has been deprecated in favour of the
+    newly-added gst_element_request_pad_simple() which does the exact
+    same thing but has a less confusing name that hopefully makes clear
+    that the function request a new pad rather than just retrieves an
+    already-existing request pad.
+
+-   gst_discoverer_info_get_tags(), which for many files returns a
+    confusing mix of stream and container tags, has been deprecated in
+    favour of the container-specific and stream-specific functions,
+    gst_discoverer_container_info_get_tags() and
+    gst_discoverer_stream_info_get_tags().
+
+-   gst_video_sink_center_rect() was deprecated in favour of the more
+    generic newly-added gst_video_center_rect().
+
+-   The GST_MEMORY_FLAG_NO_SHARE flag has been deprecated, as it tends
+    to cause problems and prevents sub-buffering. If pooling or lifetime
+    tracking is required, memories should be allocated through a custom
+    GstAllocator instead of relying on the lifetime of the buffers the
+    memories were originally attached to, which is fragile anyway.
+
+-   The GstPlayer high-level playback library is being replaced with the
+    new GstPlay library (see above). GstPlayer should be considered
+    deprecated at this point and will be marked as such in the next
+    development cycle. Applications should be ported to GstPlay.
+
+-   Gstreamer Editing Services: ges_video_transition_set_border(),
+    ges_video_transition_get_border()
+    ges_video_transition_set_inverted()
+    ges_video_transition_is_inverted() have been deprecated, use
+    ges_timeline_element_set_children_properties() instead.
 
 Miscellaneous performance, latency and memory optimisations
 
--   this section will be filled in in due course
+More video conversion fast paths
 
-Miscellaneous other changes and enhancements
+-   v210 ↔ I420, YV12, Y42B, UYVY and YUY2
+-   A420 → RGB
+
+Less jitter when waiting on the system clock
+
+-   Better system clock wait accuracy, less jitter: where available,
+    clock_nanosleep is used for higher accuracy for waits below 500
+    usecs, and waits below 2ms will first use the regular waiting system
+    and then clock_nanosleep for the remainder. The various wait
+    implementation have a latency ranging from 50 to 500+ microseconds.
+    While this is not a major issue when dealing with a low number of
+    waits per second (for ex: video), it does introduce a non-negligible
+    jitter for synchronisation of higher packet rate systems.
+
+Video decoder subframe support
 
--   this section will be filled in in due course
+-   The GstVideoDecoder base class gained API to process input at the
+    sub-frame level. That way video decoders can start decoding slices
+    before they have received the full input frame in its entirety (to
+    the extent this is supported by the codec, of course). This helps
+    with CPU utilisation and reduces latency.
 
-Tracing framework and debugging improvements
+-   This functionality is now being used in the OpenJPEG JPEG 2000
+    decoder, the FFmpeg H.264 decoder (in case of NAL-aligned input) and
+    the OpenMAX H.264/H.265 decoders (in case of NAL-aligned input).
 
--   this section will be filled in in due course
+Miscellaneous other changes and enhancements
+
+-   GstDeviceMonitor no longer fails to start just because one of the
+    device providers failed to start. That could happen for example on
+    systems where the pulseaudio device provider is installed, but
+    pulseaudio isn’t actually running but ALSA is used for audio
+    instead. In the same vein the device monitor now keeps track of
+    which providers have been started (via the new
+    gst_device_provider_is_started()) and only stops actually running
+    device providers when stopping the device monitor.
+
+-   On embedded systems it can be useful to create a registry that can
+    be shared and read by multiple processes running as different users.
+    It is now possible to set the new GST_REGISTRY_MODE environment
+    variable to specify the file mode for the registry file, which by
+    default is set to be only user readable/writable.
+
+-   GstNetClientClock will signal lost sync in case the remote time
+    resets (e.g. because device power cycles), by emitting the “synced”
+    signal with synced=FALSE parameter, so applications can take action.
+
+-   gst_value_deserialize_with_pspec() allows deserialization with a
+    hint for what the target GType should be. This allows for example
+    passing arrays of flags through the command line or
+    gst_util_set_object_arg(), eg: foo="<bar,bar+baz>".
+
+-   It’s now allowed to create an empty GstVideoOverlayComposition
+    without any rectangles by passing a NULL rectangle to
+    gst_video_overlay_composition_new(). This is useful for bindings and
+    simplifies application code in some places.
+
+Tracing framework, debugging and testing improvements
+
+-   New factories tracer to list loaded elements (and other plugin
+    features). This can be useful to collect a list of elements needed
+    for an application, which then in turn can be used to create a
+    tailored minimal GStreamer build that contains just the elements
+    needed and nothing else.
+-   New plugin-feature-loaded tracing hook for use by tracers like the
+    new factories tracer
+
+-   GstHarness: Add gst_harness_set_live() so that harnesses can be set
+    to non-live and return is-live=false in latency queries if needed.
+    Default behaviour is to always return is-live=true in latency
+    queries.
+
+-   navseek: new "hold-eos" property. When enabled, the element will
+    hold back an EOS event until the next keystroke (via navigation
+    events). This can be used to keep a video sink showing the last
+    frame of a video pipeline until a key is pressed instead of tearing
+    it down immediately on EOS.
+
+-   New fakeaudiosink element: mimics an audio sink and can be used for
+    testing and CI pipelines on systems where no audio system is
+    installed or running. It differs from fakesink in that it only
+    support audio caps and syncs to the clock by default like a normal
+    audio sink. It also implements the GstStreamVolume interface like
+    most audio sinks do.
+
+-   New videocodectestsink element for video codec conformance testing:
+    Calculates MD5 checksums for video frames and skips any padding
+    whilst doing so. Can optionally also write back the video data with
+    padding removed into a file for easy byte-by-byte comparison with
+    reference data.
 
 Tools
 
--   this section will be filled in in due course
+gst-inspect-1.0
+
+-   Can sort the list of plugins by passing --sort=name as command line
+    option
+
+gst-launch-1.0
+
+-   will now error out on top-level properties that don’t exist and
+    which were silently ignored before
+-   On Windows the high-resolution clock is enabled now, which provides
+    better clock and timer performance on Windows (see Windows section
+    below for more details).
+
+gst-play-1.0
+
+-   New --start-position command line argument to start playback from
+    the specified position
+-   Audio can be muted/unmuted in interactive mode by pressing the m
+    key.
+-   On Windows the high-resolution clock is enabled now (see Windows
+    section below for more details)
+
+gst-device-monitor-1.0
+
+-   New --include-hidden command line argument to also show “hidden”
+    device providers
+
+ges-launch-1.0
+
+-   New interactive mode that allows seeking and such. Can be disabled
+    by passing the --no-interactive argument on the command line.
+-   Option to forward tags
+-   Allow using an existing clip to determine the rendering format (both
+    topology and profile) via new --profile-from command line argument.
 
 GStreamer RTSP server
 
--   this section will be filled in in due course
+-   GstRTSPMediaFactory gained API to disable RTCP
+    (gst_rtsp_media_factory_set_enable_rtcp(), "enable-rtcp" property).
+    Previously RTCP was always allowed for all RTSP medias. With this
+    change it is possible to disable RTCP completely, no matter if the
+    client wants to do RTCP or not.
+
+-   Make a mount point of / work correctly. While not allowed by the
+    RTSP 2 spec, the RTSP 1 spec is silent on this and it is used in the
+    wild. It is now possible to use / as a mount path in
+    gst-rtsp-server, e.g. rtsp://example.com/ would work with this now.
+    Note that query/fragment parts of the URI are not necessarily
+    correctly handled, and behaviour will differ between various
+    client/server implementations; so use it if you must but don’t bug
+    us if it doesn’t work with third party clients as you’d hoped.
+
+-   multithreading fixes (races, refcounting issues, deadlocks)
+
+-   ONVIF audio backchannel fixes
+
+-   ONVIF trick mode optimisations
+
+-   rtspclientsink: new "update-sdp" signal that allows updating the SDP
+    before sending it to the server via ANNOUNCE. This can be used to
+    add additional metadata to the SDP, for example. The order and
+    number of medias must not be changed, however.
 
 GStreamer VAAPI
 
--   this section will be filled in in due course
+-   new AV1 decoder element (vaapiav1dec)
+
+-   H264 decoder: handle stereoscopic 3D video with frame packing
+    arrangement SEI messages
+
+-   H265 encoder: added Screen Content Coding extensions support
+
+-   H265 decoder: gained MAIN_444_12 profile support (decoded to
+    Y412_LE), and 4:2:2 12-bits support (decoded to Y212_LE)
+
+-   vaapipostproc: gained BT2020 color standard support
+
+-   vaapidecode: now generates caps templates dynamically at runtime in
+    order to advertise actually supported caps instead of all
+    theoretically supported caps.
+
+-   GST_VAAPI_DRM_DEVICE environment variable to force a specified DRM
+    device when a DRM display is used. It is ignored when other types of
+    displays are used. By default /dev/dri/renderD128 is used for DRM
+    display.
 
 GStreamer OMX
 
--   this section will be filled in in due course
+-   subframe support in H.264/H.265 decoders
 
 GStreamer Editing Services and NLE
 
--   this section will be filled in in due course
+-   framepositioner: new "operator" property to access blending modes in
+    the compositor
+-   timeline: Implement snapping to markers
+-   smart-mixer: Add support for d3d11compositor and glvideomixer
+-   titleclip: add "draw-shadow" child property
+-   ges:// URI support to define a timeline from a description.
+-   command-line-formatter
+    -   Add track management to timeline description
+    -   Add keyframe support
+-   ges-launch-1.0:
+    -   Add an interactive mode where we can seek etc…
+    -   Add option to forward tags
+    -   Allow using an existing clip to determine the rendering format
+        (both topology and profile) via new --profile-from command line
+        argument.
+-   Fix static build
 
 GStreamer validate
 
--   this section will be filled in in due course
+-   report: Add a way to force backtraces on reports even if not a
+    critical issue (GST_VALIDATE_ISSUE_FLAGS_FORCE_BACKTRACE)
+-   Add a flag to gst_validate_replace_variables_in_string() allow
+    defining how to resolve variables in structs
+-   Add gst_validate_bin_monitor_get_scenario() to get the bin monitor
+    scenario, which is useful for applications that use Validate
+    directly.
+-   Add an expected-values parameter to wait, message-type=XX allowing
+    more precise filtering of the message we are waiting for.
+-   Add config file support: each test can now use a config file for the
+    given media file used to test.
+-   Add support to check properties of object properties
+-   scenario: Add an "action-done" signal to signal when an action is
+    done
+-   scenario: Add a "run-command" action type
+-   scenario: Allow forcing running action on idle from scenario file
+-   scenario: Allow iterating over arrays in foreach
+-   scenario: Rename ‘interlaced’ action to ‘non-blocking’
+-   scenario: Add a non-blocking flag to the wait signal
 
 GStreamer Python Bindings
 
--   this section will be filled in in due course
+-   Fixes for Python 3.10
+-   Various build fixes
+-   at least one known breaking change caused by g-i annotation changes
+    (see below)
 
 GStreamer C# Bindings
 
--   this section will be filled in in due course
+-   Fix GstDebugGraphDetails enum
+-   Updated to latests GtkSharp
+-   Updated to include GStreamer 1.20 API
 
 GStreamer Rust Bindings and Rust Plugins
 
-The GStreamer Rust bindings are released separately with a different
-release cadence that’s tied to gtk-rs, but the latest release has
-already been updated for the upcoming new GStreamer 1.20 API.
-
-gst-plugins-rs, the module containing GStreamer plugins written in Rust,
-has also seen lots of activity with many new elements and plugins.
-
-What follows is a list of elements and plugins available in
-gst-plugins-rs, so people don’t miss out on all those potentially useful
-elements that have no C equivalent.
-
--   FIXME: add new elements
-
-Rust audio plugins
-
--   audiornnoise: New element for audio denoising which implements the
-    noise removal algorithm of the Xiph RNNoise library, in Rust
--   rsaudioecho: Port of the audioecho element from gst-plugins-good
-    rsaudioloudnorm: Live audio loudness normalization element based on
-    the FFmpeg af_loudnorm filter
--   claxondec: FLAC lossless audio codec decoder element based on the
-    pure-Rust claxon implementation
--   csoundfilter: Audio filter that can use any filter defined via the
-    Csound audio programming language
--   lewtondec: Vorbis audio decoder element based on the pure-Rust
-    lewton implementation
-
-Rust video plugins
-
--   cdgdec/cdgparse: Decoder and parser for the CD+G video codec based
-    on a pure-Rust CD+G implementation, used for example by karaoke CDs
--   cea608overlay: CEA-608 Closed Captions overlay element
--   cea608tott: CEA-608 Closed Captions to timed-text (e.g. VTT or SRT
-    subtitles) converter
--   tttocea608: CEA-608 Closed Captions from timed-text converter
--   mccenc/mccparse: MacCaption Closed Caption format encoder and parser
--   sccenc/sccparse: Scenarist Closed Caption format encoder and parser
--   dav1dec: AV1 video decoder based on the dav1d decoder implementation
-    by the VLC project
--   rav1enc: AV1 video encoder based on the fast and pure-Rust rav1e
-    encoder implementation
--   rsflvdemux: Alternative to the flvdemux FLV demuxer element from
-    gst-plugins-good, not feature-equivalent yet
--   rsgifenc/rspngenc: GIF/PNG encoder elements based on the pure-Rust
-    implementations by the image-rs project
-
-Rust text plugins
-
--   textwrap: Element for line-wrapping timed text (e.g. subtitles) for
-    better screen-fitting, including hyphenation support for some
-    languages
-
-Rust network plugins
-
--   reqwesthttpsrc: HTTP(S) source element based on the Rust
-    reqwest/hyper HTTP implementations and almost feature-equivalent
-    with the main GStreamer HTTP source souphttpsrc
--   s3src/s3sink: Source/sink element for the Amazon S3 cloud storage
--   awstranscriber: Live audio to timed text transcription element using
-    the Amazon AWS Transcribe API
-
-Generic Rust plugins
-
--   sodiumencrypter/sodiumdecrypter: Encryption/decryption element based
-    on libsodium/NaCl
--   togglerecord: Recording element that allows to pause/resume
-    recordings easily and considers keyframe boundaries
--   fallbackswitch/fallbacksrc: Elements for handling potentially
-    failing (network) sources, restarting them on errors/timeout and
-    showing a fallback stream instead
--   threadshare: Set of elements that provide alternatives for various
-    existing GStreamer elements but allow to share the streaming threads
-    between each other to reduce the number of threads
--   rsfilesrc/rsfilesink: File source/sink elements as replacements for
-    the existing filesrc/filesink elements
+-   The GStreamer Rust bindings are released separately with a different
+    release cadence that’s tied to gtk-rs, but the latest release has
+    already been updated for the upcoming new GStreamer 1.20 API (v1_20
+    feature).
+
+-   gst-plugins-rs, the module containing GStreamer plugins written in
+    Rust, has also seen lots of activity with many new elements and
+    plugins. See the New Elements section above for a list of new Rust
+    elements.
 
 Build and Dependencies
 
 -   Meson 0.59 or newer is required to build GStreamer now.
 
--   FIXME: this section will be filled in in due course
+-   The GLib requirement has been bumped to GLib 2.56 or newer (from
+    March 2018).
+
+-   The wpe plugin now requires wpe >= 2.28 and wpebackend-fdo >= 1.8
 
 Explicit opt-in required for build of certain plugins with (A)GPL dependencies
 
@@ -211,11 +1591,9 @@ even if the required dependencies are available.
 See Building plugins with (A)GPL-licensed dependencies for more details
 and a non-exhaustive list of plugins affected.
 
-gst-build: replaced by Monorepo
+gst-build: replaced by mono repository
 
--   this section will be filled in in due course
-
--   FIXME: describe + link to Monorepo FAQ
+See mono repository section above and the GStreamer mono repository FAQ.
 
 Cerbero
 
@@ -223,76 +1601,254 @@ Cerbero is a meta build system used to build GStreamer plus dependencies
 on platforms where dependencies are not readily available, such as
 Windows, Android, iOS and macOS.
 
-General improvements
+General Cerbero improvements
 
--   this section will be filled in in due course
+-   Plugin removed: libvisual
+-   New plugins: rtpmanagerbad and rist
 
-macOS / iOS
+macOS / iOS specific Cerbero improvements
 
--   this section will be filled in in due course
+-   XCode 12 support
+-   macOS OS release support is now future-proof, similar to iOS
+-   macOS Apple Silicon (ARM64) cross-compile support has been added
+-   macOS Apple Silicon (ARM64) native support is currently experimental
 
-Windows
+Windows specific Cerbero improvements
 
--   this section will be filled in in due course
+-   Visual Studio 2022 support has been added
+-   bootstrap is faster since it requires building fewer build-tools
+    recipes on Windows
+-   package is faster due to better scheduling of recipe stages and
+    elimination of unnecessary autotools regeneration
+-   The following plugins are no longer built on Windows:
+    -   a52dec (another decoder is still available in libav)
+    -   dvdread
+    -   resindvd
 
 Windows MSI installer
 
--   this section will be filled in in due course
+-   no major changes
 
-Linux
+Linux specific Cerbero improvements
 
--   this section will be filled in in due course
+-   Fedora, Debian OS release support is now more future-proof
+-   Amazon Linux 2 support has been added
 
-Android
+Android specific Cerbero improvements
 
--   this section will be filled in in due course
+-   no major changes
 
 Platform-specific changes and improvements
 
 Android
 
--   this section will be filled in in due course
+-   No major changes
 
 macOS and iOS
 
--   this section will be filled in in due course
+-   applemedia: add ProRes support to vtenc and vtdec
 
 Windows
 
--   this section will be filled in in due course
+-   On Windows the high-resolution clock is enabled now in the
+    gst-launch-1.0 and gst-play-1.0 command line tools, which provides
+    better clock and timer performance on Windows, at the cost of higher
+    power consumption. By default, without the high-resolution clock
+    enabled, the timer precision on Windows is system-dependent and may
+    be as bad as 15ms which is not good enough for many multimedia
+    applications. Developers may want to do the same in their Windows
+    applications if they think it’s a good idea for their application
+    use case, and depending on the Windows version they target. This is
+    not done automatically by GStreamer because on older Windows
+    versions (pre-Windows 10) this affects a global Windows setting and
+    also there’s a power consumption vs. performance trade-off that may
+    differ from application to application.
+
+-   dxgiscreencapsrc now supports resolution changes
+
+-   The wasapi2 audio plugin was rewritten and now has a higher rank
+    than the old wasapi plugin since it has a number of additional
+    features such as automatic stream routing, and no
+    known-but-hard-to-fix issues. The plugin is always built if the
+    Windows 10 SDK is available now.
+
+-   The wasapi device providers now detect and notify dynamic device
+    additions/removals
+
+-   d3d11screencapturesrc: new desktop capture element, including
+    GstDeviceProvider implementation to enumerate/select target monitors
+    for capture.
+
+-   Direct3D11/DXVA decoder now supports AV1 and MPEG2 codecs
+    (d3d11av1dec, d3d11mpeg2dec)
+
+-   VP9 decoding got more reliable and stable thanks to a newly written
+    codec parser
+
+-   Support for decoding interlaced H.264/AVC streams
+
+-   Hardware-accelerated video deinterlacing (d3d11deinterlace) and
+    video mixing (d3d11compositor)
+
+-   Video mixing with the Direct3D11 API (d3d11compositor)
+
+-   MediaFoundation API based hardware encoders gained the ability to
+    receive Direct3D11 textures as an input
+
+-   Seungha’s blog post “GStreamer ❤ Windows: A primer on the cool stuff
+    you’ll find in the 1.20 release” describes many of the
+    Windows-related improvements in more detail
 
 Linux
 
--   this section will be filled in in due course
+-   bluez: LDAC Bluetooth audio codec support in a2dpsink and avdtpsink,
+    as well as an LDAC RTP payloader (rtpldacpay) and an LDAC audio
+    encoder (ldacenc)
+
+-   kmssink: gained support for NV24, NV61, RGB16/BGR16 formats;
+    auto-detect NVIDIA Tegra driver
 
 Documentation improvements
 
--   this section will be filled in in due course
+-   hardware-accelerated GPU plugins will now no longer always list all
+    the element variants for all available GPUs, since those are
+    system-dependent and it’s confusing for users to see those in the
+    documentation just because the GStreamer developer who generated the
+    docs had multiple GPUs to play with at the time. Instead just show
+    the default elements.
 
-Possibly Breaking Changes
+Possibly Breaking and Other Noteworthy Behavioural Changes
+
+-   gst_parse_launch(), gst_parse_bin_from_description() and friends
+    will now error out when setting properties that don’t exist on
+    top-level bins. They were silently ignored before.
+
+-   The GstWebRTC library does not expose any objects anymore with
+    public fields. Instead properties have been added to replace that
+    functionality. If you are accessing such fields in your application,
+    switch to the corresponding properties.
 
--   FIXME: this section will be filled in in due course
--   MPEG-TS SCTE-35 API changes (FIXME: flesh out)
--   gst_parse_launch() and friends now error out on non-existing
-    properties on top-level bins where they would silently fail and
-    ignore those before.
 -   playbin and uridecodebin now emit the source-setup signal before the
     element is added to the bin and linked so that the source element is
     already configured before any scheduling query comes in, which is
-    useful for elements such as appsrc or giostreamsrc. (Merge Request)
+    useful for elements such as appsrc or giostreamsrc.
+
+-   The source element inside urisourcebin (used inside uridecodebin3
+    which is used inside playbin3) is no longer called "source". This
+    shouldn’t affect anyone hopefully, because there’s a "setup-source"
+    signal to configure the source element and no one should rely on
+    names of internal elements anyway.
+
+-   The vp8enc element now expects bps (bits per second) for the
+    "temporal-scalability-target-bitrate" property, which is consistent
+    with the "target-bitrate" property. Since additional configuration
+    is required with modern libvpx to make temporal scaling work anyway,
+    chances are that very few people will have been using this property
+
+-   vp8enc and vp9enc now default to “good quality” for the "deadline"
+    property rather then “best quality”. Having the deadline set to best
+    quality causes the encoder to be absurdly slow, most real-life users
+    will want the good quality tradeoff instead.
+
+-   The experimental GstTranscoder library API in gst-plugins-bad was
+    changed from a GObject signal-based notification mechanism to a
+    GstBus/message-based mechanism akin to GstPlayer/GstPlay.
+
+-   MPEG-TS SCTE-35 API: semantic change for SCTE-35 splice commands:
+    timestamps passed by the application should be in running time now,
+    since users of the API can’t really be expected to predict the local
+    PTS of the muxer.
+
+-   The GstContext used by souphttpsrc to share the session between
+    multiple element instances has changed. Previously it provided
+    direct access to the internal SoupSession object, now it only
+    provides access to an opaque, internal type. This change is
+    necessary because SoupSession is not thread-safe at all and can’t be
+    shared safely between arbitrary external code and souphttpsrc.
+
+-   Python bindings: GObject-introspection related Annotation fixes have
+    led to a case of a GstVideo.VideoInfo-related function signature
+    changing in the Python bindings (possibly one or two other cases
+    too). This is for a function that should never have been exposed in
+    the first place though, so the bindings are being updated to throw
+    an exception in that case, and the correct replacement API has been
+    added in form of an override.
 
 Known Issues
 
--   this section will be filled in in due course
-
--   There are a couple of known WebRTC-related regressions/blockers:
-
-    -   webrtc: DTLS setup with Chrome is broken
-    -   webrtcbin: First keyframe is usually lost
+-   nothing in particular at this point (but also see possibly breaking
+    changes section above)
 
 Contributors
 
--   this section will be filled in in due course
+Aaron Boxer, Adam Leppky, Adam Williamson, Alba Mendez, Alejandro
+González, Aleksandr Slobodeniuk, Alexander Vandenbulcke, Alex Ashley,
+Alicia Boya García, Andika Triwidada, Andoni Morales Alastruey, Andrew
+Wesie, Andrey Moiseev, Antonio Ospite, Antonio Rojas, Arthur Crippa
+Búrigo, Arun Raghavan, Ashley Brighthope, Axel Kellermann, Baek, Bastien
+Nocera, Bastien Reboulet, Benjamin Gaignard, Bing Song, Binh Truong,
+Biswapriyo Nath, Brad Hards, Brad Smith, Brady J. Garvin, Branko
+Subasic, Camilo Celis Guzman, Chris Bass, ChrisDuncanAnyvision, Chris
+White, Corentin Damman, Daniel Almeida, Daniel Knobe, Daniel Stone,
+david, David Fernandez, David Keijser, David Phung, Devarsh Thakkar,
+Dinesh Manajipet, Dmitry Samoylov, Dmitry Shusharin, Dominique Martinet,
+Doug Nazar, Ederson de Souza, Edward Hervey, Emmanuel Gil Peyrot,
+Enrique Ocaña González, Ezequiel Garcia, Fabian Orccon, Fabrice
+Fontaine, Fernando Jimenez Moreno, Florian Karydes, Francisco Javier
+Velázquez-García, François Laignel, Frederich Munch, Fredrik Pålsson,
+George Kiagiadakis, Georg Lippitsch, Göran Jönsson, Guido Günther,
+Guillaume Desmottes, Guiqin Zou, Haakon Sporsheim, Haelwenn (lanodan)
+Monnier, Haihao Xiang, Haihua Hu, Havard Graff, He Junyan, Helmut
+Januschka, Henry Wilkes, Hosang Lee, Hou Qi, Ignacio Casal Quinteiro,
+Igor Kovalenko, Ilya Kreymer, Imanol Fernandez, Jacek Tomaszewski, Jade
+Macho, Jakub Adam, Jakub Janků, Jan Alexander Steffens (heftig), Jan
+Schmidt, Jason Carrete, Jason Pereira, Jay Douglass, Jeongki Kim, Jérôme
+Laheurte, Jimmi Holst Christensen, Johan Sternerup, John Hassell, John
+Lindgren, John-Mark Bell, Jonathan Matthew, Jordan Petridis, Jose
+Quaresma, Julian Bouzas, Julien, Kai Uwe Broulik, Kasper Steensig
+Jensen, Kellermann Axel, Kevin Song, Khem Raj, Knut Inge Hvidsten, Knut
+Saastad, Kristofer Björkström, Lars Lundqvist, Lawrence Troup, Lim Siew
+Hoon, Lucas Stach, Ludvig Rappe, Luis Paulo Fernandes de Barros, Luke
+Yelavich, Mads Buvik Sandvei, Marc Leeman, Marco Felsch, Marek Vasut,
+Marian Cichy, Marijn Suijten, Marius Vlad, Markus Ebner, Mart Raudsepp,
+Matej Knopp, Mathieu Duponchelle, Matthew Waters, Matthieu De Beule,
+Mengkejiergeli Ba, Michael de Gans, Michael Olbrich, Michael Tretter,
+Michal Dzik, Miguel Paris, Mikhail Fludkov, mkba, Nazar Mokrynskyi,
+Nicholas Jackson, Nicola Murino, Nicolas Dufresne, Niklas Hambüchen,
+Nikolay Sivov, Nirbheek Chauhan, Olivier Blin, Olivier Crete, Olivier
+Crête, Paul Goulpié, Per Förlin, Peter Boba, P H, Philippe Normand,
+Philipp Zabel, Pieter Willem Jordaan, Piotrek Brzeziński, Rafał
+Dzięgiel, Rafostar, raghavendra, Raghavendra, Raju Babannavar, Raleigh
+Littles III, Randy Li, Randy Li (ayaka), Ratchanan Srirattanamet, Raul
+Tambre, reed.lawrence, Ricky Tang, Robert Rosengren, Robert Swain, Robin
+Burchell, Roman Sivriver, R S Nikhil Krishna, Ruben Gonzalez, Ruslan
+Khamidullin, Sanchayan Maity, Scott Moreau, Sebastian Dröge, Sergei
+Kovalev, Seungha Yang, Sid Sethupathi, sohwan.park, Sonny Piers, Staz M,
+Stefan Brüns, Stéphane Cerveau, Stephan Hesse, Stian Selnes, Stirling
+Westrup, Théo MAILLART, Thibault Saunier, Tim, Timo Wischer, Tim-Philipp
+Müller, Tim Schneider, Tobias Ronge, Tom Schoonjans, Tulio Beloqui,
+tyler-aicradle, U. Artie Eoff, Ung, Val Doroshchuk, VaL Doroshchuk,
+Víctor Manuel Jáquez Leal, Vivek R, Vivia Nikolaidou, Vivienne
+Watermeier, Vladimir Menshakov, Will Miller, Wim Taymans, Xabier
+Rodriguez Calvar, Xavier Claessens, Xℹ Ruoyao, Yacine Bandou, Yinhang
+Liu, youngh.lee, youngsoo.lee, yychao, Zebediah Figura, Zhang yuankun,
+Zhang Yuankun, Zhao, Zhao Zhili, , Aleksandar Topic, Antonio Ospite,
+Bastien Nocera, Benjamin Gaignard, Brad Hards, Carlos Falgueras García,
+Célestin Marot, Corentin Damman, Corentin Noël, Daniel Almeida, Daniel
+Knobe, Danny Smith, Dave Piché, Dmitry Osipenko, Fabrice Fontaine,
+fjmax, Florian Zwoch, Guillaume Desmottes, Haihua Hu, Heinrich Kruger,
+He Junyan, Jakub Adam, James Cowgill, Jan Alexander Steffens (heftig),
+Jean Felder, Jeongki Kim, Jiri Uncovsky, Joe Todd, Jordan Petridis,
+Krystian Wojtas, Marc-André Lureau, Marcin Kolny, Marc Leeman, Mark
+Nauwelaerts, Martin Reboredo, Mathieu Duponchelle, Matthew Waters,
+Mengkejiergeli Ba, Michael Gruner, Nicolas Dufresne, Nirbheek Chauhan,
+Olivier Crête, Philippe Normand, Rafał Dzięgiel, Ralf Sippl, Robert
+Mader, Sanchayan Maity, Sangchul Lee, Sebastian Dröge, Seungha Yang,
+Stéphane Cerveau, Teh Yule Kim, Thibault Saunier, Thomas Klausner, Timo
+Wischer, Tim-Philipp Müller, Tobias Reineke, Tomasz Andrzejak, Trung Do,
+Tyler Compton, Ung, Víctor Manuel Jáquez Leal, Vivia Nikolaidou, Wim
+Taymans, wngecn, Wonchul Lee, wuchang li, Xavier Claessens, Xi Ruoyao,
+Yoshiharu Hirose, Zhao,
 
 … and many others who have contributed bug reports, translations, sent
 suggestions or helped testing.
@@ -307,7 +1863,7 @@ the git 1.20 branch, which will be a stable branch.
 
 1.20.0
 
-1.20.0 is scheduled to be released around October/November 2021.
+1.20.0 is scheduled to be released around early February 2022.
 
 Schedule for 1.22
 
@@ -315,7 +1871,9 @@ Our next major feature release will be 1.22, and 1.21 will be the
 unstable development version leading up to the stable 1.22 release. The
 development of 1.21/1.22 will happen in the git main branch.
 
-The plan for the 1.22 development cycle is yet to be confirmed.
+The plan for the 1.22 development cycle is yet to be confirmed. Assuming
+no major project-wide reorganisations in the 1.22 cycle we might try and
+aim for a release around August 2022.
 
 1.22 will be backwards-compatible to the stable 1.20, 1.18, 1.16, 1.14,
 1.12, 1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series.
@@ -323,6 +1881,7 @@ The plan for the 1.22 development cycle is yet to be confirmed.
 ------------------------------------------------------------------------
 
 These release notes have been prepared by Tim-Philipp Müller with
-contributions from …
+contributions from Matthew Waters, Nicolas Dufresne, Nirbheek Chauhan,
+Sebastian Dröge and Seungha Yang.
 
 License: CC BY-SA 4.0
diff --git a/subprojects/gst-plugins-good/RELEASE b/subprojects/gst-plugins-good/RELEASE
index dc0efc3bbe..796bde464e 100644
--- a/subprojects/gst-plugins-good/RELEASE
+++ b/subprojects/gst-plugins-good/RELEASE
@@ -1,4 +1,4 @@
-This is GStreamer gst-plugins-good 1.19.3.
+This is GStreamer gst-plugins-good 1.19.90.
 
 GStreamer 1.19 is the development branch leading up to the next major
 stable version which will be 1.20.
diff --git a/subprojects/gst-plugins-good/docs/gst_plugins_cache.json b/subprojects/gst-plugins-good/docs/gst_plugins_cache.json
index bd7c939aaa..a9ba5d36b5 100644
--- a/subprojects/gst-plugins-good/docs/gst_plugins_cache.json
+++ b/subprojects/gst-plugins-good/docs/gst_plugins_cache.json
@@ -6607,7 +6607,7 @@
                         "construct": false,
                         "construct-only": false,
                         "controllable": false,
-                        "default": "GStreamer 1.19.3.1 FLV muxer",
+                        "default": "GStreamer 1.19.90 FLV muxer",
                         "mutable": "null",
                         "readable": true,
                         "type": "gchararray",
@@ -6619,7 +6619,7 @@
                         "construct": false,
                         "construct-only": false,
                         "controllable": false,
-                        "default": "GStreamer 1.19.3.1 FLV muxer",
+                        "default": "GStreamer 1.19.90 FLV muxer",
                         "mutable": "null",
                         "readable": true,
                         "type": "gchararray",
@@ -20294,7 +20294,7 @@
                         "construct": false,
                         "construct-only": false,
                         "controllable": false,
-                        "default": "GStreamer/1.19.3.1",
+                        "default": "GStreamer/1.19.90",
                         "mutable": "null",
                         "readable": true,
                         "type": "gchararray",
@@ -22251,7 +22251,7 @@
                         "construct": false,
                         "construct-only": false,
                         "controllable": false,
-                        "default": "GStreamer souphttpsrc 1.19.3.1 ",
+                        "default": "GStreamer souphttpsrc 1.19.90 ",
                         "mutable": "null",
                         "readable": true,
                         "type": "gchararray",
diff --git a/subprojects/gst-plugins-good/gst-plugins-good.doap b/subprojects/gst-plugins-good/gst-plugins-good.doap
index 076e4f646d..a807027350 100644
--- a/subprojects/gst-plugins-good/gst-plugins-good.doap
+++ b/subprojects/gst-plugins-good/gst-plugins-good.doap
@@ -34,6 +34,16 @@ the plug-in code, LGPL or LGPL-compatible for the supporting library).
 
  <release>
   <Version>
+   <revision>1.19.90</revision>
+   <branch>main</branch>
+   <name></name>
+   <created>2022-01-28</created>
+   <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-plugins-good/gst-plugins-good-1.19.90.tar.xz" />
+  </Version>
+ </release>
+
+ <release>
+  <Version>
    <revision>1.19.3</revision>
    <branch>main</branch>
    <name></name>
diff --git a/subprojects/gst-plugins-good/meson.build b/subprojects/gst-plugins-good/meson.build
index 2097f9764a..13b8f10a4f 100644
--- a/subprojects/gst-plugins-good/meson.build
+++ b/subprojects/gst-plugins-good/meson.build
@@ -1,5 +1,5 @@
 project('gst-plugins-good', 'c',
-  version : '1.19.3.1',
+  version : '1.19.90',
   meson_version : '>= 0.59',
   default_options : [ 'warning_level=1',
                       'buildtype=debugoptimized' ])
diff --git a/subprojects/gst-plugins-ugly/ChangeLog b/subprojects/gst-plugins-ugly/ChangeLog
index 9cb837bb23..588a12811c 100644
--- a/subprojects/gst-plugins-ugly/ChangeLog
+++ b/subprojects/gst-plugins-ugly/ChangeLog
@@ -1,3 +1,18 @@
+=== release 1.19.90 ===
+
+2022-01-28 14:28:35 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* NEWS:
+	* RELEASE:
+	* gst-plugins-ugly.doap:
+	* meson.build:
+	  Release 1.19.90
+
+2022-01-28 14:28:28 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* ChangeLog:
+	  Update ChangeLogs for 1.19.90
+
 2022-01-27 01:39:28 +0000  Tim-Philipp Müller <tim@centricular.com>
 
 	* po/LINGUAS:
diff --git a/subprojects/gst-plugins-ugly/NEWS b/subprojects/gst-plugins-ugly/NEWS
index e9f5227c15..1a512c1848 100644
--- a/subprojects/gst-plugins-ugly/NEWS
+++ b/subprojects/gst-plugins-ugly/NEWS
@@ -1,23 +1,19 @@
 GStreamer 1.20 Release Notes
 
 GStreamer 1.20 has not been released yet. It is scheduled for release in
-November 2021.
+late January / early February 2022.
 
 1.19.x is the unstable development version that is being developed in
-the git main branch and which will eventually result in 1.20, and 1.19.3
-is the current development release in that series
-
-It is expected that feature freeze will be in early October 2021,
-followed by one or two 1.19.9x pre-releases and the new 1.20 stable
-release around October/November 2021.
+the git main branch and which will eventually result in 1.20, and
+1.19.90 is the first release candidate in that series (1.20rc1).
 
 1.20 will be backwards-compatible to the stable 1.18, 1.16, 1.14, 1.12,
-1.10, 1.8, 1.6,, 1.4, 1.2 and 1.0 release series.
+1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series.
 
 See https://gstreamer.freedesktop.org/releases/1.20/ for the latest
 version of this document.
 
-Last updated: Monday 1 November 2021, 01:00 UTC (log)
+Last updated: Wednesday 26 January 2022, 01:00 UTC (log)
 
 Introduction
 
@@ -30,25 +26,971 @@ fixes and other improvements.
 
 Highlights
 
--   this section will be completed in due course
+-   Development in GitLab was switched to a single git repository
+    containing all the modules
+-   GstPlay: new high-level playback library, replaces GstPlayer
+-   WebM Alpha decoding support
+-   Encoding profiles can now be tweaked with additional
+    application-specified element properties
+-   Compositor: multi-threaded video conversion and mixing
+-   RTP header extensions: unified support in RTP depayloader and
+    payloader base classes
+-   SMPTE 2022-1 2-D Forward Error Correction support
+-   Smart encoding (passthrough) support for VP8, VP9, H.265 in
+    encodebin and transcodebin
+-   Runtime compatibility support for libsoup2 and libsoup3 (libsoup3
+    support experimental)
+-   Video decoder subframe support
+-   Video decoder automatic packet-loss, data corruption, and keyframe
+    request handling for RTP / WebRTC / RTSP
+-   MP4 and Matroska muxers now support profile/level/resolution changes
+    for H264/H265 input streams (i.e. codec data changing on the fly)
+-   MP4 muxing mode that initially creates a fragmented mp4 which is
+    converted to a regular mp4 on EOS
+-   Audio support for the WebKit Port for Embedded (WPE) web page source
+    element
+-   CUDA based video color space convert and rescale elements and
+    upload/download elements
+-   NVIDIA memory:NVMM support for OpenGL glupload and gldownload
+    elements
+-   Many WebRTC improvements
+-   The new VA-API plugin implemention fleshed out with more decoders
+    and new postproc elements
+-   AppSink API to retrieve events in addition to buffers and buffer
+    lists
+-   AppSrc gained more configuration options for the internal queue
+    (leakiness, limits in buffers and time, getters to read current
+    levels)
+-   Updated Rust bindings and many new Rust plugins
+-   Improved support for custom minimal GStreamer builds
+-   Support build against FFmpeg 5.0
+-   Linux Stateless CODEC support gained MPEG2 and VP9
+-   Windows Direct3D11/DXVA decoder gained AV1 and MPEG2 support
+-   Lots of new plugins, features, performance improvements and bug
+    fixes
 
 Major new features and changes
 
 Noteworthy new features and API
 
--   this section will be filled in in due course
+-   gst_element_get_request_pad() has been deprecated in favour of the
+    newly-added gst_element_request_pad_simple() which does the exact
+    same thing but has a less confusing name that hopefully makes clear
+    that the function request a new pad rather than just retrieves an
+    already-existing request pad.
+
+Development in GitLab was switched to a single git repository containing all the modules
+
+The GStreamer multimedia framework is a set of libraries and plugins
+split into a number of distinct modules which are released independently
+and which have so far been developed in separate git repositories in
+freedesktop.org GitLab.
+
+In addition to these separate git repositories there was a gst-build
+module that would use the Meson build systems’s subproject feature to
+download each individual module and then build everything in one go. It
+would also provide an uninstalled development environment that made it
+easy to work on GStreamer and use or test versions other than the
+system-installed GStreamer version.
+
+All of these modules have now (as of 28 September 2021) been merged into
+a single git repository (“Mono repository” or “monorepo”) which should
+simplify development workflows and continuous integration, especially
+where changes need to be made to multiple modules at once.
+
+This mono repository merge will primarily affect GStreamer developers
+and contributors and anyone who has workflows based on the GStreamer git
+repositories.
+
+The Rust bindings and Rust plugins modules have not been merged into the
+mono repository at this time because they follow a different release
+cycle.
+
+The mono repository lives in the existing GStreamer core git repository
+in GitLab in the new main branch and all future development will happen
+on this branch.
+
+Modules will continue to be released as separate tarballs.
+
+For more details, please see the GStreamer mono repository FAQ.
+
+GstPlay: new high-level playback library replacing GstPlayer
+
+-   GstPlay is a new high-level playback library that replaces the older
+    GstPlayer API. It is basically the same API as GstPlayer but
+    refactored to use bus messages for application notifications instead
+    of GObject signals. There is still a signal adapter object for those
+    who prefer signals. Since the existing GstPlayer API is already in
+    use in various applications, it didn’t seem like a good idea to
+    break it entirely. Instead a new API was added, and it is expected
+    that this new GstPlay API will be moved to gst-plugins-base in
+    future.
+
+-   The existing GstPlayer API is scheduled for deprecation and will be
+    removed at some point in the future (e.g. in GStreamer 1.24), so
+    application developers are urged to migrate to the new GstPlay API
+    at their earliest convenience.
+
+WebM alpha decoding
+
+-   Implement WebM alpha decoding (VP8/VP9 with alpha), which required
+    support and additions in various places. This is supported both with
+    software decoders and hardware-accelerated decoders.
+
+-   VP8/VP9 don’t support alpha components natively in the codec, so the
+    way this is implemented in WebM is by encoding the alpha plane with
+    transparency data as a separate VP8/VP9 stream. Inside the WebM
+    container (a variant of Matroska) this is coded as a single video
+    track with the “normal” VP8/VP9 video data making up the main video
+    data and each frame of video having an encoded alpha frame attached
+    to it as extra data ("BlockAdditional").
+
+-   matroskademux has been extended extract this per-frame alpha side
+    data and attach it in form of a GstVideoCodecAlphaMeta to the
+    regular video buffers. Note that this new meta is specific to this
+    VP8/VP9 alpha support and can’t be used to just add alpha support to
+    other codecs that don’t support it. Lastly, matroskademux also
+    advertises the fact that the streams contain alpha in the caps.
+
+-   The new codecalpha plugin contains various bits of infrastructure to
+    support autoplugging and debugging:
+
+    -   codecalphademux splits out the alpha stream from the metas on
+        the regular VP8/VP9 buffers
+    -   alphacombine takes two decoded raw video streams (one alpha, one
+        the regular video) and combines it into a video stream with
+        alpha
+    -   vp8alphadecodebin + vp9alphadecodebin are wrapper bins that use
+        the regular vp8dec and vp9dec software decoders to decode
+        regular and alpha streams and combine them again. To decodebin
+        these look like regular decoders which ju
+    -   The V4L2 CODEC plugin has stateless VP8/VP9 decoders that can
+        decode both alpha and non-alpha stream with a single decoder
+        instance
+
+-   A new AV12 video format was added which is basically NV12 with an
+    alpha plane, which is more convenient for many hardware-accelerated
+    decoders.
+
+-   Watch Nicolas Dufresne’s LCA 2022 talk “Bringing WebM Alpha support
+    to GStreamer” for all the details and a demo.
+
+RTP Header Extensions Base Class and Automatic Header Extension Handling in RTP Payloaders and Depayloaders
+
+-   RTP Header Extensions are specified in RFC 5285 and provide a way to
+    add small pieces of data to RTP packets in between the RTP header
+    and the RTP payload. This is often used for per-frame metadata,
+    extended timestamps or other application-specific extra data. There
+    are several commonly-used extensions specified in various RFCs, but
+    senders are free to put any kind of data in there, as long as sender
+    and receiver both know what that data is. Receivers that don’t know
+    about the header extensions will just skip the extra data without
+    ever looking at it. These header extensions can often be combined
+    with any kind of payload format, so may need to be supported by many
+    RTP payloader and depayloader elements.
+
+-   Inserting and extracting RTP header extension data has so far been a
+    bit inconvenient in GStreamer: There are functions to add and
+    retrieve RTP header extension data from RTP packets, but nothing
+    works automatically, even for common extensions. People would have
+    to do the insertion/extraction either in custom elements
+    before/after the RTP payloader/depayloader, or inside pad probes,
+    which isn’t very nice.
+
+-   This release adds various pieces of new infrastructure for generic
+    RTP header extension handling, as well as some implementations for
+    common extensions:
+
+    -   GstRTPHeaderExtension is a new helper base class for reading and
+        writing RTP header extensions. Nominally this subclasses
+        GstElement, but only so these extensions are stored in the
+        registry where they can be looked up by URI or name. They don’t
+        have pads and don’t get added to the pipeline graph as an
+        element.
+
+    -   "add-extension" and "clear-extension" action signals on RTP
+        payloaders and depayloaders for manual extension management
+
+    -   The "request-extension" signal will be emitted if an extension
+        is encountered that requires explicit mapping by the application
+
+    -   new "auto-header-extension" property on RTP payloaders and
+        depayloaders for automatic handling of known header extensions.
+        This is enabled by default. The extensions must be signalled via
+        caps / SDP.
+
+    -   RTP header extension implementations:
+
+        -   rtphdrextclientaudiolevel: Client-to-Mixer Audio Level
+            Indication (RFC 6464) (also see below)
+        -   rtphdrextcolorspace: Color Space extension, extends RTP
+            packets with color space and high dynamic range (HDR)
+            information
+        -   rtphdrexttwcc: Transport Wide Congestion Control support
+
+-   gst_rtp_buffer_remove_extension_data() is a new helper function to
+    remove an RTP header extension from an RTP buffer
+
+-   The existing gst_rtp_buffer_set_extension_data() now also supports
+    shrinking the extension data in size
+
+AppSink and AppSrc improvements
+
+-   appsink: new API to pull events out of appsink in addition to
+    buffers and buffer lists.
+
+    There was previously no way for users to receive incoming events
+    from appsink properly serialised with the data flow, even if they
+    are serialised events. The reason for that is that the only way to
+    intercept events was via a pad probe on the appsink sink pad, but
+    there is also internal queuing inside of appsink, so it’s difficult
+    to ascertain the right order of everything in all cases.
+
+    There is now a new "new-serialized-event" signal which will be
+    emitted when there’s a new event pending (just like the existing
+    "new-sample" signal). The "emit-signals" property must be set to
+    TRUE in order to activate this (but it’s also fine to just pull from
+    the application thread without using the signals).
+
+    gst_app_sink_pull_object() and gst_app_sink_try_pull_object() can be
+    used to pull out either an event or a new sample carrying a buffer
+    or buffer list, whatever is next in the queue.
+
+    EOS events will be filtered and will not be returned. EOS handling
+    can be done the ususal way, same as with _pull_sample().
+
+-   appsrc: allow configuration of internal queue limits in time and
+    buffers and add leaky mode.
+
+    There is internal queuing inside appsrc so the application thread
+    can push data into the element which will then be picked up by the
+    source element’s streaming thread and pushed into the pipeline from
+    that streaming thread. This queue is unlimited by default and until
+    now it was only possible to set a maximum size limit in bytes. When
+    that byte limit is reached, the pushing thread (application thread)
+    would be blocked until more space becomes available.
+
+    A limit in bytes is not particularly useful for many use cases, so
+    now it is possible to also configure limits in time and buffers
+    using the new "max-time" and "max-buffers" properties. Of course
+    there are also matching new read-only"current-level-buffers" and
+    "current-level-time properties" properties to query the current fill
+    level of the internal queue in time and buffers.
+
+    And as if that wasn’t enough the internal queue can also be
+    configured as leaky using the new "leaky-type" property. That way
+    when the queue is full the application thread won’t be blocked when
+    it tries to push in more data, but instead either the new buffer
+    will be dropped or the oldest data in the queue will be dropped.
+
+Better string serialization of nested GstCaps and GstStructures
+
+-   New string serialisation format for structs and caps that can handle
+    nested structs and caps properly by using brackets to delimit nested
+    items (e.g. some-struct, some-field=[nested-struct, nested=true]).
+    Unlike the default format the new variant can also support more than
+    one level of nesting. For backwards-compatibility reasons the old
+    format is still output by default when serialising caps and structs
+    using the existing API. The new functions gst_caps_serialize() and
+    gst_structure_serialize() can be used to output strings in the new
+    format.
+
+Convenience API for custom GstMetas
+
+-   New convenience API to register and create custom GstMetas:
+    gst_meta_register_custom() and gst_buffer_add_custom_meta(). Such
+    custom meta is backed by a GstStructure and does not require that
+    users of the API expose their GstMeta implementation as public API
+    for other components to make use of it. In addition, it provides a
+    simpler interface by ignoring the impl vs. api distinction that the
+    regular API exposes. This new API is meant to be the meta
+    counterpart to custom events and messages, and to be more convenient
+    than the lower-level API when the absolute best performance isn’t a
+    requirement. The reason it’s less performant than a “proper” meta is
+    that a proper meta is just a C struct in the end whereas this goes
+    through the GstStructure API which has a bit more overhead, which
+    for most scenarios is negligible however. This new API is useful for
+    experimentation or proprietary metas, but also has some limitations:
+    it can only be used if there’s a single producer of these metas;
+    it’s not allowed to register the same custom meta multiple times or
+    from multiple places.
+
+Additional Element Properties on Encoding Profiles
+
+-   GstEncodingProfile: The new "element-properties" and
+    gst_encoding_profile_set_element_properties() API allows
+    applications to set additional element properties on encoding
+    profiles to configure muxers and encoders. So far the encoding
+    profile template was the only place where this could be specified,
+    but often what applications want to do is take a ready-made encoding
+    profile shipped by GStreamer or the application and then tweak the
+    settings on top of that, which is now possible with this API. Since
+    applications can’t always know in advance what encoder element will
+    be used in the end, it’s even possible to specify properties on a
+    per-element basis.
+
+    Encoding Profiles are used in the encodebin, transcodebin and
+    camerabin elements and APIs to configure output formats (containers
+    and elementary streams).
+
+Audio Level Indication Meta for RFC 6464
+
+-   New GstAudioLevelMeta containing Audio Level Indication as per RFC
+    6464
+
+-   The level element has been updated to add GstAudioLevelMeta on
+    buffers if the "audio-level-meta" property is set to TRUE. This can
+    then in turn be picked up by RTP payloaders to signal the audio
+    level to receivers through RTP header extensions (see above).
+
+-   New Client-to-Mixer Audio Level Indication (RFC6464) RTP Header
+    Extension which should be automatically created and used by RTP
+    payloaders and depayloaders if their "auto-header-extension"
+    property is enabled and if the extension is part of the RTP caps.
+
+Automatic packet loss, data corruption and keyframe request handling for video decoders
+
+-   The GstVideoDecoder base class has gained various new APIs to
+    automatically handle packet loss and data corruption better by
+    default, especially in RTP, RTSP and WebRTC streaming scenarios, and
+    to give subclasses more control about how they want to handle
+    missing data:
+
+    -   Video decoder subclasses can mark output frames as corrupted via
+        the new GST_VIDEO_CODEC_FRAME_FLAG_CORRUPTED flag
+
+    -   A new "discard-corrupted-frames" property allows applications to
+        configure decoders so that corrupted frames are directly
+        discarded instead of being forwarded inside the pipeline. This
+        is a replacement for the "output-corrupt" property of the FFmpeg
+        decoders.
+
+    -   RTP depayloaders can now signal to decoders that data is missing
+        when sending GAP events for lost packets. GAP events can be sent
+        for various reason in a GStreamer pipeline. Often they are just
+        used to let downstream elements know that there isn’t a buffer
+        available at the moment, so downstream elements can move on
+        instead of waiting for one. They are also sent by RTP
+        depayloaders in the case that packets are missing, however, and
+        so far a decoder was not able to differentiate the two cases.
+        This has been remedied now: GAP events can be decorated with
+        gst_event_set_gap_flags() and GST_GAP_FLAG_MISSING_DATA to let
+        decoders now what happened, and decoders can then use that in
+        some cases to handle missing data better.
+
+    -   The GstVideoDecoder::handle_missing_data vfunc was added to
+        inform subclasses about packet loss or missing data and let them
+        handle it in their own way if they like.
+
+    -   gst_video_decoder_set_needs_sync_point() lets subclasses signal
+        that they need the stream to start with a sync point. If
+        enabled, the base class will discard all non-sync point frames
+        in the beginning and after a flush and does not pass them to the
+        subclass. Furthermore, if the first frame is not a sync point,
+        the base class will try and request a sync frame from upstream
+        by sending a force-key-unit event (see next items).
+
+    -   New "automatic-request-sync-points" and
+        "automatic-request-sync-point-flags" properties to automatically
+        request sync points when needed, e.g. on packet loss or if the
+        first frame is not a keyframe. Applications may want to enable
+        this on decoders operating in e.g. RTP/WebRTC/RTSP receiver
+        pipelines.
+
+    -   The new "min-force-key-unit-interval" property can be used to
+        ensure there’s a minimal interval between keyframe requests to
+        upstream (and/or the sender) and we’re not flooding the sender
+        with key unit requests.
+
+    -   gst_video_decoder_request_sync_point() allows subclasses to
+        request a new sync point (e.g. if they choose to do their own
+        missing data handling). This will still honour the
+        "min-force-key-unit-interval" property if set.
+
+Improved support for custom minimal GStreamer builds
+
+-   Element registration and registration of other plugin features
+    inside plugin init functions has been improved in order to
+    facilitate minimal custom GStreamer builds.
+
+-   A number of new macros have been added to declare and create
+    per-element and per-pluginfeature register functions in all plugins,
+    and then call those from the per-plugin plugin_init functions:
+
+    -   GST_ELEMENT_REGISTER_DEFINE,
+        GST_DEVICE_PROVIDER_REGISTER_DEFINE,
+        GST_DYNAMIC_TYPE_REGISTER_DEFINE, GST_TYPE_FIND_REGISTER_DEFINE
+        for the actual registration call with GStreamer
+    -   GST_ELEMENT_REGISTER, GST_DEVICE_PROVIDER_REGISTER,
+        GST_DYNAMIC_TYPE_REGISTER, GST_PLUGIN_STATIC_REGISTER,
+        GST_TYPE_FIND_REGISTER to call the registration function defined
+        by the REGISTER_DEFINE macro
+    -   GST_ELEMENT_REGISTER_DECLARE,
+        GST_DEVICE_PROVIDER_REGISTER_DECLARE,
+        GST_DYNAMIC_TYPE_REGISTER_DECLARE,
+        GST_TYPE_FIND_REGISTER_DECLARE to declare the registration
+        function defined by the REGISTER_DEFINE macro
+    -   and various variants for advanced use cases.
+
+-   This means that applications can call the per-element and
+    per-pluginfeature registration functions for only the elements they
+    need instead of registering plugins as a whole with all kinds of
+    elements that may not be required (e.g. encoder and decoder instead
+    of just decoder). In case of static linking all unused functions and
+    their dependencies would be removed in this case by the linker,
+    which helps minimise binary size for custom builds.
+
+-   gst_init() will automatically call a gst_init_static_plugins()
+    function if one exists.
+
+-   See the GStreamer static build documentation and Stéphane’s blog
+    post Generate a minimal GStreamer build, tailored to your needs for
+    more details.
 
 New elements
 
--   this section will be filled in in due course
+-   New aesdec and aesenc elements for AES encryption and decryption in
+    a custom format.
+
+-   New encodebin2 element with dynamic/sometimes source pads in order
+    to support the option of doing the muxing outside of encodebin,
+    e.g. in combination with a splitmuxsink.
+
+-   New fakeaudiosink and videocodectestsink elements for testing and
+    debugging (see below for more details)
+
+-   rtpisacpay, rtpisacdepay: new RTP payloader and depayloader for iSAC
+    audio codec
+
+-   rtpst2022-1-fecdec, rtpst2022-1-fecenc: new elements providing SMPTE
+    2022-1 2-D Forward Error Correction. More details in Mathieu’s blog
+    post.
+
+-   isac: new plugin wrapping the Internet Speech Audio Codec reference
+    encoder and decoder from the WebRTC project.
+
+-   asio: plugin for Steinberg ASIO (Audio Streaming Input/Output) API
+
+-   gssrc, gssink: add source and sink for Google Cloud Storage
+
+-   onnx: new plugin to apply ONNX neural network models to video
+
+-   openaptx: aptX and aptX-HD codecs using libopenaptx (v0.2.0)
+
+-   qroverlay, debugqroverlay: new elements that allows overlaying data
+    on top of video in form of a QR code
+
+-   cvtracker: new OpenCV-based tracker element
+
+-   av1parse, vp9parse: new parsers for AV1 and VP9 video
+
+-   va: work on the new VA-API plugin implementation for
+    hardware-accelerated video decoding and encoding has continued at
+    pace, with various new decoders and filters having joined the
+    initial vah264dec:
+
+    -   vah265dec: VA-API H.265 decoder
+    -   vavp8dec: VA-API VP8 decoder
+    -   vavp9dec: VA-API VP9 decoder
+    -   vaav1dec: VA-API AV1 decoder
+    -   vampeg2dec: VA-API MPEG-2 decoder
+    -   vadeinterlace: : VA-API deinterlace filter
+    -   vapostproc: : VA-API postproc filter (color conversion,
+        resizing, cropping, color balance, video rotation, skin tone
+        enhancement, denoise, sharpen)
+
+    See Víctor’s blog post “GstVA in GStreamer 1.20” for more details
+    and what’s coming up next.
+
+-   vaapiav1dec: new AV1 decoder element (in gstreamer-vaapi)
+
+-   msdkav1dec: hardware-accelerated AV1 decoder using the Intel Media
+    SDK / oneVPL
+
+-   nvcodec plugin for NVIDIA NVCODEC API for hardware-accelerated video
+    encoding and decoding:
+
+    -   cudaconvert, cudascale: new CUDA based video color space convert
+        and rescale elements
+    -   cudaupload, cudadownload: new helper elements for memory
+        transfer between CUDA and system memory spaces
+    -   nvvp8sldec, nvvp9sldec: new GstCodecs-based VP8/VP9 decoders
+
+-   Various new hardware-accelerated elements for Windows:
+
+    -   d3d11screencapturesrc: new desktop capture element, including a
+        GstDeviceProvider implementation to enumerate/select target
+        monitors for capture.
+    -   d3d11av1dec and d3d11mpeg2dec: AV1 and MPEG-2 decoders
+    -   d3d11deinterlace: deinterlacing filter
+    -   d3d11compositor: video composing element
+    -   see Windows section below for more details
+
+-   new Rust plugins:
+
+    -   audiornnoise: Removes noise from an audio stream
+    -   awstranscribeparse: Parses AWS audio transcripts into timed text
+        buffers
+    -   ccdetect: Detects if valid closed captions are present in a
+        closed captions stream
+    -   cea608tojson: Converts CEA-608 Closed Captions to a JSON
+        representation
+    -   cmafmux: CMAF fragmented MP4 muxer
+    -   dashmp4mux: DASH fragmented MP4 muxer
+    -   isofmp4mux: ISO fragmented MP4 muxer
+    -   ebur128level: EBU R128 Loudness Level Measurement
+    -   ffv1dec: FFV1 video decoder
+    -   gtk4paintablesink: GTK4 video sink, which provides a
+        GdkPaintable that can be rendered in various widgets
+    -   hlssink3: HTTP Live Streaming sink
+    -   hrtfrender: Head-Related Transfer Function (HRTF) renderer
+    -   hsvdetector: HSV colorspace detector
+    -   hsvfilter: HSV colorspace filter
+    -   jsongstenc: Wraps buffers containing any valid top-level JSON
+        structures into higher level JSON objects, and outputs those as
+        ndjson
+    -   jsongstparse: Parses ndjson as output by jsongstenc
+    -   jsontovtt: converts JSON to WebVTT subtitles
+    -   regex: Applies regular expression operations on text
+    -   roundedcorners: Adds rounded corners to video
+    -   spotifyaudiosrc: Spotify source
+    -   textahead: Display upcoming text buffers ahead (e.g. for
+        Karaoke)
+    -   transcriberbin: passthrough bin that transcribes raw audio to
+        closed captions using awstranscriber and puts the captions as
+        metas onto the video
+    -   tttojson: Converts timed text to a JSON representation
+    -   uriplaylistbin: Playlist source bin
+    -   webpdec-rs: WebP image decoder with animation support
+
+-   New plugin codecalpha with elements to assist with WebM Alpha
+    decoding
+
+    -   codecalphademux: Split stream with GstVideoCodecAlphaMeta into
+        two streams
+    -   alphacombine: Combine two raw video stream (I420 or NV12) as one
+        stream with alpha channel (A420 or AV12)
+    -   vp8alphadecodebin: A bin to handle software decoding of VP8 with
+        alpha
+    -   vp9alphadecodebin: A bin to handle software decoding of VP9 with
+        alpha
+
+-   New hardware accelerated elements for Linux:
+
+    -   v4l2slmpeg2dec: Support for Linux Stateless MPEG2 decoders
+    -   v4l2slvp9dec: Support for Linux Stateless VP9 decoders
+    -   v4l2slvp8alphadecodebin: Support HW accelerated VP8 with alpha
+        layer decoding
+    -   v4l2slvp9alphadecodebin: Support HW accelerated VP9 with alpha
+        layer decoding
 
 New element features and additions
 
--   this section will be filled in in due course
+-   assrender: handle more font mime types; better interaction with
+    matroskademux for embedded fonts
+
+-   audiobuffersplit: Add support for specifying output buffer size in
+    bytes (not just duration)
+
+-   audiolatency: new "samplesperbuffer" property so users can configure
+    the number of samples per buffer. The default value is 240 samples
+    which is equivalent to 5ms latency with a sample rate of 48000,
+    which might be larger than actual buffer size of audio capture
+    device.
+
+-   audiomixer, audiointerleave, GstAudioAggregator: now keep a count of
+    samples that are dropped or processed as statistic and can be made
+    to post QoS messages on the bus whenever samples are dropped by
+    setting the "qos-messages" property on input pads.
+
+-   audiomixer, compositor: improved handling of new inputs added at
+    runtime. New API was added to the GstAggregator base class to allow
+    subclasses to opt into an aggregation mode where inactive pads are
+    ignored when processing input buffers
+    (gst_aggregator_set_ignore_inactive_pads(),
+    gst_aggregator_pad_is_inactive()). An “inactive pad” in this context
+    is a pad which, in live mode, hasn’t yet received a first buffer,
+    but has been waited on at least once. What would happen usually in
+    this case is that the aggregator would wait for data on this pad
+    every time, up to the maximum configured latency. This would
+    inadvertently push mixer elements in live mode to the configured
+    latency envelope and delay processing when new inputs are added at
+    runtime until these inputs have actually produced data. This is
+    usually undesirable. With this new API, new inputs can be added
+    (requested) and configured and they won’t delay the data processing.
+    Applications can opt into this new behaviour by setting the
+    "ignore-inactive-pads" property on compositor, audiomixer or other
+    GstAudioAggregator-based elements.
+
+-   cccombiner: implement “scheduling” of captions. So far cccombiner’s
+    behaviour was essentially that of a funnel: it strictly looked at
+    input timestamps to associate together video and caption buffers.
+    Now it will try to smoothly schedule caption buffers in order to
+    have exactly one per output video buffer. This might involve
+    rewriting input captions, for example when the input is CDP then
+    sequence counters are rewritten, time codes are dropped and
+    potentially re-injected if the input video frame had a time code
+    meta. This can also lead to the input drifting from synchronization,
+    when there isn’t enough padding in the input stream to catch up. In
+    that case the element will start dropping old caption buffers once
+    the number of buffers in its internal queue reaches a certain limit
+    (configurable via the "max-scheduled" property). The new original
+    funnel-like behaviour can be restored by setting the "scheduling"
+    property to FALSE.
+
+-   ccconverter: new "cdp-mode" property to specify which sections to
+    include in CDP packets (timecode, CC data, service info). Various
+    software, including ffmpeg’s Decklink support, fails parsing CDP
+    packets that contain anything but CC data in the CDP packets.
+
+-   clocksync: new "sync-to-first" property for automatic timestamp
+    offset setup: if set clocksync will set up the "ts-offset" value
+    based on the first buffer and the pipeline’s running time when the
+    first buffer arrived. The newly configured "ts-offset" in this case
+    would be the value that allows outputting the first buffer without
+    waiting on the clock. This is useful for example to feed a non-live
+    input into an already-running pipeline.
+
+-   compositor:
+
+    -   multi-threaded input conversion and compositing. Set the
+        "max-threads" property to activate this.
+    -   new "sizing-policy" property to support display aspect ratio
+        (DAR)-aware scaling. By default the image is scaled to fill the
+        configured destination rectangle without padding and without
+        keeping the aspect ratio. With sizing-policy=keep-aspect-ratio
+        the input image is scaled to fit the destination rectangle
+        specified by GstCompositorPad:{xpos, ypos, width, height}
+        properties preserving the aspect ratio. As a result, the image
+        will be centered in the destination rectangle with padding if
+        necessary.
+    -   new "zero-size-is-unscaled" property on input pads. By default
+        pad width=0 or pad height=0 mean that the stream should not be
+        scaled in that dimension. But if the "zero-size-is-unscaled"
+        property is set to FALSE a width or height of 0 is instead
+        interpreted to mean that the input image on that pad should not
+        be composited, which is useful when creating animations where an
+        input image is made smaller and smaller until it disappears.
+    -   improved handling of new inputs at runtime via
+        "ignore-inactive-pads"property (see above for details)
+    -   allow output format with alpha even if none of the inputs have
+        alpha (also glvideomixer and other GstVideoAggregator
+        subclasses)
+
+-   dashsink: add h265 codec support and signals for allowing custom
+    playlist/fragment output
+
+-   decodebin3:
+
+    -   improved decoder selection, especially for hardware decoders
+    -   make input activation “atomic” when adding inputs dynamically
+    -   better interleave handling: take into account decoder latency
+        for interleave size
+
+-   decklink:
+
+    -   Updated DeckLink SDK to 11.2 to support DeckLink 8K Pro
+    -   decklinkvideosrc:
+        -   More accurate and stable capture timestamps: use the
+            hardware reference clock time when the frame was finished
+            being captured instead of a clock time much further down the
+            road.
+        -   Automatically detect widescreen vs. normal NTSC/PAL
+
+-   encodebin:
+
+    -   add “smart encoding” support for H.265, VP8 and VP9 (i.e. only
+        re-encode where needed and otherwise pass through encoded video
+        as-is).
+    -   H264/H265 smart encoding improvements: respect user-specified
+        stream-format, but if not specified default to avc3/hvc1 with
+        in-band SPS/PPS/VPS signalling for more flexibility.
+    -   new encodebin2 element with dynamic/sometimes source pads in
+        order to support the option of doing the muxing outside of
+        encodebin, e.g. in combination with splitmuxsink.
+    -   add APIs to set element properties on encoding profiles (see
+        below)
+
+-   errorignore: new "ignore-eos" property to also ignore FLOW_EOS from
+    downstream elements
+
+-   giosrc: add support for growing source files: applications can
+    specify that the underlying file being read is growing by setting
+    the "is-growing" property. If set, the source won’t EOS when it
+    reaches the end of the file, but will instead start monitoring it
+    and will start reading data again whenever a change is detected. The
+    new "waiting-data" and "done-waiting-data" signals keep the
+    application informed about the current state.
+
+-   gtksink, gtkglsink:
+
+    -   scroll event support: forwarded as navigation events into the
+        pipeline
+    -   "video-aspect-ratio-override" property to force a specific
+        aspect ratio
+    -   "rotate-method" property and support automatic rotation based on
+        image tags
+
+-   identity: new "stats" property allows applications to retrieve the
+    number of bytes and buffers that have passed through so far.
+
+-   interlace: add support for more formats, esp 10-bit, 12-bit and
+    16-bit ones
+
+-   jack: new "low-latency" property for automatic latency-optimized
+    setting and "port-names" property to select ports explicitly
+
+-   jpegdec: support output conversion to RGB using libjpeg-turbo (for
+    certain input files)
+
+-   line21dec:
+
+    -   "mode" property to control whether and how detected closed
+        captions should be inserted in the list of existing close
+        caption metas on the input frame (if any): add, drop, or
+        replace.
+    -   "ntsc-only" property to only look for captions if video has NTSC
+        resolution
+
+-   line21enc: new "remove-caption-meta" to remove metas from output
+    buffers after encoding the captions into the video data; support for
+    CDP closed captions
+
+-   matroskademux, matroskamux: Add support for ffv1, a lossless
+    intra-frame video coding format.
+
+-   matroskamux: accept in-band SPS/PPS/VPS for H264 and H265
+    (i.e. stream-format avc3 and hev1) which allows on-the-fly
+    profile/level/resolution changes.
+
+-   matroskamux: new "cluster-timestamp-offset" property, useful for use
+    cases where the container timestamps should map to some absolute
+    wall clock time, for example.
+
+-   rtpsrc: add "caps" property to allow explicit setting of the caps
+    where needed
+
+-   mpegts: support SCTE-35 passthrough via new "send-scte35-events"
+    property on MPEG-TS demuxer tsdemux. When enabled, SCTE 35 sections
+    (eg ad placement opportunities) are forwarded as events donwstream
+    where they can be picked up again by mpegtsmux. This required a
+    semantic change in the SCTE-35 section API: timestamps are now in
+    running time instead of muxer pts.
+
+-   tsdemux: Handle PCR-less MPEG-TS streams; more robust timestamp
+    handling in certain corner cases and for poorly muxed streams.
+
+-   mpegtsmux:
+
+    -   More conformance improvements to make MPEG-TS analyzers happy:
+        -   PCR timing accuracy: Improvements to the way mpegtsmux
+            outputs PCR observations in CBR mode, so that a PCR
+            observation is always inserted when needed, so that we never
+            miss the configured pcr-interval, as that triggers various
+            MPEG-TS analyser errors.
+        -   Improved PCR/SI scheduling
+    -   Don’t write PCR until PAT/PMT are output to make sure streams
+        start cleanly with a PAT/PMT.
+    -   Allow overriding the automatic PMT PID selection via
+        application-supplied PMT_%d fields in the prog-map
+        structure/property.
+
+-   mp4mux:
+
+    -   new "first-moov-then-finalise" mode for fragmented output where
+        the output will start with a self-contained moov atom for the
+        first fragment, and then produce regular fragments. Then at the
+        end when the file is finalised, the initial moov is invalidated
+        and a new moov is written covering the entire file. This way the
+        file is a “fragmented mp4” file while it is still being written
+        out, and remains playable at all times, but at the end it is
+        turned into a regular mp4 file (with former fragment headers
+        remaining as unused junk data in the file).
+    -   support H.264 avc3 and H.265 hvc1 stream formats as input where
+        the codec data is signalled in-band inside the bitstream instead
+        of caps/file headers.
+    -   support profile/level/resolution changes for H264/H265 input
+        streams (i.e. codec data changing on the fly). Each codec_data
+        is put into its own SampleTableEntry inside the stsd, unless the
+        input is in avc3 stream format in which case it’s written
+        in-band an not in the headers.
+
+-   multifilesink: new ""min-keyframe-distance"" property to make
+    minimum distance between keyframes in next-file=key-frame mode
+    configurable instead of hard-coding it to 10 seconds.
+
+-   mxfdemux has seen a big refactoring to support non-frame wrappings
+    and more accurate timestamp/seek handling for some formats
+
+-   msdk plugin for hardware-accelerated video encoding and decoding
+    using the Intel Media SDK:
+
+    -   oneVPL support (Intel oneAPI Video Processing Library)
+    -   AV1 decoding support
+    -   H.264 decoder now supports constrained-high and progressive-high
+        profiles
+    -   H.264 encoder:
+        -   more configuration options (properties):
+            "intra-refresh-type", "min-qp" , "max-qp", "p-pyramid",
+            "dblk-idc"
+    -   H.265 encoder:
+        -   can output main-still-picture profile
+        -   now inserts HDR SEIs (mastering display colour volume and
+            content light level)
+        -   more configuration options (properties):
+            "intra-refresh-type", "min-qp" , "max-qp", "p-pyramid",
+            "b-pyramid", "dblk-idc", "transform-skip"
+        -   support for RGB 10bit format
+    -   External bitrate control in encoders
+    -   Video post proc element msdkvpp gained support for 12-bit pixel
+        formats P012_LE, Y212_LE and Y412_LE
+
+-   nvh264sldec: interlaced stream support
+
+-   openh264enc: support main, high, constrained-high and
+    progressive-high profiles
+
+-   openjpeg: support for multithreaded decoding and encoding
+
+-   rtspsrc: now supports IPv6 also for tunneled mode (RTSP-over-HTTP);
+    new "ignore-x-server-reply" property to ignore the
+    x-server-ip-address server header reply in case of HTTP tunneling,
+    as it is often broken.
+
+-   souphttpsrc: Runtime compatibility support for libsoup2 and
+    libsoup3. libsoup3 is the latest major version of libsoup, but
+    libsoup2 and libsoup3 can’t co-exist in the same process because
+    there is no namespacing or versioning for GObject types. As a
+    result, it would be awkward if the GStreamer souphttpsrc plugin
+    linked to a specific version of libsoup, because it would only work
+    with applications that use the same version of libsoup. To make this
+    work, the soup plugin now tries to determine the libsoup version
+    used by the application (and its other dependencies) at runtime on
+    systems where GStreamer is linked dynamically. libsoup3 support is
+    still considered somewhat experimental at this point.
+
+-   srtsrc, srtsink: add signals for the application to accept/reject
+    incoming connections
+
+-   timeoverlay: new elapsed-running-time time mode which shows the
+    running time since the first running time (and each flush-stop).
+
+-   udpsrc: new timestamping mode to retrieve packet receive timestamps
+    from the kernel via socket control messages (SO_TIMESTAMPNS) on
+    supported platforms
+
+-   uritranscodebin: new setup-source and element-setup signals for
+    applications to configure elements used
+
+-   v4l2codecs plugin gained support for 4x4 and 32x32 tile formats
+    enabling some platforms or direct renders. Important memory usage
+    improvement.
+
+-   v4l2slh264dec now implements the final Linux uAPI as shipped on
+    Linux 5.11 and later.
+
+-   valve: add "drop-mode" property and provide two new modes of
+    operation: in drop-mode=forward-sticky-events sticky events
+    (stream-start, segment, tags, caps, etc.) are forwarded downstream
+    even when dropping is enabled; drop-mode=transform-to-gap will in
+    addition also convert buffers into gap events when dropping is
+    enabled, which lets downstream elements know that time is advancing
+    and might allow for preroll in many scenarios. By default all events
+    and all buffers are dropped when dropping is enabled, which can
+    cause problems with caps negotiation not progressing or branches not
+    prerolling when dropping is enabled.
+
+-   videocrop: support for many more pixel formats, e.g. planar YUV
+    formats with > 8bits and GBR* video formats; can now also accept
+    video not backed by system memory as long as downstream supports the
+    GstCropMeta
+
+-   videotestsrc: new smpte-rp-219 pattern for SMPTE75 RP-219 conformant
+    color bars
+
+-   vp8enc: finish support for temporal scalability: two new properties
+    ("temporal-scalability-layer-flags",
+    "temporal-scalability-layer-sync-flags") and a unit change on the
+    "temporal-scalability-target-bitrate" property (now expects bps);
+    also make temporal scalability details available to RTP payloaders
+    as buffer metadata.
+
+-   vp9enc: new properties to tweak encoder performance:
+
+    -   "aq-mode" to configure adaptive quantization modes
+    -   "frame-parallel-decoding" to configure whether to create a
+        bitstream that reduces decoding dependencies between frames
+        which allows staged parallel processing of more than one video
+        frames in the decoder. (Defaults to TRUE)
+    -   "row-mt", "tile-columns" and "tile-rows" so multithreading can
+        be enabled on a per-tile basis, instead of on a per tile-column
+        basis. In combination with the new "tile-rows" property, this
+        allows the encoder to make much better use of the available CPU
+        power.
+
+-   vp9dec, vp9enc: add support for 10-bit 4:2:0 and 4:2:2 YUV, as well
+    as 8-bit 4:4:4
+
+-   vp8enc, vp9enc now default to “good quality” for the deadline
+    property rather then “best quality”. Having the deadline set to best
+    quality causes the encoder to be absurdly slow, most real-life users
+    will prefer good-enough quality with better performance instead.
+
+-   wpesrc:
+
+    -   implement audio support: a new sometimes source pad will be
+        created for each audio stream created by the web engine.
+    -   move wpesrc to wpevideosrc and add a wrapper bin wpesrc to also
+        support audio
+    -   also handles web:// URIs now (same as cefsrc)
+    -   post messages with the estimated load progress on the bus
 
-Plugin and library moves
+-   x265enc: add negative DTS support, which means timestamps are now
+    offset by 1h same as with x264enc
+
+RTP Payloaders and Depayloaders
+
+-   rtpisacpay, rtpisacdepay: new RTP payloader and depayloader for iSAC
+    audio codec
+
+-   rtph264depay:
+
+    -   new "request-keyframe" property to make the depayloader
+        automatically request a new keyframe from the sender on packet
+        loss, consistent with the new property on rtpvp8depay.
+    -   new "wait-for-keyframe" property to make depayloader wait for a
+        new keyframe at the beginning and after packet loss (only
+        effective if the depayloader outputs AUs), consistent with the
+        existing property on rtpvp8depay.
+
+-   rtpopuspay, rtpopusdepay: support libwebrtc-compatible multichannel
+    audio in addition to the previously supported multichannel audio
+    modes
+
+-   rtpopuspay: add DTX (Discontinuous Transmission) support
+
+-   rtpvp8depay: new "request-keyframe" property to make the depayloader
+    automatically request a new keyframe from the sender on packet loss.
+
+-   rtpvp8pay: temporal scaling support
 
--   this section will be filled in in due course
+-   rtpvp9depay: Improved SVC handling (aggregate all layers)
+
+RTP Infrastructure
+
+-   rtpst2022-1-fecdec, rtpst2022-1-fecenc: new elements providing SMPTE
+    2022-1 2-D Forward Error Correction. More details in Mathieu’s blog
+    post.
+
+-   rtpreddec: BUNDLE support
+
+-   rtpredenc, rtpulpfecenc: add support for Transport-wide Congestion
+    Control (TWCC)
+
+-   rtpsession: new "twcc-feedback-interval" property to allow RTCP TWCC
+    reports to be scheduled on a timer instead of per marker-bit.
+
+Plugin and library moves
 
 -   There were no plugin moves or library moves in this cycle.
 
@@ -56,8 +998,6 @@ Plugin removals
 
 The following elements or plugins have been removed:
 
--   this section will be filled in in due course
-
 -   The ofa audio fingerprinting plugin has been removed. The MusicIP
     database has been defunct for years so this plugin is likely neither
     useful nor used by anyone.
@@ -71,135 +1011,575 @@ The following elements or plugins have been removed:
 
 Miscellaneous API additions
 
--   this section will be filled in in due course
+Core
+
+-   gst_buffer_new_memdup() is a convenience function for the
+    widely-used gst_buffer_new_wrapped(g_memdup(data,size),size)
+    pattern.
+
+-   gst_caps_features_new_single() creates a new single GstCapsFeatures,
+    avoiding the need to use the vararg function with NULL terminator
+    for simple cases.
+
+-   gst_element_type_set_skip_documentation() can be used by plugins to
+    signal that certain elements should not be included in the GStreamer
+    plugin documentation. This is useful for plugins where elements are
+    registered dynamically based on hardware capabilities and/or where
+    the available plugins and properties vary from system to system.
+    This is used in the d3d11 plugin for example to ensure that only the
+    list of default elements is advertised in the documentation.
+
+-   gst_type_find_suggest_empty_simple() is a new convenience function
+    for typefinders for cases where there’s only a media type and no
+    other fields.
+
+-   New API to create elements and set properties at construction time,
+    which is not only convenient, but also allows GStreamer elements to
+    have construct-only properties: gst_element_factory_make_full(),
+    gst_element_factory_make_valist(),
+    gst_element_factory_make_with_properties(),
+    gst_element_factory_create_full(),
+    gst_element_factory_create_valist(),
+    gst_element_factory_create_with_properties().
+
+-   GstSharedTaskPool: new “shared” task pool subclass with slightly
+    different default behaviour than the existing GstTaskPool which
+    would create unlimited number of threads for new tasks. The shared
+    taskpool creates up to N threads (default: 1) and then distributes
+    pending tasks to those threads round-robin style, and blocks if no
+    thread is available. It is possible to join tasks. This can be used
+    by plugins to implement simple multi-threaded processing and is used
+    for the new multi-threaded video conversion and compositing done in
+    GstVideoAggregator, videoconverter and compositor.
+
+Plugins Base Utils library
+
+-   GstDiscoverer:
+
+    -   gst_discoverer_container_info_get_tags() was added to retrieve
+        global/container tags (vs. per-stream tags). Per-Stream tags can
+        be retrieved via the existing
+        gst_discoverer_stream_info_get_tags().
+        gst_discoverer_info_get_tags(), which for many files returns a
+        confusing mix of stream and container tags, has been deprecated
+        in favour of the container/stream-specific functions.
+    -   gst_discoverer_stream_info_get_stream_number() returns a unique
+        integer identifier for a given stream within the given
+        GstDiscoverer context. (If this matches the stream number inside
+        the container bitstream that’s by coincidence and not by
+        design.)
+
+-   gst_pb_utils_get_caps_description_flags() can be used to query
+    whether certain caps represent a container, audio, video, image,
+    subtitles, tags, or something else. This only works for formats
+    known to GStreamer.
+
+-   gst_pb_utils_get_file_extension_from_caps() returns a possible file
+    extension for given caps.
+
+-   gst_codec_utils_h264_get_profile_flags_level(): Parses profile,
+    flags, and level from H264 AvcC codec_data. The format of H264 AVCC
+    extradata/sequence_header is documented in the ITU-T H.264
+    specification section 7.3.2.1.1 as well as in ISO/IEC 14496-15
+    section 5.3.3.1.2.
+
+-   gst_codec_utils_caps_get_mime_codec() to convert caps to a RFC 6381
+    compatible MIME codec string codec. Useful for providing the codecs
+    field inside the Content-Type HTTP header for containerized formats,
+    such as mp4 or matroska.
+
+GStreamer OpenGL integration library and plugins
+
+-   glcolorconvert: added suppport for converting the video formats
+    A420, AV12, BGR, BGRA, RGBP and BGRP.
+
+-   Added support to GstGLBuffer for persistent buffer mappings where a
+    Pixel Buffer Object (PBO) can be mapped by both the CPU and the GPU.
+    This removes a memcpy() when uploading textures or vertices
+    particularly when software decoders (e.g. libav) are direct
+    rendering into our memory. Improves transfer performance
+    significantly. Requires OpenGL 4.4, GL_ARB_buffer_storage or
+    GL_EXT_buffer_storage
+
+-   Added various helper functions for handling 4x4 matrices of affine
+    transformations as used by GstVideoAffineTransformationMeta.
+
+-   Add support to GstGLContext for allowing the application to control
+    the config (EGLConfig, GLXConfig, etc) used when creating the OpenGL
+    context. This allows the ability to choose between RGB16 or RGB10A2
+    or RGBA8 back/front buffer configurations that were previously
+    hardcoded. GstGLContext also supports retrieving the configuration
+    it was created with or from an externally provide OpenGL context
+    handle. This infrastructure is also used to create a compatible
+    config from an application/externally provided OpenGL context in
+    order to improve compatibility with other OpenGL frameworks and GUI
+    toolkits. A new environment variable GST_GL_CONFIG was also added to
+    be able to request a specific configuration from the command line.
+    Note: different platforms will have different functionality
+    available.
+
+-   Add support for choosing between EGL and WGL at runtime when running
+    on Windows. Previously this was a build-time switch. Allows use in
+    e.g. Gtk applications on Windows that target EGL/ANGLE without
+    recompiling GStreamer. gst_gl_display_new_with_type() can be used by
+    applications to choose a specific display type to use.
+
+-   Build fixes to explicitly check for Broadcom-specific libraries on
+    older versions of the Raspberry Pi platform. The Broadcom OpenGL ES
+    and EGL libraries have different filenames. Using the vc4 Mesa
+    driver on the Raspberry Pi is not affected.
+
+-   Added support to glupload and gldownload for transferring RGBA
+    buffers using the memory:NVMM available on the Nvidia Tegra family
+    of embedded devices.
+
+-   Added support for choosing libOpenGL and libGLX as used in a GLVND
+    environment on unix-based platforms. This allows using desktop
+    OpenGL and EGL without pulling in any GLX symbols as would be
+    required with libGL.
+
+Video library
+
+-   New raw video formats:
+
+    -   AV12 (NV12 with alpha plane)
+    -   RGBP and BGRP (planar RGB formats)
+    -   ARGB64 variants with specified endianness instead of host
+        endianness:
+        -   ARGB64_LE, ARGB64_BE
+        -   RGBA64_BE, RGBA64_LE
+        -   BGRA64_BE, BGRA64_LE
+        -   ABGR64_BE, ABGR64_LE
+
+-   gst_video_orientation_from_tag() is new convenience API to parse the
+    image orientation from a GstTagList.
+
+-   GstVideoDecoder subframe support (see below)
+
+-   GstVideoCodecState now also carries some HDR metadata
+
+-   Ancillary video data: implement transform functions for AFD/Bar
+    metas, so they will be forwarded in more cases
+
+MPEG-TS library
+
+This library only handles section parsing and such, see above for
+changes to the actual mpegtsmux and mpegtsdemux elements.
+
+-   many additions and improvements to SCTE-35 section parsing
+-   new API for fetching extended descriptors:
+    gst_mpegts_find_descriptor_with_extension()
+-   add support for SIT sections (Selection Information Tables)
+-   expose event-from-section constructor gst_event_new_mpegts_section()
+-   parse Audio Preselection Descriptor needed for Dolby AC-4
+
+GstWebRTC library + webrtcbin
+
+-   Change the way in which sink pads and transceivers are matched
+    together to support easier usage. If a pad is created without a
+    specific index (i.e. using sink_%u as the pad template), then an
+    available compatible transceiver will be searched for. If a specific
+    index is requested (i.e. sink_1) then if a transceiver for that
+    m-line already exists, that transceiver must match the new sink pad
+    request. If there is no transceiver available in either scenario, a
+    new transceiver is created. If a mixture of both sink_1 and sink_%u
+    requests result in an impossible situation, an error will be
+    produced at pad request time or from create offer/answer.
+
+-   webrtcbin now uses regular ICE nomination instead of libnice’s
+    default of aggressive ICE nomination. Regular ICE nomination is the
+    default recommended by various relevant standards and improves
+    connectivity in specific network scenarios.
+
+-   Add support for limiting the port range used for RTP with the
+    addition of the min-rtp-port and max-rtp-port properties on the ICE
+    object.
+
+-   Expose the SCTP transport as a property on webrtcbin to more closely
+    match the WebRTC specification.
+
+-   Added support for taking into account the data channel transport
+    state when determining the value of the "connection-state" property.
+    Previous versions of the WebRTC spec did not include the data
+    channel state when computing this value.
+
+-   Add configuration for choosing the size of the underlying sockets
+    used for transporting media data
+
+-   Always advertise support for the transport-cc RTCP feedback protocol
+    as rtpbin supports it. For full support, the configured caps (input
+    or through codec-preferences) need to include the relevant RTP
+    header extension.
+
+-   Numerous fixes to caps and media handling to fail-fast when an
+    incompatible situation is detected.
+
+-   Improved support for attaching the required media after a remote
+    offer has been set.
+
+-   Add support for dynamically changing the amount of FEC used for a
+    particular stream.
+
+-   webrtcbin now stops further SDP processing at the first error it
+    encounters.
+
+-   Completed support for either local or the remote closing a data
+    channel.
+
+-   Various fixes when performing BUNDLEing of the media streams in
+    relation to RTX and FEC usage.
+
+-   Add support for writing out QoS DSCP marking on outgoing packets to
+    improve reliability in some network scenarios.
+
+-   Improvements to the statistics returned by the get-stats signal
+    including the addition of the raw statistics from the internal
+    RTPSource, the TWCC stats when available.
+
+-   The webrtc library does not expose any objects anymore with public
+    fields. Instead properties have been added to replace that
+    functionality. If you are accessing such fields in your application,
+    switch to the corresponding properties.
+
+GstCodecs and Video Parsers
+
+-   Support for render delays to improve throughput across all CODECs
+    (used with NVDEC and V4L2).
+-   lots of improvements to parsers and the codec parsing decoder base
+    classes (H264, H265, VP8, VP9, AV1, MPEG-2) used for various
+    hardware-accelerate decoder APIs.
+
+Bindings support
+
+-   gst_allocation_params_new() allocates a GstAllocationParams struct
+    on the heap. This should only be used by bindings (and freed via
+    gst_allocation_params_free() then). In C code you would allocate
+    this on the stack and only init it in place.
+
+-   gst_debug_log_literal() can be used to log a string to the debug log
+    without going through any printf format expansion and associated
+    overhead. This is mostly useful for bindings such as the Rust
+    bindings which may have done their own formatting already .
+
+-   Provide non-inlined versions of refcounting APIs for various
+    GStreamer mini objects, so that they can be consumed by bindings
+    (e.g. gstreamer-sharp): gst_buffer_ref, gst_buffer_unref,
+    gst_clear_buffer, gst_buffer_copy, gst_buffer_replace,
+    gst_buffer_list_ref, gst_buffer_list_unref, gst_clear_buffer_list,
+    gst_buffer_list_copy, gst_buffer_list_replace, gst_buffer_list_take,
+    gst_caps_ref, gst_caps_unref, gst_clear_caps, gst_caps_replace,
+    gst_caps_take, gst_context_ref, gst_context_unref, gst_context_copy,
+    gst_context_replace, gst_event_replace, gst_event_steal,
+    gst_event_take, gst_event_ref, gst_event_unref, gst_clear_event,
+    gst_event_copy, gst_memory_ref, gst_memory_unref, gst_message_ref,
+    gst_message_unref, gst_clear_message, gst_message_copy,
+    gst_message_replace, gst_message_take, gst_promise_ref,
+    gst_promise_unref, gst_query_ref, gst_query_unref, gst_clear_query,
+    gst_query_copy, gst_query_replace, gst_query_take, gst_sample_ref,
+    gst_sample_unref, gst_sample_copy, gst_tag_list_ref,
+    gst_tag_list_unref, gst_clear_tag_list, gst_tag_list_replace,
+    gst_tag_list_take, gst_uri_copy, gst_uri_ref, gst_uri_unref,
+    gst_clear_uri.
+
+-   expose a GType for GstMiniObject
+
+-   gst_device_provider_probe() now returns non-floating device object
+
+API Deprecations
+
+-   gst_element_get_request_pad() has been deprecated in favour of the
+    newly-added gst_element_request_pad_simple() which does the exact
+    same thing but has a less confusing name that hopefully makes clear
+    that the function request a new pad rather than just retrieves an
+    already-existing request pad.
+
+-   gst_discoverer_info_get_tags(), which for many files returns a
+    confusing mix of stream and container tags, has been deprecated in
+    favour of the container-specific and stream-specific functions,
+    gst_discoverer_container_info_get_tags() and
+    gst_discoverer_stream_info_get_tags().
+
+-   gst_video_sink_center_rect() was deprecated in favour of the more
+    generic newly-added gst_video_center_rect().
+
+-   The GST_MEMORY_FLAG_NO_SHARE flag has been deprecated, as it tends
+    to cause problems and prevents sub-buffering. If pooling or lifetime
+    tracking is required, memories should be allocated through a custom
+    GstAllocator instead of relying on the lifetime of the buffers the
+    memories were originally attached to, which is fragile anyway.
+
+-   The GstPlayer high-level playback library is being replaced with the
+    new GstPlay library (see above). GstPlayer should be considered
+    deprecated at this point and will be marked as such in the next
+    development cycle. Applications should be ported to GstPlay.
+
+-   Gstreamer Editing Services: ges_video_transition_set_border(),
+    ges_video_transition_get_border()
+    ges_video_transition_set_inverted()
+    ges_video_transition_is_inverted() have been deprecated, use
+    ges_timeline_element_set_children_properties() instead.
 
 Miscellaneous performance, latency and memory optimisations
 
--   this section will be filled in in due course
+More video conversion fast paths
 
-Miscellaneous other changes and enhancements
+-   v210 ↔ I420, YV12, Y42B, UYVY and YUY2
+-   A420 → RGB
+
+Less jitter when waiting on the system clock
+
+-   Better system clock wait accuracy, less jitter: where available,
+    clock_nanosleep is used for higher accuracy for waits below 500
+    usecs, and waits below 2ms will first use the regular waiting system
+    and then clock_nanosleep for the remainder. The various wait
+    implementation have a latency ranging from 50 to 500+ microseconds.
+    While this is not a major issue when dealing with a low number of
+    waits per second (for ex: video), it does introduce a non-negligible
+    jitter for synchronisation of higher packet rate systems.
+
+Video decoder subframe support
 
--   this section will be filled in in due course
+-   The GstVideoDecoder base class gained API to process input at the
+    sub-frame level. That way video decoders can start decoding slices
+    before they have received the full input frame in its entirety (to
+    the extent this is supported by the codec, of course). This helps
+    with CPU utilisation and reduces latency.
 
-Tracing framework and debugging improvements
+-   This functionality is now being used in the OpenJPEG JPEG 2000
+    decoder, the FFmpeg H.264 decoder (in case of NAL-aligned input) and
+    the OpenMAX H.264/H.265 decoders (in case of NAL-aligned input).
 
--   this section will be filled in in due course
+Miscellaneous other changes and enhancements
+
+-   GstDeviceMonitor no longer fails to start just because one of the
+    device providers failed to start. That could happen for example on
+    systems where the pulseaudio device provider is installed, but
+    pulseaudio isn’t actually running but ALSA is used for audio
+    instead. In the same vein the device monitor now keeps track of
+    which providers have been started (via the new
+    gst_device_provider_is_started()) and only stops actually running
+    device providers when stopping the device monitor.
+
+-   On embedded systems it can be useful to create a registry that can
+    be shared and read by multiple processes running as different users.
+    It is now possible to set the new GST_REGISTRY_MODE environment
+    variable to specify the file mode for the registry file, which by
+    default is set to be only user readable/writable.
+
+-   GstNetClientClock will signal lost sync in case the remote time
+    resets (e.g. because device power cycles), by emitting the “synced”
+    signal with synced=FALSE parameter, so applications can take action.
+
+-   gst_value_deserialize_with_pspec() allows deserialization with a
+    hint for what the target GType should be. This allows for example
+    passing arrays of flags through the command line or
+    gst_util_set_object_arg(), eg: foo="<bar,bar+baz>".
+
+-   It’s now allowed to create an empty GstVideoOverlayComposition
+    without any rectangles by passing a NULL rectangle to
+    gst_video_overlay_composition_new(). This is useful for bindings and
+    simplifies application code in some places.
+
+Tracing framework, debugging and testing improvements
+
+-   New factories tracer to list loaded elements (and other plugin
+    features). This can be useful to collect a list of elements needed
+    for an application, which then in turn can be used to create a
+    tailored minimal GStreamer build that contains just the elements
+    needed and nothing else.
+-   New plugin-feature-loaded tracing hook for use by tracers like the
+    new factories tracer
+
+-   GstHarness: Add gst_harness_set_live() so that harnesses can be set
+    to non-live and return is-live=false in latency queries if needed.
+    Default behaviour is to always return is-live=true in latency
+    queries.
+
+-   navseek: new "hold-eos" property. When enabled, the element will
+    hold back an EOS event until the next keystroke (via navigation
+    events). This can be used to keep a video sink showing the last
+    frame of a video pipeline until a key is pressed instead of tearing
+    it down immediately on EOS.
+
+-   New fakeaudiosink element: mimics an audio sink and can be used for
+    testing and CI pipelines on systems where no audio system is
+    installed or running. It differs from fakesink in that it only
+    support audio caps and syncs to the clock by default like a normal
+    audio sink. It also implements the GstStreamVolume interface like
+    most audio sinks do.
+
+-   New videocodectestsink element for video codec conformance testing:
+    Calculates MD5 checksums for video frames and skips any padding
+    whilst doing so. Can optionally also write back the video data with
+    padding removed into a file for easy byte-by-byte comparison with
+    reference data.
 
 Tools
 
--   this section will be filled in in due course
+gst-inspect-1.0
+
+-   Can sort the list of plugins by passing --sort=name as command line
+    option
+
+gst-launch-1.0
+
+-   will now error out on top-level properties that don’t exist and
+    which were silently ignored before
+-   On Windows the high-resolution clock is enabled now, which provides
+    better clock and timer performance on Windows (see Windows section
+    below for more details).
+
+gst-play-1.0
+
+-   New --start-position command line argument to start playback from
+    the specified position
+-   Audio can be muted/unmuted in interactive mode by pressing the m
+    key.
+-   On Windows the high-resolution clock is enabled now (see Windows
+    section below for more details)
+
+gst-device-monitor-1.0
+
+-   New --include-hidden command line argument to also show “hidden”
+    device providers
+
+ges-launch-1.0
+
+-   New interactive mode that allows seeking and such. Can be disabled
+    by passing the --no-interactive argument on the command line.
+-   Option to forward tags
+-   Allow using an existing clip to determine the rendering format (both
+    topology and profile) via new --profile-from command line argument.
 
 GStreamer RTSP server
 
--   this section will be filled in in due course
+-   GstRTSPMediaFactory gained API to disable RTCP
+    (gst_rtsp_media_factory_set_enable_rtcp(), "enable-rtcp" property).
+    Previously RTCP was always allowed for all RTSP medias. With this
+    change it is possible to disable RTCP completely, no matter if the
+    client wants to do RTCP or not.
+
+-   Make a mount point of / work correctly. While not allowed by the
+    RTSP 2 spec, the RTSP 1 spec is silent on this and it is used in the
+    wild. It is now possible to use / as a mount path in
+    gst-rtsp-server, e.g. rtsp://example.com/ would work with this now.
+    Note that query/fragment parts of the URI are not necessarily
+    correctly handled, and behaviour will differ between various
+    client/server implementations; so use it if you must but don’t bug
+    us if it doesn’t work with third party clients as you’d hoped.
+
+-   multithreading fixes (races, refcounting issues, deadlocks)
+
+-   ONVIF audio backchannel fixes
+
+-   ONVIF trick mode optimisations
+
+-   rtspclientsink: new "update-sdp" signal that allows updating the SDP
+    before sending it to the server via ANNOUNCE. This can be used to
+    add additional metadata to the SDP, for example. The order and
+    number of medias must not be changed, however.
 
 GStreamer VAAPI
 
--   this section will be filled in in due course
+-   new AV1 decoder element (vaapiav1dec)
+
+-   H264 decoder: handle stereoscopic 3D video with frame packing
+    arrangement SEI messages
+
+-   H265 encoder: added Screen Content Coding extensions support
+
+-   H265 decoder: gained MAIN_444_12 profile support (decoded to
+    Y412_LE), and 4:2:2 12-bits support (decoded to Y212_LE)
+
+-   vaapipostproc: gained BT2020 color standard support
+
+-   vaapidecode: now generates caps templates dynamically at runtime in
+    order to advertise actually supported caps instead of all
+    theoretically supported caps.
+
+-   GST_VAAPI_DRM_DEVICE environment variable to force a specified DRM
+    device when a DRM display is used. It is ignored when other types of
+    displays are used. By default /dev/dri/renderD128 is used for DRM
+    display.
 
 GStreamer OMX
 
--   this section will be filled in in due course
+-   subframe support in H.264/H.265 decoders
 
 GStreamer Editing Services and NLE
 
--   this section will be filled in in due course
+-   framepositioner: new "operator" property to access blending modes in
+    the compositor
+-   timeline: Implement snapping to markers
+-   smart-mixer: Add support for d3d11compositor and glvideomixer
+-   titleclip: add "draw-shadow" child property
+-   ges:// URI support to define a timeline from a description.
+-   command-line-formatter
+    -   Add track management to timeline description
+    -   Add keyframe support
+-   ges-launch-1.0:
+    -   Add an interactive mode where we can seek etc…
+    -   Add option to forward tags
+    -   Allow using an existing clip to determine the rendering format
+        (both topology and profile) via new --profile-from command line
+        argument.
+-   Fix static build
 
 GStreamer validate
 
--   this section will be filled in in due course
+-   report: Add a way to force backtraces on reports even if not a
+    critical issue (GST_VALIDATE_ISSUE_FLAGS_FORCE_BACKTRACE)
+-   Add a flag to gst_validate_replace_variables_in_string() allow
+    defining how to resolve variables in structs
+-   Add gst_validate_bin_monitor_get_scenario() to get the bin monitor
+    scenario, which is useful for applications that use Validate
+    directly.
+-   Add an expected-values parameter to wait, message-type=XX allowing
+    more precise filtering of the message we are waiting for.
+-   Add config file support: each test can now use a config file for the
+    given media file used to test.
+-   Add support to check properties of object properties
+-   scenario: Add an "action-done" signal to signal when an action is
+    done
+-   scenario: Add a "run-command" action type
+-   scenario: Allow forcing running action on idle from scenario file
+-   scenario: Allow iterating over arrays in foreach
+-   scenario: Rename ‘interlaced’ action to ‘non-blocking’
+-   scenario: Add a non-blocking flag to the wait signal
 
 GStreamer Python Bindings
 
--   this section will be filled in in due course
+-   Fixes for Python 3.10
+-   Various build fixes
+-   at least one known breaking change caused by g-i annotation changes
+    (see below)
 
 GStreamer C# Bindings
 
--   this section will be filled in in due course
+-   Fix GstDebugGraphDetails enum
+-   Updated to latests GtkSharp
+-   Updated to include GStreamer 1.20 API
 
 GStreamer Rust Bindings and Rust Plugins
 
-The GStreamer Rust bindings are released separately with a different
-release cadence that’s tied to gtk-rs, but the latest release has
-already been updated for the upcoming new GStreamer 1.20 API.
-
-gst-plugins-rs, the module containing GStreamer plugins written in Rust,
-has also seen lots of activity with many new elements and plugins.
-
-What follows is a list of elements and plugins available in
-gst-plugins-rs, so people don’t miss out on all those potentially useful
-elements that have no C equivalent.
-
--   FIXME: add new elements
-
-Rust audio plugins
-
--   audiornnoise: New element for audio denoising which implements the
-    noise removal algorithm of the Xiph RNNoise library, in Rust
--   rsaudioecho: Port of the audioecho element from gst-plugins-good
-    rsaudioloudnorm: Live audio loudness normalization element based on
-    the FFmpeg af_loudnorm filter
--   claxondec: FLAC lossless audio codec decoder element based on the
-    pure-Rust claxon implementation
--   csoundfilter: Audio filter that can use any filter defined via the
-    Csound audio programming language
--   lewtondec: Vorbis audio decoder element based on the pure-Rust
-    lewton implementation
-
-Rust video plugins
-
--   cdgdec/cdgparse: Decoder and parser for the CD+G video codec based
-    on a pure-Rust CD+G implementation, used for example by karaoke CDs
--   cea608overlay: CEA-608 Closed Captions overlay element
--   cea608tott: CEA-608 Closed Captions to timed-text (e.g. VTT or SRT
-    subtitles) converter
--   tttocea608: CEA-608 Closed Captions from timed-text converter
--   mccenc/mccparse: MacCaption Closed Caption format encoder and parser
--   sccenc/sccparse: Scenarist Closed Caption format encoder and parser
--   dav1dec: AV1 video decoder based on the dav1d decoder implementation
-    by the VLC project
--   rav1enc: AV1 video encoder based on the fast and pure-Rust rav1e
-    encoder implementation
--   rsflvdemux: Alternative to the flvdemux FLV demuxer element from
-    gst-plugins-good, not feature-equivalent yet
--   rsgifenc/rspngenc: GIF/PNG encoder elements based on the pure-Rust
-    implementations by the image-rs project
-
-Rust text plugins
-
--   textwrap: Element for line-wrapping timed text (e.g. subtitles) for
-    better screen-fitting, including hyphenation support for some
-    languages
-
-Rust network plugins
-
--   reqwesthttpsrc: HTTP(S) source element based on the Rust
-    reqwest/hyper HTTP implementations and almost feature-equivalent
-    with the main GStreamer HTTP source souphttpsrc
--   s3src/s3sink: Source/sink element for the Amazon S3 cloud storage
--   awstranscriber: Live audio to timed text transcription element using
-    the Amazon AWS Transcribe API
-
-Generic Rust plugins
-
--   sodiumencrypter/sodiumdecrypter: Encryption/decryption element based
-    on libsodium/NaCl
--   togglerecord: Recording element that allows to pause/resume
-    recordings easily and considers keyframe boundaries
--   fallbackswitch/fallbacksrc: Elements for handling potentially
-    failing (network) sources, restarting them on errors/timeout and
-    showing a fallback stream instead
--   threadshare: Set of elements that provide alternatives for various
-    existing GStreamer elements but allow to share the streaming threads
-    between each other to reduce the number of threads
--   rsfilesrc/rsfilesink: File source/sink elements as replacements for
-    the existing filesrc/filesink elements
+-   The GStreamer Rust bindings are released separately with a different
+    release cadence that’s tied to gtk-rs, but the latest release has
+    already been updated for the upcoming new GStreamer 1.20 API (v1_20
+    feature).
+
+-   gst-plugins-rs, the module containing GStreamer plugins written in
+    Rust, has also seen lots of activity with many new elements and
+    plugins. See the New Elements section above for a list of new Rust
+    elements.
 
 Build and Dependencies
 
 -   Meson 0.59 or newer is required to build GStreamer now.
 
--   FIXME: this section will be filled in in due course
+-   The GLib requirement has been bumped to GLib 2.56 or newer (from
+    March 2018).
+
+-   The wpe plugin now requires wpe >= 2.28 and wpebackend-fdo >= 1.8
 
 Explicit opt-in required for build of certain plugins with (A)GPL dependencies
 
@@ -211,11 +1591,9 @@ even if the required dependencies are available.
 See Building plugins with (A)GPL-licensed dependencies for more details
 and a non-exhaustive list of plugins affected.
 
-gst-build: replaced by Monorepo
+gst-build: replaced by mono repository
 
--   this section will be filled in in due course
-
--   FIXME: describe + link to Monorepo FAQ
+See mono repository section above and the GStreamer mono repository FAQ.
 
 Cerbero
 
@@ -223,76 +1601,254 @@ Cerbero is a meta build system used to build GStreamer plus dependencies
 on platforms where dependencies are not readily available, such as
 Windows, Android, iOS and macOS.
 
-General improvements
+General Cerbero improvements
 
--   this section will be filled in in due course
+-   Plugin removed: libvisual
+-   New plugins: rtpmanagerbad and rist
 
-macOS / iOS
+macOS / iOS specific Cerbero improvements
 
--   this section will be filled in in due course
+-   XCode 12 support
+-   macOS OS release support is now future-proof, similar to iOS
+-   macOS Apple Silicon (ARM64) cross-compile support has been added
+-   macOS Apple Silicon (ARM64) native support is currently experimental
 
-Windows
+Windows specific Cerbero improvements
 
--   this section will be filled in in due course
+-   Visual Studio 2022 support has been added
+-   bootstrap is faster since it requires building fewer build-tools
+    recipes on Windows
+-   package is faster due to better scheduling of recipe stages and
+    elimination of unnecessary autotools regeneration
+-   The following plugins are no longer built on Windows:
+    -   a52dec (another decoder is still available in libav)
+    -   dvdread
+    -   resindvd
 
 Windows MSI installer
 
--   this section will be filled in in due course
+-   no major changes
 
-Linux
+Linux specific Cerbero improvements
 
--   this section will be filled in in due course
+-   Fedora, Debian OS release support is now more future-proof
+-   Amazon Linux 2 support has been added
 
-Android
+Android specific Cerbero improvements
 
--   this section will be filled in in due course
+-   no major changes
 
 Platform-specific changes and improvements
 
 Android
 
--   this section will be filled in in due course
+-   No major changes
 
 macOS and iOS
 
--   this section will be filled in in due course
+-   applemedia: add ProRes support to vtenc and vtdec
 
 Windows
 
--   this section will be filled in in due course
+-   On Windows the high-resolution clock is enabled now in the
+    gst-launch-1.0 and gst-play-1.0 command line tools, which provides
+    better clock and timer performance on Windows, at the cost of higher
+    power consumption. By default, without the high-resolution clock
+    enabled, the timer precision on Windows is system-dependent and may
+    be as bad as 15ms which is not good enough for many multimedia
+    applications. Developers may want to do the same in their Windows
+    applications if they think it’s a good idea for their application
+    use case, and depending on the Windows version they target. This is
+    not done automatically by GStreamer because on older Windows
+    versions (pre-Windows 10) this affects a global Windows setting and
+    also there’s a power consumption vs. performance trade-off that may
+    differ from application to application.
+
+-   dxgiscreencapsrc now supports resolution changes
+
+-   The wasapi2 audio plugin was rewritten and now has a higher rank
+    than the old wasapi plugin since it has a number of additional
+    features such as automatic stream routing, and no
+    known-but-hard-to-fix issues. The plugin is always built if the
+    Windows 10 SDK is available now.
+
+-   The wasapi device providers now detect and notify dynamic device
+    additions/removals
+
+-   d3d11screencapturesrc: new desktop capture element, including
+    GstDeviceProvider implementation to enumerate/select target monitors
+    for capture.
+
+-   Direct3D11/DXVA decoder now supports AV1 and MPEG2 codecs
+    (d3d11av1dec, d3d11mpeg2dec)
+
+-   VP9 decoding got more reliable and stable thanks to a newly written
+    codec parser
+
+-   Support for decoding interlaced H.264/AVC streams
+
+-   Hardware-accelerated video deinterlacing (d3d11deinterlace) and
+    video mixing (d3d11compositor)
+
+-   Video mixing with the Direct3D11 API (d3d11compositor)
+
+-   MediaFoundation API based hardware encoders gained the ability to
+    receive Direct3D11 textures as an input
+
+-   Seungha’s blog post “GStreamer ❤ Windows: A primer on the cool stuff
+    you’ll find in the 1.20 release” describes many of the
+    Windows-related improvements in more detail
 
 Linux
 
--   this section will be filled in in due course
+-   bluez: LDAC Bluetooth audio codec support in a2dpsink and avdtpsink,
+    as well as an LDAC RTP payloader (rtpldacpay) and an LDAC audio
+    encoder (ldacenc)
+
+-   kmssink: gained support for NV24, NV61, RGB16/BGR16 formats;
+    auto-detect NVIDIA Tegra driver
 
 Documentation improvements
 
--   this section will be filled in in due course
+-   hardware-accelerated GPU plugins will now no longer always list all
+    the element variants for all available GPUs, since those are
+    system-dependent and it’s confusing for users to see those in the
+    documentation just because the GStreamer developer who generated the
+    docs had multiple GPUs to play with at the time. Instead just show
+    the default elements.
 
-Possibly Breaking Changes
+Possibly Breaking and Other Noteworthy Behavioural Changes
+
+-   gst_parse_launch(), gst_parse_bin_from_description() and friends
+    will now error out when setting properties that don’t exist on
+    top-level bins. They were silently ignored before.
+
+-   The GstWebRTC library does not expose any objects anymore with
+    public fields. Instead properties have been added to replace that
+    functionality. If you are accessing such fields in your application,
+    switch to the corresponding properties.
 
--   FIXME: this section will be filled in in due course
--   MPEG-TS SCTE-35 API changes (FIXME: flesh out)
--   gst_parse_launch() and friends now error out on non-existing
-    properties on top-level bins where they would silently fail and
-    ignore those before.
 -   playbin and uridecodebin now emit the source-setup signal before the
     element is added to the bin and linked so that the source element is
     already configured before any scheduling query comes in, which is
-    useful for elements such as appsrc or giostreamsrc. (Merge Request)
+    useful for elements such as appsrc or giostreamsrc.
+
+-   The source element inside urisourcebin (used inside uridecodebin3
+    which is used inside playbin3) is no longer called "source". This
+    shouldn’t affect anyone hopefully, because there’s a "setup-source"
+    signal to configure the source element and no one should rely on
+    names of internal elements anyway.
+
+-   The vp8enc element now expects bps (bits per second) for the
+    "temporal-scalability-target-bitrate" property, which is consistent
+    with the "target-bitrate" property. Since additional configuration
+    is required with modern libvpx to make temporal scaling work anyway,
+    chances are that very few people will have been using this property
+
+-   vp8enc and vp9enc now default to “good quality” for the "deadline"
+    property rather then “best quality”. Having the deadline set to best
+    quality causes the encoder to be absurdly slow, most real-life users
+    will want the good quality tradeoff instead.
+
+-   The experimental GstTranscoder library API in gst-plugins-bad was
+    changed from a GObject signal-based notification mechanism to a
+    GstBus/message-based mechanism akin to GstPlayer/GstPlay.
+
+-   MPEG-TS SCTE-35 API: semantic change for SCTE-35 splice commands:
+    timestamps passed by the application should be in running time now,
+    since users of the API can’t really be expected to predict the local
+    PTS of the muxer.
+
+-   The GstContext used by souphttpsrc to share the session between
+    multiple element instances has changed. Previously it provided
+    direct access to the internal SoupSession object, now it only
+    provides access to an opaque, internal type. This change is
+    necessary because SoupSession is not thread-safe at all and can’t be
+    shared safely between arbitrary external code and souphttpsrc.
+
+-   Python bindings: GObject-introspection related Annotation fixes have
+    led to a case of a GstVideo.VideoInfo-related function signature
+    changing in the Python bindings (possibly one or two other cases
+    too). This is for a function that should never have been exposed in
+    the first place though, so the bindings are being updated to throw
+    an exception in that case, and the correct replacement API has been
+    added in form of an override.
 
 Known Issues
 
--   this section will be filled in in due course
-
--   There are a couple of known WebRTC-related regressions/blockers:
-
-    -   webrtc: DTLS setup with Chrome is broken
-    -   webrtcbin: First keyframe is usually lost
+-   nothing in particular at this point (but also see possibly breaking
+    changes section above)
 
 Contributors
 
--   this section will be filled in in due course
+Aaron Boxer, Adam Leppky, Adam Williamson, Alba Mendez, Alejandro
+González, Aleksandr Slobodeniuk, Alexander Vandenbulcke, Alex Ashley,
+Alicia Boya García, Andika Triwidada, Andoni Morales Alastruey, Andrew
+Wesie, Andrey Moiseev, Antonio Ospite, Antonio Rojas, Arthur Crippa
+Búrigo, Arun Raghavan, Ashley Brighthope, Axel Kellermann, Baek, Bastien
+Nocera, Bastien Reboulet, Benjamin Gaignard, Bing Song, Binh Truong,
+Biswapriyo Nath, Brad Hards, Brad Smith, Brady J. Garvin, Branko
+Subasic, Camilo Celis Guzman, Chris Bass, ChrisDuncanAnyvision, Chris
+White, Corentin Damman, Daniel Almeida, Daniel Knobe, Daniel Stone,
+david, David Fernandez, David Keijser, David Phung, Devarsh Thakkar,
+Dinesh Manajipet, Dmitry Samoylov, Dmitry Shusharin, Dominique Martinet,
+Doug Nazar, Ederson de Souza, Edward Hervey, Emmanuel Gil Peyrot,
+Enrique Ocaña González, Ezequiel Garcia, Fabian Orccon, Fabrice
+Fontaine, Fernando Jimenez Moreno, Florian Karydes, Francisco Javier
+Velázquez-García, François Laignel, Frederich Munch, Fredrik Pålsson,
+George Kiagiadakis, Georg Lippitsch, Göran Jönsson, Guido Günther,
+Guillaume Desmottes, Guiqin Zou, Haakon Sporsheim, Haelwenn (lanodan)
+Monnier, Haihao Xiang, Haihua Hu, Havard Graff, He Junyan, Helmut
+Januschka, Henry Wilkes, Hosang Lee, Hou Qi, Ignacio Casal Quinteiro,
+Igor Kovalenko, Ilya Kreymer, Imanol Fernandez, Jacek Tomaszewski, Jade
+Macho, Jakub Adam, Jakub Janků, Jan Alexander Steffens (heftig), Jan
+Schmidt, Jason Carrete, Jason Pereira, Jay Douglass, Jeongki Kim, Jérôme
+Laheurte, Jimmi Holst Christensen, Johan Sternerup, John Hassell, John
+Lindgren, John-Mark Bell, Jonathan Matthew, Jordan Petridis, Jose
+Quaresma, Julian Bouzas, Julien, Kai Uwe Broulik, Kasper Steensig
+Jensen, Kellermann Axel, Kevin Song, Khem Raj, Knut Inge Hvidsten, Knut
+Saastad, Kristofer Björkström, Lars Lundqvist, Lawrence Troup, Lim Siew
+Hoon, Lucas Stach, Ludvig Rappe, Luis Paulo Fernandes de Barros, Luke
+Yelavich, Mads Buvik Sandvei, Marc Leeman, Marco Felsch, Marek Vasut,
+Marian Cichy, Marijn Suijten, Marius Vlad, Markus Ebner, Mart Raudsepp,
+Matej Knopp, Mathieu Duponchelle, Matthew Waters, Matthieu De Beule,
+Mengkejiergeli Ba, Michael de Gans, Michael Olbrich, Michael Tretter,
+Michal Dzik, Miguel Paris, Mikhail Fludkov, mkba, Nazar Mokrynskyi,
+Nicholas Jackson, Nicola Murino, Nicolas Dufresne, Niklas Hambüchen,
+Nikolay Sivov, Nirbheek Chauhan, Olivier Blin, Olivier Crete, Olivier
+Crête, Paul Goulpié, Per Förlin, Peter Boba, P H, Philippe Normand,
+Philipp Zabel, Pieter Willem Jordaan, Piotrek Brzeziński, Rafał
+Dzięgiel, Rafostar, raghavendra, Raghavendra, Raju Babannavar, Raleigh
+Littles III, Randy Li, Randy Li (ayaka), Ratchanan Srirattanamet, Raul
+Tambre, reed.lawrence, Ricky Tang, Robert Rosengren, Robert Swain, Robin
+Burchell, Roman Sivriver, R S Nikhil Krishna, Ruben Gonzalez, Ruslan
+Khamidullin, Sanchayan Maity, Scott Moreau, Sebastian Dröge, Sergei
+Kovalev, Seungha Yang, Sid Sethupathi, sohwan.park, Sonny Piers, Staz M,
+Stefan Brüns, Stéphane Cerveau, Stephan Hesse, Stian Selnes, Stirling
+Westrup, Théo MAILLART, Thibault Saunier, Tim, Timo Wischer, Tim-Philipp
+Müller, Tim Schneider, Tobias Ronge, Tom Schoonjans, Tulio Beloqui,
+tyler-aicradle, U. Artie Eoff, Ung, Val Doroshchuk, VaL Doroshchuk,
+Víctor Manuel Jáquez Leal, Vivek R, Vivia Nikolaidou, Vivienne
+Watermeier, Vladimir Menshakov, Will Miller, Wim Taymans, Xabier
+Rodriguez Calvar, Xavier Claessens, Xℹ Ruoyao, Yacine Bandou, Yinhang
+Liu, youngh.lee, youngsoo.lee, yychao, Zebediah Figura, Zhang yuankun,
+Zhang Yuankun, Zhao, Zhao Zhili, , Aleksandar Topic, Antonio Ospite,
+Bastien Nocera, Benjamin Gaignard, Brad Hards, Carlos Falgueras García,
+Célestin Marot, Corentin Damman, Corentin Noël, Daniel Almeida, Daniel
+Knobe, Danny Smith, Dave Piché, Dmitry Osipenko, Fabrice Fontaine,
+fjmax, Florian Zwoch, Guillaume Desmottes, Haihua Hu, Heinrich Kruger,
+He Junyan, Jakub Adam, James Cowgill, Jan Alexander Steffens (heftig),
+Jean Felder, Jeongki Kim, Jiri Uncovsky, Joe Todd, Jordan Petridis,
+Krystian Wojtas, Marc-André Lureau, Marcin Kolny, Marc Leeman, Mark
+Nauwelaerts, Martin Reboredo, Mathieu Duponchelle, Matthew Waters,
+Mengkejiergeli Ba, Michael Gruner, Nicolas Dufresne, Nirbheek Chauhan,
+Olivier Crête, Philippe Normand, Rafał Dzięgiel, Ralf Sippl, Robert
+Mader, Sanchayan Maity, Sangchul Lee, Sebastian Dröge, Seungha Yang,
+Stéphane Cerveau, Teh Yule Kim, Thibault Saunier, Thomas Klausner, Timo
+Wischer, Tim-Philipp Müller, Tobias Reineke, Tomasz Andrzejak, Trung Do,
+Tyler Compton, Ung, Víctor Manuel Jáquez Leal, Vivia Nikolaidou, Wim
+Taymans, wngecn, Wonchul Lee, wuchang li, Xavier Claessens, Xi Ruoyao,
+Yoshiharu Hirose, Zhao,
 
 … and many others who have contributed bug reports, translations, sent
 suggestions or helped testing.
@@ -307,7 +1863,7 @@ the git 1.20 branch, which will be a stable branch.
 
 1.20.0
 
-1.20.0 is scheduled to be released around October/November 2021.
+1.20.0 is scheduled to be released around early February 2022.
 
 Schedule for 1.22
 
@@ -315,7 +1871,9 @@ Our next major feature release will be 1.22, and 1.21 will be the
 unstable development version leading up to the stable 1.22 release. The
 development of 1.21/1.22 will happen in the git main branch.
 
-The plan for the 1.22 development cycle is yet to be confirmed.
+The plan for the 1.22 development cycle is yet to be confirmed. Assuming
+no major project-wide reorganisations in the 1.22 cycle we might try and
+aim for a release around August 2022.
 
 1.22 will be backwards-compatible to the stable 1.20, 1.18, 1.16, 1.14,
 1.12, 1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series.
@@ -323,6 +1881,7 @@ The plan for the 1.22 development cycle is yet to be confirmed.
 ------------------------------------------------------------------------
 
 These release notes have been prepared by Tim-Philipp Müller with
-contributions from …
+contributions from Matthew Waters, Nicolas Dufresne, Nirbheek Chauhan,
+Sebastian Dröge and Seungha Yang.
 
 License: CC BY-SA 4.0
diff --git a/subprojects/gst-plugins-ugly/RELEASE b/subprojects/gst-plugins-ugly/RELEASE
index 537d723c34..4d0a301fee 100644
--- a/subprojects/gst-plugins-ugly/RELEASE
+++ b/subprojects/gst-plugins-ugly/RELEASE
@@ -1,4 +1,4 @@
-This is GStreamer gst-plugins-ugly 1.19.3.
+This is GStreamer gst-plugins-ugly 1.19.90.
 
 GStreamer 1.19 is the development branch leading up to the next major
 stable version which will be 1.20.
diff --git a/subprojects/gst-plugins-ugly/gst-plugins-ugly.doap b/subprojects/gst-plugins-ugly/gst-plugins-ugly.doap
index 1d29d9dbcb..68dba52806 100644
--- a/subprojects/gst-plugins-ugly/gst-plugins-ugly.doap
+++ b/subprojects/gst-plugins-ugly/gst-plugins-ugly.doap
@@ -35,6 +35,16 @@ might be widely known to present patent problems.
 
  <release>
   <Version>
+   <revision>1.19.90</revision>
+   <branch>main</branch>
+   <name></name>
+   <created>2022-01-28</created>
+   <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-plugins-ugly/gst-plugins-ugly-1.19.90.tar.xz" />
+  </Version>
+ </release>
+
+ <release>
+  <Version>
    <revision>1.19.3</revision>
    <branch>main</branch>
    <name></name>
diff --git a/subprojects/gst-plugins-ugly/meson.build b/subprojects/gst-plugins-ugly/meson.build
index b9c6772ba0..2b812c0133 100644
--- a/subprojects/gst-plugins-ugly/meson.build
+++ b/subprojects/gst-plugins-ugly/meson.build
@@ -1,5 +1,5 @@
 project('gst-plugins-ugly', 'c',
-  version : '1.19.3.1',
+  version : '1.19.90',
   meson_version : '>= 0.59',
   default_options : [ 'warning_level=1',
                       'buildtype=debugoptimized' ])
diff --git a/subprojects/gst-python/ChangeLog b/subprojects/gst-python/ChangeLog
index f88f7b041b..a9b31afac6 100644
--- a/subprojects/gst-python/ChangeLog
+++ b/subprojects/gst-python/ChangeLog
@@ -1,3 +1,18 @@
+=== release 1.19.90 ===
+
+2022-01-28 14:28:35 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* NEWS:
+	* RELEASE:
+	* gst-python.doap:
+	* meson.build:
+	  Release 1.19.90
+
+2022-01-28 14:28:28 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* ChangeLog:
+	  Update ChangeLogs for 1.19.90
+
 2022-01-26 00:02:49 +0100  Mathieu Duponchelle <mathieu@centricular.com>
 
 	* gi/overrides/GstAudio.py:
diff --git a/subprojects/gst-python/NEWS b/subprojects/gst-python/NEWS
index e9f5227c15..1a512c1848 100644
--- a/subprojects/gst-python/NEWS
+++ b/subprojects/gst-python/NEWS
@@ -1,23 +1,19 @@
 GStreamer 1.20 Release Notes
 
 GStreamer 1.20 has not been released yet. It is scheduled for release in
-November 2021.
+late January / early February 2022.
 
 1.19.x is the unstable development version that is being developed in
-the git main branch and which will eventually result in 1.20, and 1.19.3
-is the current development release in that series
-
-It is expected that feature freeze will be in early October 2021,
-followed by one or two 1.19.9x pre-releases and the new 1.20 stable
-release around October/November 2021.
+the git main branch and which will eventually result in 1.20, and
+1.19.90 is the first release candidate in that series (1.20rc1).
 
 1.20 will be backwards-compatible to the stable 1.18, 1.16, 1.14, 1.12,
-1.10, 1.8, 1.6,, 1.4, 1.2 and 1.0 release series.
+1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series.
 
 See https://gstreamer.freedesktop.org/releases/1.20/ for the latest
 version of this document.
 
-Last updated: Monday 1 November 2021, 01:00 UTC (log)
+Last updated: Wednesday 26 January 2022, 01:00 UTC (log)
 
 Introduction
 
@@ -30,25 +26,971 @@ fixes and other improvements.
 
 Highlights
 
--   this section will be completed in due course
+-   Development in GitLab was switched to a single git repository
+    containing all the modules
+-   GstPlay: new high-level playback library, replaces GstPlayer
+-   WebM Alpha decoding support
+-   Encoding profiles can now be tweaked with additional
+    application-specified element properties
+-   Compositor: multi-threaded video conversion and mixing
+-   RTP header extensions: unified support in RTP depayloader and
+    payloader base classes
+-   SMPTE 2022-1 2-D Forward Error Correction support
+-   Smart encoding (passthrough) support for VP8, VP9, H.265 in
+    encodebin and transcodebin
+-   Runtime compatibility support for libsoup2 and libsoup3 (libsoup3
+    support experimental)
+-   Video decoder subframe support
+-   Video decoder automatic packet-loss, data corruption, and keyframe
+    request handling for RTP / WebRTC / RTSP
+-   MP4 and Matroska muxers now support profile/level/resolution changes
+    for H264/H265 input streams (i.e. codec data changing on the fly)
+-   MP4 muxing mode that initially creates a fragmented mp4 which is
+    converted to a regular mp4 on EOS
+-   Audio support for the WebKit Port for Embedded (WPE) web page source
+    element
+-   CUDA based video color space convert and rescale elements and
+    upload/download elements
+-   NVIDIA memory:NVMM support for OpenGL glupload and gldownload
+    elements
+-   Many WebRTC improvements
+-   The new VA-API plugin implemention fleshed out with more decoders
+    and new postproc elements
+-   AppSink API to retrieve events in addition to buffers and buffer
+    lists
+-   AppSrc gained more configuration options for the internal queue
+    (leakiness, limits in buffers and time, getters to read current
+    levels)
+-   Updated Rust bindings and many new Rust plugins
+-   Improved support for custom minimal GStreamer builds
+-   Support build against FFmpeg 5.0
+-   Linux Stateless CODEC support gained MPEG2 and VP9
+-   Windows Direct3D11/DXVA decoder gained AV1 and MPEG2 support
+-   Lots of new plugins, features, performance improvements and bug
+    fixes
 
 Major new features and changes
 
 Noteworthy new features and API
 
--   this section will be filled in in due course
+-   gst_element_get_request_pad() has been deprecated in favour of the
+    newly-added gst_element_request_pad_simple() which does the exact
+    same thing but has a less confusing name that hopefully makes clear
+    that the function request a new pad rather than just retrieves an
+    already-existing request pad.
+
+Development in GitLab was switched to a single git repository containing all the modules
+
+The GStreamer multimedia framework is a set of libraries and plugins
+split into a number of distinct modules which are released independently
+and which have so far been developed in separate git repositories in
+freedesktop.org GitLab.
+
+In addition to these separate git repositories there was a gst-build
+module that would use the Meson build systems’s subproject feature to
+download each individual module and then build everything in one go. It
+would also provide an uninstalled development environment that made it
+easy to work on GStreamer and use or test versions other than the
+system-installed GStreamer version.
+
+All of these modules have now (as of 28 September 2021) been merged into
+a single git repository (“Mono repository” or “monorepo”) which should
+simplify development workflows and continuous integration, especially
+where changes need to be made to multiple modules at once.
+
+This mono repository merge will primarily affect GStreamer developers
+and contributors and anyone who has workflows based on the GStreamer git
+repositories.
+
+The Rust bindings and Rust plugins modules have not been merged into the
+mono repository at this time because they follow a different release
+cycle.
+
+The mono repository lives in the existing GStreamer core git repository
+in GitLab in the new main branch and all future development will happen
+on this branch.
+
+Modules will continue to be released as separate tarballs.
+
+For more details, please see the GStreamer mono repository FAQ.
+
+GstPlay: new high-level playback library replacing GstPlayer
+
+-   GstPlay is a new high-level playback library that replaces the older
+    GstPlayer API. It is basically the same API as GstPlayer but
+    refactored to use bus messages for application notifications instead
+    of GObject signals. There is still a signal adapter object for those
+    who prefer signals. Since the existing GstPlayer API is already in
+    use in various applications, it didn’t seem like a good idea to
+    break it entirely. Instead a new API was added, and it is expected
+    that this new GstPlay API will be moved to gst-plugins-base in
+    future.
+
+-   The existing GstPlayer API is scheduled for deprecation and will be
+    removed at some point in the future (e.g. in GStreamer 1.24), so
+    application developers are urged to migrate to the new GstPlay API
+    at their earliest convenience.
+
+WebM alpha decoding
+
+-   Implement WebM alpha decoding (VP8/VP9 with alpha), which required
+    support and additions in various places. This is supported both with
+    software decoders and hardware-accelerated decoders.
+
+-   VP8/VP9 don’t support alpha components natively in the codec, so the
+    way this is implemented in WebM is by encoding the alpha plane with
+    transparency data as a separate VP8/VP9 stream. Inside the WebM
+    container (a variant of Matroska) this is coded as a single video
+    track with the “normal” VP8/VP9 video data making up the main video
+    data and each frame of video having an encoded alpha frame attached
+    to it as extra data ("BlockAdditional").
+
+-   matroskademux has been extended extract this per-frame alpha side
+    data and attach it in form of a GstVideoCodecAlphaMeta to the
+    regular video buffers. Note that this new meta is specific to this
+    VP8/VP9 alpha support and can’t be used to just add alpha support to
+    other codecs that don’t support it. Lastly, matroskademux also
+    advertises the fact that the streams contain alpha in the caps.
+
+-   The new codecalpha plugin contains various bits of infrastructure to
+    support autoplugging and debugging:
+
+    -   codecalphademux splits out the alpha stream from the metas on
+        the regular VP8/VP9 buffers
+    -   alphacombine takes two decoded raw video streams (one alpha, one
+        the regular video) and combines it into a video stream with
+        alpha
+    -   vp8alphadecodebin + vp9alphadecodebin are wrapper bins that use
+        the regular vp8dec and vp9dec software decoders to decode
+        regular and alpha streams and combine them again. To decodebin
+        these look like regular decoders which ju
+    -   The V4L2 CODEC plugin has stateless VP8/VP9 decoders that can
+        decode both alpha and non-alpha stream with a single decoder
+        instance
+
+-   A new AV12 video format was added which is basically NV12 with an
+    alpha plane, which is more convenient for many hardware-accelerated
+    decoders.
+
+-   Watch Nicolas Dufresne’s LCA 2022 talk “Bringing WebM Alpha support
+    to GStreamer” for all the details and a demo.
+
+RTP Header Extensions Base Class and Automatic Header Extension Handling in RTP Payloaders and Depayloaders
+
+-   RTP Header Extensions are specified in RFC 5285 and provide a way to
+    add small pieces of data to RTP packets in between the RTP header
+    and the RTP payload. This is often used for per-frame metadata,
+    extended timestamps or other application-specific extra data. There
+    are several commonly-used extensions specified in various RFCs, but
+    senders are free to put any kind of data in there, as long as sender
+    and receiver both know what that data is. Receivers that don’t know
+    about the header extensions will just skip the extra data without
+    ever looking at it. These header extensions can often be combined
+    with any kind of payload format, so may need to be supported by many
+    RTP payloader and depayloader elements.
+
+-   Inserting and extracting RTP header extension data has so far been a
+    bit inconvenient in GStreamer: There are functions to add and
+    retrieve RTP header extension data from RTP packets, but nothing
+    works automatically, even for common extensions. People would have
+    to do the insertion/extraction either in custom elements
+    before/after the RTP payloader/depayloader, or inside pad probes,
+    which isn’t very nice.
+
+-   This release adds various pieces of new infrastructure for generic
+    RTP header extension handling, as well as some implementations for
+    common extensions:
+
+    -   GstRTPHeaderExtension is a new helper base class for reading and
+        writing RTP header extensions. Nominally this subclasses
+        GstElement, but only so these extensions are stored in the
+        registry where they can be looked up by URI or name. They don’t
+        have pads and don’t get added to the pipeline graph as an
+        element.
+
+    -   "add-extension" and "clear-extension" action signals on RTP
+        payloaders and depayloaders for manual extension management
+
+    -   The "request-extension" signal will be emitted if an extension
+        is encountered that requires explicit mapping by the application
+
+    -   new "auto-header-extension" property on RTP payloaders and
+        depayloaders for automatic handling of known header extensions.
+        This is enabled by default. The extensions must be signalled via
+        caps / SDP.
+
+    -   RTP header extension implementations:
+
+        -   rtphdrextclientaudiolevel: Client-to-Mixer Audio Level
+            Indication (RFC 6464) (also see below)
+        -   rtphdrextcolorspace: Color Space extension, extends RTP
+            packets with color space and high dynamic range (HDR)
+            information
+        -   rtphdrexttwcc: Transport Wide Congestion Control support
+
+-   gst_rtp_buffer_remove_extension_data() is a new helper function to
+    remove an RTP header extension from an RTP buffer
+
+-   The existing gst_rtp_buffer_set_extension_data() now also supports
+    shrinking the extension data in size
+
+AppSink and AppSrc improvements
+
+-   appsink: new API to pull events out of appsink in addition to
+    buffers and buffer lists.
+
+    There was previously no way for users to receive incoming events
+    from appsink properly serialised with the data flow, even if they
+    are serialised events. The reason for that is that the only way to
+    intercept events was via a pad probe on the appsink sink pad, but
+    there is also internal queuing inside of appsink, so it’s difficult
+    to ascertain the right order of everything in all cases.
+
+    There is now a new "new-serialized-event" signal which will be
+    emitted when there’s a new event pending (just like the existing
+    "new-sample" signal). The "emit-signals" property must be set to
+    TRUE in order to activate this (but it’s also fine to just pull from
+    the application thread without using the signals).
+
+    gst_app_sink_pull_object() and gst_app_sink_try_pull_object() can be
+    used to pull out either an event or a new sample carrying a buffer
+    or buffer list, whatever is next in the queue.
+
+    EOS events will be filtered and will not be returned. EOS handling
+    can be done the ususal way, same as with _pull_sample().
+
+-   appsrc: allow configuration of internal queue limits in time and
+    buffers and add leaky mode.
+
+    There is internal queuing inside appsrc so the application thread
+    can push data into the element which will then be picked up by the
+    source element’s streaming thread and pushed into the pipeline from
+    that streaming thread. This queue is unlimited by default and until
+    now it was only possible to set a maximum size limit in bytes. When
+    that byte limit is reached, the pushing thread (application thread)
+    would be blocked until more space becomes available.
+
+    A limit in bytes is not particularly useful for many use cases, so
+    now it is possible to also configure limits in time and buffers
+    using the new "max-time" and "max-buffers" properties. Of course
+    there are also matching new read-only"current-level-buffers" and
+    "current-level-time properties" properties to query the current fill
+    level of the internal queue in time and buffers.
+
+    And as if that wasn’t enough the internal queue can also be
+    configured as leaky using the new "leaky-type" property. That way
+    when the queue is full the application thread won’t be blocked when
+    it tries to push in more data, but instead either the new buffer
+    will be dropped or the oldest data in the queue will be dropped.
+
+Better string serialization of nested GstCaps and GstStructures
+
+-   New string serialisation format for structs and caps that can handle
+    nested structs and caps properly by using brackets to delimit nested
+    items (e.g. some-struct, some-field=[nested-struct, nested=true]).
+    Unlike the default format the new variant can also support more than
+    one level of nesting. For backwards-compatibility reasons the old
+    format is still output by default when serialising caps and structs
+    using the existing API. The new functions gst_caps_serialize() and
+    gst_structure_serialize() can be used to output strings in the new
+    format.
+
+Convenience API for custom GstMetas
+
+-   New convenience API to register and create custom GstMetas:
+    gst_meta_register_custom() and gst_buffer_add_custom_meta(). Such
+    custom meta is backed by a GstStructure and does not require that
+    users of the API expose their GstMeta implementation as public API
+    for other components to make use of it. In addition, it provides a
+    simpler interface by ignoring the impl vs. api distinction that the
+    regular API exposes. This new API is meant to be the meta
+    counterpart to custom events and messages, and to be more convenient
+    than the lower-level API when the absolute best performance isn’t a
+    requirement. The reason it’s less performant than a “proper” meta is
+    that a proper meta is just a C struct in the end whereas this goes
+    through the GstStructure API which has a bit more overhead, which
+    for most scenarios is negligible however. This new API is useful for
+    experimentation or proprietary metas, but also has some limitations:
+    it can only be used if there’s a single producer of these metas;
+    it’s not allowed to register the same custom meta multiple times or
+    from multiple places.
+
+Additional Element Properties on Encoding Profiles
+
+-   GstEncodingProfile: The new "element-properties" and
+    gst_encoding_profile_set_element_properties() API allows
+    applications to set additional element properties on encoding
+    profiles to configure muxers and encoders. So far the encoding
+    profile template was the only place where this could be specified,
+    but often what applications want to do is take a ready-made encoding
+    profile shipped by GStreamer or the application and then tweak the
+    settings on top of that, which is now possible with this API. Since
+    applications can’t always know in advance what encoder element will
+    be used in the end, it’s even possible to specify properties on a
+    per-element basis.
+
+    Encoding Profiles are used in the encodebin, transcodebin and
+    camerabin elements and APIs to configure output formats (containers
+    and elementary streams).
+
+Audio Level Indication Meta for RFC 6464
+
+-   New GstAudioLevelMeta containing Audio Level Indication as per RFC
+    6464
+
+-   The level element has been updated to add GstAudioLevelMeta on
+    buffers if the "audio-level-meta" property is set to TRUE. This can
+    then in turn be picked up by RTP payloaders to signal the audio
+    level to receivers through RTP header extensions (see above).
+
+-   New Client-to-Mixer Audio Level Indication (RFC6464) RTP Header
+    Extension which should be automatically created and used by RTP
+    payloaders and depayloaders if their "auto-header-extension"
+    property is enabled and if the extension is part of the RTP caps.
+
+Automatic packet loss, data corruption and keyframe request handling for video decoders
+
+-   The GstVideoDecoder base class has gained various new APIs to
+    automatically handle packet loss and data corruption better by
+    default, especially in RTP, RTSP and WebRTC streaming scenarios, and
+    to give subclasses more control about how they want to handle
+    missing data:
+
+    -   Video decoder subclasses can mark output frames as corrupted via
+        the new GST_VIDEO_CODEC_FRAME_FLAG_CORRUPTED flag
+
+    -   A new "discard-corrupted-frames" property allows applications to
+        configure decoders so that corrupted frames are directly
+        discarded instead of being forwarded inside the pipeline. This
+        is a replacement for the "output-corrupt" property of the FFmpeg
+        decoders.
+
+    -   RTP depayloaders can now signal to decoders that data is missing
+        when sending GAP events for lost packets. GAP events can be sent
+        for various reason in a GStreamer pipeline. Often they are just
+        used to let downstream elements know that there isn’t a buffer
+        available at the moment, so downstream elements can move on
+        instead of waiting for one. They are also sent by RTP
+        depayloaders in the case that packets are missing, however, and
+        so far a decoder was not able to differentiate the two cases.
+        This has been remedied now: GAP events can be decorated with
+        gst_event_set_gap_flags() and GST_GAP_FLAG_MISSING_DATA to let
+        decoders now what happened, and decoders can then use that in
+        some cases to handle missing data better.
+
+    -   The GstVideoDecoder::handle_missing_data vfunc was added to
+        inform subclasses about packet loss or missing data and let them
+        handle it in their own way if they like.
+
+    -   gst_video_decoder_set_needs_sync_point() lets subclasses signal
+        that they need the stream to start with a sync point. If
+        enabled, the base class will discard all non-sync point frames
+        in the beginning and after a flush and does not pass them to the
+        subclass. Furthermore, if the first frame is not a sync point,
+        the base class will try and request a sync frame from upstream
+        by sending a force-key-unit event (see next items).
+
+    -   New "automatic-request-sync-points" and
+        "automatic-request-sync-point-flags" properties to automatically
+        request sync points when needed, e.g. on packet loss or if the
+        first frame is not a keyframe. Applications may want to enable
+        this on decoders operating in e.g. RTP/WebRTC/RTSP receiver
+        pipelines.
+
+    -   The new "min-force-key-unit-interval" property can be used to
+        ensure there’s a minimal interval between keyframe requests to
+        upstream (and/or the sender) and we’re not flooding the sender
+        with key unit requests.
+
+    -   gst_video_decoder_request_sync_point() allows subclasses to
+        request a new sync point (e.g. if they choose to do their own
+        missing data handling). This will still honour the
+        "min-force-key-unit-interval" property if set.
+
+Improved support for custom minimal GStreamer builds
+
+-   Element registration and registration of other plugin features
+    inside plugin init functions has been improved in order to
+    facilitate minimal custom GStreamer builds.
+
+-   A number of new macros have been added to declare and create
+    per-element and per-pluginfeature register functions in all plugins,
+    and then call those from the per-plugin plugin_init functions:
+
+    -   GST_ELEMENT_REGISTER_DEFINE,
+        GST_DEVICE_PROVIDER_REGISTER_DEFINE,
+        GST_DYNAMIC_TYPE_REGISTER_DEFINE, GST_TYPE_FIND_REGISTER_DEFINE
+        for the actual registration call with GStreamer
+    -   GST_ELEMENT_REGISTER, GST_DEVICE_PROVIDER_REGISTER,
+        GST_DYNAMIC_TYPE_REGISTER, GST_PLUGIN_STATIC_REGISTER,
+        GST_TYPE_FIND_REGISTER to call the registration function defined
+        by the REGISTER_DEFINE macro
+    -   GST_ELEMENT_REGISTER_DECLARE,
+        GST_DEVICE_PROVIDER_REGISTER_DECLARE,
+        GST_DYNAMIC_TYPE_REGISTER_DECLARE,
+        GST_TYPE_FIND_REGISTER_DECLARE to declare the registration
+        function defined by the REGISTER_DEFINE macro
+    -   and various variants for advanced use cases.
+
+-   This means that applications can call the per-element and
+    per-pluginfeature registration functions for only the elements they
+    need instead of registering plugins as a whole with all kinds of
+    elements that may not be required (e.g. encoder and decoder instead
+    of just decoder). In case of static linking all unused functions and
+    their dependencies would be removed in this case by the linker,
+    which helps minimise binary size for custom builds.
+
+-   gst_init() will automatically call a gst_init_static_plugins()
+    function if one exists.
+
+-   See the GStreamer static build documentation and Stéphane’s blog
+    post Generate a minimal GStreamer build, tailored to your needs for
+    more details.
 
 New elements
 
--   this section will be filled in in due course
+-   New aesdec and aesenc elements for AES encryption and decryption in
+    a custom format.
+
+-   New encodebin2 element with dynamic/sometimes source pads in order
+    to support the option of doing the muxing outside of encodebin,
+    e.g. in combination with a splitmuxsink.
+
+-   New fakeaudiosink and videocodectestsink elements for testing and
+    debugging (see below for more details)
+
+-   rtpisacpay, rtpisacdepay: new RTP payloader and depayloader for iSAC
+    audio codec
+
+-   rtpst2022-1-fecdec, rtpst2022-1-fecenc: new elements providing SMPTE
+    2022-1 2-D Forward Error Correction. More details in Mathieu’s blog
+    post.
+
+-   isac: new plugin wrapping the Internet Speech Audio Codec reference
+    encoder and decoder from the WebRTC project.
+
+-   asio: plugin for Steinberg ASIO (Audio Streaming Input/Output) API
+
+-   gssrc, gssink: add source and sink for Google Cloud Storage
+
+-   onnx: new plugin to apply ONNX neural network models to video
+
+-   openaptx: aptX and aptX-HD codecs using libopenaptx (v0.2.0)
+
+-   qroverlay, debugqroverlay: new elements that allows overlaying data
+    on top of video in form of a QR code
+
+-   cvtracker: new OpenCV-based tracker element
+
+-   av1parse, vp9parse: new parsers for AV1 and VP9 video
+
+-   va: work on the new VA-API plugin implementation for
+    hardware-accelerated video decoding and encoding has continued at
+    pace, with various new decoders and filters having joined the
+    initial vah264dec:
+
+    -   vah265dec: VA-API H.265 decoder
+    -   vavp8dec: VA-API VP8 decoder
+    -   vavp9dec: VA-API VP9 decoder
+    -   vaav1dec: VA-API AV1 decoder
+    -   vampeg2dec: VA-API MPEG-2 decoder
+    -   vadeinterlace: : VA-API deinterlace filter
+    -   vapostproc: : VA-API postproc filter (color conversion,
+        resizing, cropping, color balance, video rotation, skin tone
+        enhancement, denoise, sharpen)
+
+    See Víctor’s blog post “GstVA in GStreamer 1.20” for more details
+    and what’s coming up next.
+
+-   vaapiav1dec: new AV1 decoder element (in gstreamer-vaapi)
+
+-   msdkav1dec: hardware-accelerated AV1 decoder using the Intel Media
+    SDK / oneVPL
+
+-   nvcodec plugin for NVIDIA NVCODEC API for hardware-accelerated video
+    encoding and decoding:
+
+    -   cudaconvert, cudascale: new CUDA based video color space convert
+        and rescale elements
+    -   cudaupload, cudadownload: new helper elements for memory
+        transfer between CUDA and system memory spaces
+    -   nvvp8sldec, nvvp9sldec: new GstCodecs-based VP8/VP9 decoders
+
+-   Various new hardware-accelerated elements for Windows:
+
+    -   d3d11screencapturesrc: new desktop capture element, including a
+        GstDeviceProvider implementation to enumerate/select target
+        monitors for capture.
+    -   d3d11av1dec and d3d11mpeg2dec: AV1 and MPEG-2 decoders
+    -   d3d11deinterlace: deinterlacing filter
+    -   d3d11compositor: video composing element
+    -   see Windows section below for more details
+
+-   new Rust plugins:
+
+    -   audiornnoise: Removes noise from an audio stream
+    -   awstranscribeparse: Parses AWS audio transcripts into timed text
+        buffers
+    -   ccdetect: Detects if valid closed captions are present in a
+        closed captions stream
+    -   cea608tojson: Converts CEA-608 Closed Captions to a JSON
+        representation
+    -   cmafmux: CMAF fragmented MP4 muxer
+    -   dashmp4mux: DASH fragmented MP4 muxer
+    -   isofmp4mux: ISO fragmented MP4 muxer
+    -   ebur128level: EBU R128 Loudness Level Measurement
+    -   ffv1dec: FFV1 video decoder
+    -   gtk4paintablesink: GTK4 video sink, which provides a
+        GdkPaintable that can be rendered in various widgets
+    -   hlssink3: HTTP Live Streaming sink
+    -   hrtfrender: Head-Related Transfer Function (HRTF) renderer
+    -   hsvdetector: HSV colorspace detector
+    -   hsvfilter: HSV colorspace filter
+    -   jsongstenc: Wraps buffers containing any valid top-level JSON
+        structures into higher level JSON objects, and outputs those as
+        ndjson
+    -   jsongstparse: Parses ndjson as output by jsongstenc
+    -   jsontovtt: converts JSON to WebVTT subtitles
+    -   regex: Applies regular expression operations on text
+    -   roundedcorners: Adds rounded corners to video
+    -   spotifyaudiosrc: Spotify source
+    -   textahead: Display upcoming text buffers ahead (e.g. for
+        Karaoke)
+    -   transcriberbin: passthrough bin that transcribes raw audio to
+        closed captions using awstranscriber and puts the captions as
+        metas onto the video
+    -   tttojson: Converts timed text to a JSON representation
+    -   uriplaylistbin: Playlist source bin
+    -   webpdec-rs: WebP image decoder with animation support
+
+-   New plugin codecalpha with elements to assist with WebM Alpha
+    decoding
+
+    -   codecalphademux: Split stream with GstVideoCodecAlphaMeta into
+        two streams
+    -   alphacombine: Combine two raw video stream (I420 or NV12) as one
+        stream with alpha channel (A420 or AV12)
+    -   vp8alphadecodebin: A bin to handle software decoding of VP8 with
+        alpha
+    -   vp9alphadecodebin: A bin to handle software decoding of VP9 with
+        alpha
+
+-   New hardware accelerated elements for Linux:
+
+    -   v4l2slmpeg2dec: Support for Linux Stateless MPEG2 decoders
+    -   v4l2slvp9dec: Support for Linux Stateless VP9 decoders
+    -   v4l2slvp8alphadecodebin: Support HW accelerated VP8 with alpha
+        layer decoding
+    -   v4l2slvp9alphadecodebin: Support HW accelerated VP9 with alpha
+        layer decoding
 
 New element features and additions
 
--   this section will be filled in in due course
+-   assrender: handle more font mime types; better interaction with
+    matroskademux for embedded fonts
+
+-   audiobuffersplit: Add support for specifying output buffer size in
+    bytes (not just duration)
+
+-   audiolatency: new "samplesperbuffer" property so users can configure
+    the number of samples per buffer. The default value is 240 samples
+    which is equivalent to 5ms latency with a sample rate of 48000,
+    which might be larger than actual buffer size of audio capture
+    device.
+
+-   audiomixer, audiointerleave, GstAudioAggregator: now keep a count of
+    samples that are dropped or processed as statistic and can be made
+    to post QoS messages on the bus whenever samples are dropped by
+    setting the "qos-messages" property on input pads.
+
+-   audiomixer, compositor: improved handling of new inputs added at
+    runtime. New API was added to the GstAggregator base class to allow
+    subclasses to opt into an aggregation mode where inactive pads are
+    ignored when processing input buffers
+    (gst_aggregator_set_ignore_inactive_pads(),
+    gst_aggregator_pad_is_inactive()). An “inactive pad” in this context
+    is a pad which, in live mode, hasn’t yet received a first buffer,
+    but has been waited on at least once. What would happen usually in
+    this case is that the aggregator would wait for data on this pad
+    every time, up to the maximum configured latency. This would
+    inadvertently push mixer elements in live mode to the configured
+    latency envelope and delay processing when new inputs are added at
+    runtime until these inputs have actually produced data. This is
+    usually undesirable. With this new API, new inputs can be added
+    (requested) and configured and they won’t delay the data processing.
+    Applications can opt into this new behaviour by setting the
+    "ignore-inactive-pads" property on compositor, audiomixer or other
+    GstAudioAggregator-based elements.
+
+-   cccombiner: implement “scheduling” of captions. So far cccombiner’s
+    behaviour was essentially that of a funnel: it strictly looked at
+    input timestamps to associate together video and caption buffers.
+    Now it will try to smoothly schedule caption buffers in order to
+    have exactly one per output video buffer. This might involve
+    rewriting input captions, for example when the input is CDP then
+    sequence counters are rewritten, time codes are dropped and
+    potentially re-injected if the input video frame had a time code
+    meta. This can also lead to the input drifting from synchronization,
+    when there isn’t enough padding in the input stream to catch up. In
+    that case the element will start dropping old caption buffers once
+    the number of buffers in its internal queue reaches a certain limit
+    (configurable via the "max-scheduled" property). The new original
+    funnel-like behaviour can be restored by setting the "scheduling"
+    property to FALSE.
+
+-   ccconverter: new "cdp-mode" property to specify which sections to
+    include in CDP packets (timecode, CC data, service info). Various
+    software, including ffmpeg’s Decklink support, fails parsing CDP
+    packets that contain anything but CC data in the CDP packets.
+
+-   clocksync: new "sync-to-first" property for automatic timestamp
+    offset setup: if set clocksync will set up the "ts-offset" value
+    based on the first buffer and the pipeline’s running time when the
+    first buffer arrived. The newly configured "ts-offset" in this case
+    would be the value that allows outputting the first buffer without
+    waiting on the clock. This is useful for example to feed a non-live
+    input into an already-running pipeline.
+
+-   compositor:
+
+    -   multi-threaded input conversion and compositing. Set the
+        "max-threads" property to activate this.
+    -   new "sizing-policy" property to support display aspect ratio
+        (DAR)-aware scaling. By default the image is scaled to fill the
+        configured destination rectangle without padding and without
+        keeping the aspect ratio. With sizing-policy=keep-aspect-ratio
+        the input image is scaled to fit the destination rectangle
+        specified by GstCompositorPad:{xpos, ypos, width, height}
+        properties preserving the aspect ratio. As a result, the image
+        will be centered in the destination rectangle with padding if
+        necessary.
+    -   new "zero-size-is-unscaled" property on input pads. By default
+        pad width=0 or pad height=0 mean that the stream should not be
+        scaled in that dimension. But if the "zero-size-is-unscaled"
+        property is set to FALSE a width or height of 0 is instead
+        interpreted to mean that the input image on that pad should not
+        be composited, which is useful when creating animations where an
+        input image is made smaller and smaller until it disappears.
+    -   improved handling of new inputs at runtime via
+        "ignore-inactive-pads"property (see above for details)
+    -   allow output format with alpha even if none of the inputs have
+        alpha (also glvideomixer and other GstVideoAggregator
+        subclasses)
+
+-   dashsink: add h265 codec support and signals for allowing custom
+    playlist/fragment output
+
+-   decodebin3:
+
+    -   improved decoder selection, especially for hardware decoders
+    -   make input activation “atomic” when adding inputs dynamically
+    -   better interleave handling: take into account decoder latency
+        for interleave size
+
+-   decklink:
+
+    -   Updated DeckLink SDK to 11.2 to support DeckLink 8K Pro
+    -   decklinkvideosrc:
+        -   More accurate and stable capture timestamps: use the
+            hardware reference clock time when the frame was finished
+            being captured instead of a clock time much further down the
+            road.
+        -   Automatically detect widescreen vs. normal NTSC/PAL
+
+-   encodebin:
+
+    -   add “smart encoding” support for H.265, VP8 and VP9 (i.e. only
+        re-encode where needed and otherwise pass through encoded video
+        as-is).
+    -   H264/H265 smart encoding improvements: respect user-specified
+        stream-format, but if not specified default to avc3/hvc1 with
+        in-band SPS/PPS/VPS signalling for more flexibility.
+    -   new encodebin2 element with dynamic/sometimes source pads in
+        order to support the option of doing the muxing outside of
+        encodebin, e.g. in combination with splitmuxsink.
+    -   add APIs to set element properties on encoding profiles (see
+        below)
+
+-   errorignore: new "ignore-eos" property to also ignore FLOW_EOS from
+    downstream elements
+
+-   giosrc: add support for growing source files: applications can
+    specify that the underlying file being read is growing by setting
+    the "is-growing" property. If set, the source won’t EOS when it
+    reaches the end of the file, but will instead start monitoring it
+    and will start reading data again whenever a change is detected. The
+    new "waiting-data" and "done-waiting-data" signals keep the
+    application informed about the current state.
+
+-   gtksink, gtkglsink:
+
+    -   scroll event support: forwarded as navigation events into the
+        pipeline
+    -   "video-aspect-ratio-override" property to force a specific
+        aspect ratio
+    -   "rotate-method" property and support automatic rotation based on
+        image tags
+
+-   identity: new "stats" property allows applications to retrieve the
+    number of bytes and buffers that have passed through so far.
+
+-   interlace: add support for more formats, esp 10-bit, 12-bit and
+    16-bit ones
+
+-   jack: new "low-latency" property for automatic latency-optimized
+    setting and "port-names" property to select ports explicitly
+
+-   jpegdec: support output conversion to RGB using libjpeg-turbo (for
+    certain input files)
+
+-   line21dec:
+
+    -   "mode" property to control whether and how detected closed
+        captions should be inserted in the list of existing close
+        caption metas on the input frame (if any): add, drop, or
+        replace.
+    -   "ntsc-only" property to only look for captions if video has NTSC
+        resolution
+
+-   line21enc: new "remove-caption-meta" to remove metas from output
+    buffers after encoding the captions into the video data; support for
+    CDP closed captions
+
+-   matroskademux, matroskamux: Add support for ffv1, a lossless
+    intra-frame video coding format.
+
+-   matroskamux: accept in-band SPS/PPS/VPS for H264 and H265
+    (i.e. stream-format avc3 and hev1) which allows on-the-fly
+    profile/level/resolution changes.
+
+-   matroskamux: new "cluster-timestamp-offset" property, useful for use
+    cases where the container timestamps should map to some absolute
+    wall clock time, for example.
+
+-   rtpsrc: add "caps" property to allow explicit setting of the caps
+    where needed
+
+-   mpegts: support SCTE-35 passthrough via new "send-scte35-events"
+    property on MPEG-TS demuxer tsdemux. When enabled, SCTE 35 sections
+    (eg ad placement opportunities) are forwarded as events donwstream
+    where they can be picked up again by mpegtsmux. This required a
+    semantic change in the SCTE-35 section API: timestamps are now in
+    running time instead of muxer pts.
+
+-   tsdemux: Handle PCR-less MPEG-TS streams; more robust timestamp
+    handling in certain corner cases and for poorly muxed streams.
+
+-   mpegtsmux:
+
+    -   More conformance improvements to make MPEG-TS analyzers happy:
+        -   PCR timing accuracy: Improvements to the way mpegtsmux
+            outputs PCR observations in CBR mode, so that a PCR
+            observation is always inserted when needed, so that we never
+            miss the configured pcr-interval, as that triggers various
+            MPEG-TS analyser errors.
+        -   Improved PCR/SI scheduling
+    -   Don’t write PCR until PAT/PMT are output to make sure streams
+        start cleanly with a PAT/PMT.
+    -   Allow overriding the automatic PMT PID selection via
+        application-supplied PMT_%d fields in the prog-map
+        structure/property.
+
+-   mp4mux:
+
+    -   new "first-moov-then-finalise" mode for fragmented output where
+        the output will start with a self-contained moov atom for the
+        first fragment, and then produce regular fragments. Then at the
+        end when the file is finalised, the initial moov is invalidated
+        and a new moov is written covering the entire file. This way the
+        file is a “fragmented mp4” file while it is still being written
+        out, and remains playable at all times, but at the end it is
+        turned into a regular mp4 file (with former fragment headers
+        remaining as unused junk data in the file).
+    -   support H.264 avc3 and H.265 hvc1 stream formats as input where
+        the codec data is signalled in-band inside the bitstream instead
+        of caps/file headers.
+    -   support profile/level/resolution changes for H264/H265 input
+        streams (i.e. codec data changing on the fly). Each codec_data
+        is put into its own SampleTableEntry inside the stsd, unless the
+        input is in avc3 stream format in which case it’s written
+        in-band an not in the headers.
+
+-   multifilesink: new ""min-keyframe-distance"" property to make
+    minimum distance between keyframes in next-file=key-frame mode
+    configurable instead of hard-coding it to 10 seconds.
+
+-   mxfdemux has seen a big refactoring to support non-frame wrappings
+    and more accurate timestamp/seek handling for some formats
+
+-   msdk plugin for hardware-accelerated video encoding and decoding
+    using the Intel Media SDK:
+
+    -   oneVPL support (Intel oneAPI Video Processing Library)
+    -   AV1 decoding support
+    -   H.264 decoder now supports constrained-high and progressive-high
+        profiles
+    -   H.264 encoder:
+        -   more configuration options (properties):
+            "intra-refresh-type", "min-qp" , "max-qp", "p-pyramid",
+            "dblk-idc"
+    -   H.265 encoder:
+        -   can output main-still-picture profile
+        -   now inserts HDR SEIs (mastering display colour volume and
+            content light level)
+        -   more configuration options (properties):
+            "intra-refresh-type", "min-qp" , "max-qp", "p-pyramid",
+            "b-pyramid", "dblk-idc", "transform-skip"
+        -   support for RGB 10bit format
+    -   External bitrate control in encoders
+    -   Video post proc element msdkvpp gained support for 12-bit pixel
+        formats P012_LE, Y212_LE and Y412_LE
+
+-   nvh264sldec: interlaced stream support
+
+-   openh264enc: support main, high, constrained-high and
+    progressive-high profiles
+
+-   openjpeg: support for multithreaded decoding and encoding
+
+-   rtspsrc: now supports IPv6 also for tunneled mode (RTSP-over-HTTP);
+    new "ignore-x-server-reply" property to ignore the
+    x-server-ip-address server header reply in case of HTTP tunneling,
+    as it is often broken.
+
+-   souphttpsrc: Runtime compatibility support for libsoup2 and
+    libsoup3. libsoup3 is the latest major version of libsoup, but
+    libsoup2 and libsoup3 can’t co-exist in the same process because
+    there is no namespacing or versioning for GObject types. As a
+    result, it would be awkward if the GStreamer souphttpsrc plugin
+    linked to a specific version of libsoup, because it would only work
+    with applications that use the same version of libsoup. To make this
+    work, the soup plugin now tries to determine the libsoup version
+    used by the application (and its other dependencies) at runtime on
+    systems where GStreamer is linked dynamically. libsoup3 support is
+    still considered somewhat experimental at this point.
+
+-   srtsrc, srtsink: add signals for the application to accept/reject
+    incoming connections
+
+-   timeoverlay: new elapsed-running-time time mode which shows the
+    running time since the first running time (and each flush-stop).
+
+-   udpsrc: new timestamping mode to retrieve packet receive timestamps
+    from the kernel via socket control messages (SO_TIMESTAMPNS) on
+    supported platforms
+
+-   uritranscodebin: new setup-source and element-setup signals for
+    applications to configure elements used
+
+-   v4l2codecs plugin gained support for 4x4 and 32x32 tile formats
+    enabling some platforms or direct renders. Important memory usage
+    improvement.
+
+-   v4l2slh264dec now implements the final Linux uAPI as shipped on
+    Linux 5.11 and later.
+
+-   valve: add "drop-mode" property and provide two new modes of
+    operation: in drop-mode=forward-sticky-events sticky events
+    (stream-start, segment, tags, caps, etc.) are forwarded downstream
+    even when dropping is enabled; drop-mode=transform-to-gap will in
+    addition also convert buffers into gap events when dropping is
+    enabled, which lets downstream elements know that time is advancing
+    and might allow for preroll in many scenarios. By default all events
+    and all buffers are dropped when dropping is enabled, which can
+    cause problems with caps negotiation not progressing or branches not
+    prerolling when dropping is enabled.
+
+-   videocrop: support for many more pixel formats, e.g. planar YUV
+    formats with > 8bits and GBR* video formats; can now also accept
+    video not backed by system memory as long as downstream supports the
+    GstCropMeta
+
+-   videotestsrc: new smpte-rp-219 pattern for SMPTE75 RP-219 conformant
+    color bars
+
+-   vp8enc: finish support for temporal scalability: two new properties
+    ("temporal-scalability-layer-flags",
+    "temporal-scalability-layer-sync-flags") and a unit change on the
+    "temporal-scalability-target-bitrate" property (now expects bps);
+    also make temporal scalability details available to RTP payloaders
+    as buffer metadata.
+
+-   vp9enc: new properties to tweak encoder performance:
+
+    -   "aq-mode" to configure adaptive quantization modes
+    -   "frame-parallel-decoding" to configure whether to create a
+        bitstream that reduces decoding dependencies between frames
+        which allows staged parallel processing of more than one video
+        frames in the decoder. (Defaults to TRUE)
+    -   "row-mt", "tile-columns" and "tile-rows" so multithreading can
+        be enabled on a per-tile basis, instead of on a per tile-column
+        basis. In combination with the new "tile-rows" property, this
+        allows the encoder to make much better use of the available CPU
+        power.
+
+-   vp9dec, vp9enc: add support for 10-bit 4:2:0 and 4:2:2 YUV, as well
+    as 8-bit 4:4:4
+
+-   vp8enc, vp9enc now default to “good quality” for the deadline
+    property rather then “best quality”. Having the deadline set to best
+    quality causes the encoder to be absurdly slow, most real-life users
+    will prefer good-enough quality with better performance instead.
+
+-   wpesrc:
+
+    -   implement audio support: a new sometimes source pad will be
+        created for each audio stream created by the web engine.
+    -   move wpesrc to wpevideosrc and add a wrapper bin wpesrc to also
+        support audio
+    -   also handles web:// URIs now (same as cefsrc)
+    -   post messages with the estimated load progress on the bus
 
-Plugin and library moves
+-   x265enc: add negative DTS support, which means timestamps are now
+    offset by 1h same as with x264enc
+
+RTP Payloaders and Depayloaders
+
+-   rtpisacpay, rtpisacdepay: new RTP payloader and depayloader for iSAC
+    audio codec
+
+-   rtph264depay:
+
+    -   new "request-keyframe" property to make the depayloader
+        automatically request a new keyframe from the sender on packet
+        loss, consistent with the new property on rtpvp8depay.
+    -   new "wait-for-keyframe" property to make depayloader wait for a
+        new keyframe at the beginning and after packet loss (only
+        effective if the depayloader outputs AUs), consistent with the
+        existing property on rtpvp8depay.
+
+-   rtpopuspay, rtpopusdepay: support libwebrtc-compatible multichannel
+    audio in addition to the previously supported multichannel audio
+    modes
+
+-   rtpopuspay: add DTX (Discontinuous Transmission) support
+
+-   rtpvp8depay: new "request-keyframe" property to make the depayloader
+    automatically request a new keyframe from the sender on packet loss.
+
+-   rtpvp8pay: temporal scaling support
 
--   this section will be filled in in due course
+-   rtpvp9depay: Improved SVC handling (aggregate all layers)
+
+RTP Infrastructure
+
+-   rtpst2022-1-fecdec, rtpst2022-1-fecenc: new elements providing SMPTE
+    2022-1 2-D Forward Error Correction. More details in Mathieu’s blog
+    post.
+
+-   rtpreddec: BUNDLE support
+
+-   rtpredenc, rtpulpfecenc: add support for Transport-wide Congestion
+    Control (TWCC)
+
+-   rtpsession: new "twcc-feedback-interval" property to allow RTCP TWCC
+    reports to be scheduled on a timer instead of per marker-bit.
+
+Plugin and library moves
 
 -   There were no plugin moves or library moves in this cycle.
 
@@ -56,8 +998,6 @@ Plugin removals
 
 The following elements or plugins have been removed:
 
--   this section will be filled in in due course
-
 -   The ofa audio fingerprinting plugin has been removed. The MusicIP
     database has been defunct for years so this plugin is likely neither
     useful nor used by anyone.
@@ -71,135 +1011,575 @@ The following elements or plugins have been removed:
 
 Miscellaneous API additions
 
--   this section will be filled in in due course
+Core
+
+-   gst_buffer_new_memdup() is a convenience function for the
+    widely-used gst_buffer_new_wrapped(g_memdup(data,size),size)
+    pattern.
+
+-   gst_caps_features_new_single() creates a new single GstCapsFeatures,
+    avoiding the need to use the vararg function with NULL terminator
+    for simple cases.
+
+-   gst_element_type_set_skip_documentation() can be used by plugins to
+    signal that certain elements should not be included in the GStreamer
+    plugin documentation. This is useful for plugins where elements are
+    registered dynamically based on hardware capabilities and/or where
+    the available plugins and properties vary from system to system.
+    This is used in the d3d11 plugin for example to ensure that only the
+    list of default elements is advertised in the documentation.
+
+-   gst_type_find_suggest_empty_simple() is a new convenience function
+    for typefinders for cases where there’s only a media type and no
+    other fields.
+
+-   New API to create elements and set properties at construction time,
+    which is not only convenient, but also allows GStreamer elements to
+    have construct-only properties: gst_element_factory_make_full(),
+    gst_element_factory_make_valist(),
+    gst_element_factory_make_with_properties(),
+    gst_element_factory_create_full(),
+    gst_element_factory_create_valist(),
+    gst_element_factory_create_with_properties().
+
+-   GstSharedTaskPool: new “shared” task pool subclass with slightly
+    different default behaviour than the existing GstTaskPool which
+    would create unlimited number of threads for new tasks. The shared
+    taskpool creates up to N threads (default: 1) and then distributes
+    pending tasks to those threads round-robin style, and blocks if no
+    thread is available. It is possible to join tasks. This can be used
+    by plugins to implement simple multi-threaded processing and is used
+    for the new multi-threaded video conversion and compositing done in
+    GstVideoAggregator, videoconverter and compositor.
+
+Plugins Base Utils library
+
+-   GstDiscoverer:
+
+    -   gst_discoverer_container_info_get_tags() was added to retrieve
+        global/container tags (vs. per-stream tags). Per-Stream tags can
+        be retrieved via the existing
+        gst_discoverer_stream_info_get_tags().
+        gst_discoverer_info_get_tags(), which for many files returns a
+        confusing mix of stream and container tags, has been deprecated
+        in favour of the container/stream-specific functions.
+    -   gst_discoverer_stream_info_get_stream_number() returns a unique
+        integer identifier for a given stream within the given
+        GstDiscoverer context. (If this matches the stream number inside
+        the container bitstream that’s by coincidence and not by
+        design.)
+
+-   gst_pb_utils_get_caps_description_flags() can be used to query
+    whether certain caps represent a container, audio, video, image,
+    subtitles, tags, or something else. This only works for formats
+    known to GStreamer.
+
+-   gst_pb_utils_get_file_extension_from_caps() returns a possible file
+    extension for given caps.
+
+-   gst_codec_utils_h264_get_profile_flags_level(): Parses profile,
+    flags, and level from H264 AvcC codec_data. The format of H264 AVCC
+    extradata/sequence_header is documented in the ITU-T H.264
+    specification section 7.3.2.1.1 as well as in ISO/IEC 14496-15
+    section 5.3.3.1.2.
+
+-   gst_codec_utils_caps_get_mime_codec() to convert caps to a RFC 6381
+    compatible MIME codec string codec. Useful for providing the codecs
+    field inside the Content-Type HTTP header for containerized formats,
+    such as mp4 or matroska.
+
+GStreamer OpenGL integration library and plugins
+
+-   glcolorconvert: added suppport for converting the video formats
+    A420, AV12, BGR, BGRA, RGBP and BGRP.
+
+-   Added support to GstGLBuffer for persistent buffer mappings where a
+    Pixel Buffer Object (PBO) can be mapped by both the CPU and the GPU.
+    This removes a memcpy() when uploading textures or vertices
+    particularly when software decoders (e.g. libav) are direct
+    rendering into our memory. Improves transfer performance
+    significantly. Requires OpenGL 4.4, GL_ARB_buffer_storage or
+    GL_EXT_buffer_storage
+
+-   Added various helper functions for handling 4x4 matrices of affine
+    transformations as used by GstVideoAffineTransformationMeta.
+
+-   Add support to GstGLContext for allowing the application to control
+    the config (EGLConfig, GLXConfig, etc) used when creating the OpenGL
+    context. This allows the ability to choose between RGB16 or RGB10A2
+    or RGBA8 back/front buffer configurations that were previously
+    hardcoded. GstGLContext also supports retrieving the configuration
+    it was created with or from an externally provide OpenGL context
+    handle. This infrastructure is also used to create a compatible
+    config from an application/externally provided OpenGL context in
+    order to improve compatibility with other OpenGL frameworks and GUI
+    toolkits. A new environment variable GST_GL_CONFIG was also added to
+    be able to request a specific configuration from the command line.
+    Note: different platforms will have different functionality
+    available.
+
+-   Add support for choosing between EGL and WGL at runtime when running
+    on Windows. Previously this was a build-time switch. Allows use in
+    e.g. Gtk applications on Windows that target EGL/ANGLE without
+    recompiling GStreamer. gst_gl_display_new_with_type() can be used by
+    applications to choose a specific display type to use.
+
+-   Build fixes to explicitly check for Broadcom-specific libraries on
+    older versions of the Raspberry Pi platform. The Broadcom OpenGL ES
+    and EGL libraries have different filenames. Using the vc4 Mesa
+    driver on the Raspberry Pi is not affected.
+
+-   Added support to glupload and gldownload for transferring RGBA
+    buffers using the memory:NVMM available on the Nvidia Tegra family
+    of embedded devices.
+
+-   Added support for choosing libOpenGL and libGLX as used in a GLVND
+    environment on unix-based platforms. This allows using desktop
+    OpenGL and EGL without pulling in any GLX symbols as would be
+    required with libGL.
+
+Video library
+
+-   New raw video formats:
+
+    -   AV12 (NV12 with alpha plane)
+    -   RGBP and BGRP (planar RGB formats)
+    -   ARGB64 variants with specified endianness instead of host
+        endianness:
+        -   ARGB64_LE, ARGB64_BE
+        -   RGBA64_BE, RGBA64_LE
+        -   BGRA64_BE, BGRA64_LE
+        -   ABGR64_BE, ABGR64_LE
+
+-   gst_video_orientation_from_tag() is new convenience API to parse the
+    image orientation from a GstTagList.
+
+-   GstVideoDecoder subframe support (see below)
+
+-   GstVideoCodecState now also carries some HDR metadata
+
+-   Ancillary video data: implement transform functions for AFD/Bar
+    metas, so they will be forwarded in more cases
+
+MPEG-TS library
+
+This library only handles section parsing and such, see above for
+changes to the actual mpegtsmux and mpegtsdemux elements.
+
+-   many additions and improvements to SCTE-35 section parsing
+-   new API for fetching extended descriptors:
+    gst_mpegts_find_descriptor_with_extension()
+-   add support for SIT sections (Selection Information Tables)
+-   expose event-from-section constructor gst_event_new_mpegts_section()
+-   parse Audio Preselection Descriptor needed for Dolby AC-4
+
+GstWebRTC library + webrtcbin
+
+-   Change the way in which sink pads and transceivers are matched
+    together to support easier usage. If a pad is created without a
+    specific index (i.e. using sink_%u as the pad template), then an
+    available compatible transceiver will be searched for. If a specific
+    index is requested (i.e. sink_1) then if a transceiver for that
+    m-line already exists, that transceiver must match the new sink pad
+    request. If there is no transceiver available in either scenario, a
+    new transceiver is created. If a mixture of both sink_1 and sink_%u
+    requests result in an impossible situation, an error will be
+    produced at pad request time or from create offer/answer.
+
+-   webrtcbin now uses regular ICE nomination instead of libnice’s
+    default of aggressive ICE nomination. Regular ICE nomination is the
+    default recommended by various relevant standards and improves
+    connectivity in specific network scenarios.
+
+-   Add support for limiting the port range used for RTP with the
+    addition of the min-rtp-port and max-rtp-port properties on the ICE
+    object.
+
+-   Expose the SCTP transport as a property on webrtcbin to more closely
+    match the WebRTC specification.
+
+-   Added support for taking into account the data channel transport
+    state when determining the value of the "connection-state" property.
+    Previous versions of the WebRTC spec did not include the data
+    channel state when computing this value.
+
+-   Add configuration for choosing the size of the underlying sockets
+    used for transporting media data
+
+-   Always advertise support for the transport-cc RTCP feedback protocol
+    as rtpbin supports it. For full support, the configured caps (input
+    or through codec-preferences) need to include the relevant RTP
+    header extension.
+
+-   Numerous fixes to caps and media handling to fail-fast when an
+    incompatible situation is detected.
+
+-   Improved support for attaching the required media after a remote
+    offer has been set.
+
+-   Add support for dynamically changing the amount of FEC used for a
+    particular stream.
+
+-   webrtcbin now stops further SDP processing at the first error it
+    encounters.
+
+-   Completed support for either local or the remote closing a data
+    channel.
+
+-   Various fixes when performing BUNDLEing of the media streams in
+    relation to RTX and FEC usage.
+
+-   Add support for writing out QoS DSCP marking on outgoing packets to
+    improve reliability in some network scenarios.
+
+-   Improvements to the statistics returned by the get-stats signal
+    including the addition of the raw statistics from the internal
+    RTPSource, the TWCC stats when available.
+
+-   The webrtc library does not expose any objects anymore with public
+    fields. Instead properties have been added to replace that
+    functionality. If you are accessing such fields in your application,
+    switch to the corresponding properties.
+
+GstCodecs and Video Parsers
+
+-   Support for render delays to improve throughput across all CODECs
+    (used with NVDEC and V4L2).
+-   lots of improvements to parsers and the codec parsing decoder base
+    classes (H264, H265, VP8, VP9, AV1, MPEG-2) used for various
+    hardware-accelerate decoder APIs.
+
+Bindings support
+
+-   gst_allocation_params_new() allocates a GstAllocationParams struct
+    on the heap. This should only be used by bindings (and freed via
+    gst_allocation_params_free() then). In C code you would allocate
+    this on the stack and only init it in place.
+
+-   gst_debug_log_literal() can be used to log a string to the debug log
+    without going through any printf format expansion and associated
+    overhead. This is mostly useful for bindings such as the Rust
+    bindings which may have done their own formatting already .
+
+-   Provide non-inlined versions of refcounting APIs for various
+    GStreamer mini objects, so that they can be consumed by bindings
+    (e.g. gstreamer-sharp): gst_buffer_ref, gst_buffer_unref,
+    gst_clear_buffer, gst_buffer_copy, gst_buffer_replace,
+    gst_buffer_list_ref, gst_buffer_list_unref, gst_clear_buffer_list,
+    gst_buffer_list_copy, gst_buffer_list_replace, gst_buffer_list_take,
+    gst_caps_ref, gst_caps_unref, gst_clear_caps, gst_caps_replace,
+    gst_caps_take, gst_context_ref, gst_context_unref, gst_context_copy,
+    gst_context_replace, gst_event_replace, gst_event_steal,
+    gst_event_take, gst_event_ref, gst_event_unref, gst_clear_event,
+    gst_event_copy, gst_memory_ref, gst_memory_unref, gst_message_ref,
+    gst_message_unref, gst_clear_message, gst_message_copy,
+    gst_message_replace, gst_message_take, gst_promise_ref,
+    gst_promise_unref, gst_query_ref, gst_query_unref, gst_clear_query,
+    gst_query_copy, gst_query_replace, gst_query_take, gst_sample_ref,
+    gst_sample_unref, gst_sample_copy, gst_tag_list_ref,
+    gst_tag_list_unref, gst_clear_tag_list, gst_tag_list_replace,
+    gst_tag_list_take, gst_uri_copy, gst_uri_ref, gst_uri_unref,
+    gst_clear_uri.
+
+-   expose a GType for GstMiniObject
+
+-   gst_device_provider_probe() now returns non-floating device object
+
+API Deprecations
+
+-   gst_element_get_request_pad() has been deprecated in favour of the
+    newly-added gst_element_request_pad_simple() which does the exact
+    same thing but has a less confusing name that hopefully makes clear
+    that the function request a new pad rather than just retrieves an
+    already-existing request pad.
+
+-   gst_discoverer_info_get_tags(), which for many files returns a
+    confusing mix of stream and container tags, has been deprecated in
+    favour of the container-specific and stream-specific functions,
+    gst_discoverer_container_info_get_tags() and
+    gst_discoverer_stream_info_get_tags().
+
+-   gst_video_sink_center_rect() was deprecated in favour of the more
+    generic newly-added gst_video_center_rect().
+
+-   The GST_MEMORY_FLAG_NO_SHARE flag has been deprecated, as it tends
+    to cause problems and prevents sub-buffering. If pooling or lifetime
+    tracking is required, memories should be allocated through a custom
+    GstAllocator instead of relying on the lifetime of the buffers the
+    memories were originally attached to, which is fragile anyway.
+
+-   The GstPlayer high-level playback library is being replaced with the
+    new GstPlay library (see above). GstPlayer should be considered
+    deprecated at this point and will be marked as such in the next
+    development cycle. Applications should be ported to GstPlay.
+
+-   Gstreamer Editing Services: ges_video_transition_set_border(),
+    ges_video_transition_get_border()
+    ges_video_transition_set_inverted()
+    ges_video_transition_is_inverted() have been deprecated, use
+    ges_timeline_element_set_children_properties() instead.
 
 Miscellaneous performance, latency and memory optimisations
 
--   this section will be filled in in due course
+More video conversion fast paths
 
-Miscellaneous other changes and enhancements
+-   v210 ↔ I420, YV12, Y42B, UYVY and YUY2
+-   A420 → RGB
+
+Less jitter when waiting on the system clock
+
+-   Better system clock wait accuracy, less jitter: where available,
+    clock_nanosleep is used for higher accuracy for waits below 500
+    usecs, and waits below 2ms will first use the regular waiting system
+    and then clock_nanosleep for the remainder. The various wait
+    implementation have a latency ranging from 50 to 500+ microseconds.
+    While this is not a major issue when dealing with a low number of
+    waits per second (for ex: video), it does introduce a non-negligible
+    jitter for synchronisation of higher packet rate systems.
+
+Video decoder subframe support
 
--   this section will be filled in in due course
+-   The GstVideoDecoder base class gained API to process input at the
+    sub-frame level. That way video decoders can start decoding slices
+    before they have received the full input frame in its entirety (to
+    the extent this is supported by the codec, of course). This helps
+    with CPU utilisation and reduces latency.
 
-Tracing framework and debugging improvements
+-   This functionality is now being used in the OpenJPEG JPEG 2000
+    decoder, the FFmpeg H.264 decoder (in case of NAL-aligned input) and
+    the OpenMAX H.264/H.265 decoders (in case of NAL-aligned input).
 
--   this section will be filled in in due course
+Miscellaneous other changes and enhancements
+
+-   GstDeviceMonitor no longer fails to start just because one of the
+    device providers failed to start. That could happen for example on
+    systems where the pulseaudio device provider is installed, but
+    pulseaudio isn’t actually running but ALSA is used for audio
+    instead. In the same vein the device monitor now keeps track of
+    which providers have been started (via the new
+    gst_device_provider_is_started()) and only stops actually running
+    device providers when stopping the device monitor.
+
+-   On embedded systems it can be useful to create a registry that can
+    be shared and read by multiple processes running as different users.
+    It is now possible to set the new GST_REGISTRY_MODE environment
+    variable to specify the file mode for the registry file, which by
+    default is set to be only user readable/writable.
+
+-   GstNetClientClock will signal lost sync in case the remote time
+    resets (e.g. because device power cycles), by emitting the “synced”
+    signal with synced=FALSE parameter, so applications can take action.
+
+-   gst_value_deserialize_with_pspec() allows deserialization with a
+    hint for what the target GType should be. This allows for example
+    passing arrays of flags through the command line or
+    gst_util_set_object_arg(), eg: foo="<bar,bar+baz>".
+
+-   It’s now allowed to create an empty GstVideoOverlayComposition
+    without any rectangles by passing a NULL rectangle to
+    gst_video_overlay_composition_new(). This is useful for bindings and
+    simplifies application code in some places.
+
+Tracing framework, debugging and testing improvements
+
+-   New factories tracer to list loaded elements (and other plugin
+    features). This can be useful to collect a list of elements needed
+    for an application, which then in turn can be used to create a
+    tailored minimal GStreamer build that contains just the elements
+    needed and nothing else.
+-   New plugin-feature-loaded tracing hook for use by tracers like the
+    new factories tracer
+
+-   GstHarness: Add gst_harness_set_live() so that harnesses can be set
+    to non-live and return is-live=false in latency queries if needed.
+    Default behaviour is to always return is-live=true in latency
+    queries.
+
+-   navseek: new "hold-eos" property. When enabled, the element will
+    hold back an EOS event until the next keystroke (via navigation
+    events). This can be used to keep a video sink showing the last
+    frame of a video pipeline until a key is pressed instead of tearing
+    it down immediately on EOS.
+
+-   New fakeaudiosink element: mimics an audio sink and can be used for
+    testing and CI pipelines on systems where no audio system is
+    installed or running. It differs from fakesink in that it only
+    support audio caps and syncs to the clock by default like a normal
+    audio sink. It also implements the GstStreamVolume interface like
+    most audio sinks do.
+
+-   New videocodectestsink element for video codec conformance testing:
+    Calculates MD5 checksums for video frames and skips any padding
+    whilst doing so. Can optionally also write back the video data with
+    padding removed into a file for easy byte-by-byte comparison with
+    reference data.
 
 Tools
 
--   this section will be filled in in due course
+gst-inspect-1.0
+
+-   Can sort the list of plugins by passing --sort=name as command line
+    option
+
+gst-launch-1.0
+
+-   will now error out on top-level properties that don’t exist and
+    which were silently ignored before
+-   On Windows the high-resolution clock is enabled now, which provides
+    better clock and timer performance on Windows (see Windows section
+    below for more details).
+
+gst-play-1.0
+
+-   New --start-position command line argument to start playback from
+    the specified position
+-   Audio can be muted/unmuted in interactive mode by pressing the m
+    key.
+-   On Windows the high-resolution clock is enabled now (see Windows
+    section below for more details)
+
+gst-device-monitor-1.0
+
+-   New --include-hidden command line argument to also show “hidden”
+    device providers
+
+ges-launch-1.0
+
+-   New interactive mode that allows seeking and such. Can be disabled
+    by passing the --no-interactive argument on the command line.
+-   Option to forward tags
+-   Allow using an existing clip to determine the rendering format (both
+    topology and profile) via new --profile-from command line argument.
 
 GStreamer RTSP server
 
--   this section will be filled in in due course
+-   GstRTSPMediaFactory gained API to disable RTCP
+    (gst_rtsp_media_factory_set_enable_rtcp(), "enable-rtcp" property).
+    Previously RTCP was always allowed for all RTSP medias. With this
+    change it is possible to disable RTCP completely, no matter if the
+    client wants to do RTCP or not.
+
+-   Make a mount point of / work correctly. While not allowed by the
+    RTSP 2 spec, the RTSP 1 spec is silent on this and it is used in the
+    wild. It is now possible to use / as a mount path in
+    gst-rtsp-server, e.g. rtsp://example.com/ would work with this now.
+    Note that query/fragment parts of the URI are not necessarily
+    correctly handled, and behaviour will differ between various
+    client/server implementations; so use it if you must but don’t bug
+    us if it doesn’t work with third party clients as you’d hoped.
+
+-   multithreading fixes (races, refcounting issues, deadlocks)
+
+-   ONVIF audio backchannel fixes
+
+-   ONVIF trick mode optimisations
+
+-   rtspclientsink: new "update-sdp" signal that allows updating the SDP
+    before sending it to the server via ANNOUNCE. This can be used to
+    add additional metadata to the SDP, for example. The order and
+    number of medias must not be changed, however.
 
 GStreamer VAAPI
 
--   this section will be filled in in due course
+-   new AV1 decoder element (vaapiav1dec)
+
+-   H264 decoder: handle stereoscopic 3D video with frame packing
+    arrangement SEI messages
+
+-   H265 encoder: added Screen Content Coding extensions support
+
+-   H265 decoder: gained MAIN_444_12 profile support (decoded to
+    Y412_LE), and 4:2:2 12-bits support (decoded to Y212_LE)
+
+-   vaapipostproc: gained BT2020 color standard support
+
+-   vaapidecode: now generates caps templates dynamically at runtime in
+    order to advertise actually supported caps instead of all
+    theoretically supported caps.
+
+-   GST_VAAPI_DRM_DEVICE environment variable to force a specified DRM
+    device when a DRM display is used. It is ignored when other types of
+    displays are used. By default /dev/dri/renderD128 is used for DRM
+    display.
 
 GStreamer OMX
 
--   this section will be filled in in due course
+-   subframe support in H.264/H.265 decoders
 
 GStreamer Editing Services and NLE
 
--   this section will be filled in in due course
+-   framepositioner: new "operator" property to access blending modes in
+    the compositor
+-   timeline: Implement snapping to markers
+-   smart-mixer: Add support for d3d11compositor and glvideomixer
+-   titleclip: add "draw-shadow" child property
+-   ges:// URI support to define a timeline from a description.
+-   command-line-formatter
+    -   Add track management to timeline description
+    -   Add keyframe support
+-   ges-launch-1.0:
+    -   Add an interactive mode where we can seek etc…
+    -   Add option to forward tags
+    -   Allow using an existing clip to determine the rendering format
+        (both topology and profile) via new --profile-from command line
+        argument.
+-   Fix static build
 
 GStreamer validate
 
--   this section will be filled in in due course
+-   report: Add a way to force backtraces on reports even if not a
+    critical issue (GST_VALIDATE_ISSUE_FLAGS_FORCE_BACKTRACE)
+-   Add a flag to gst_validate_replace_variables_in_string() allow
+    defining how to resolve variables in structs
+-   Add gst_validate_bin_monitor_get_scenario() to get the bin monitor
+    scenario, which is useful for applications that use Validate
+    directly.
+-   Add an expected-values parameter to wait, message-type=XX allowing
+    more precise filtering of the message we are waiting for.
+-   Add config file support: each test can now use a config file for the
+    given media file used to test.
+-   Add support to check properties of object properties
+-   scenario: Add an "action-done" signal to signal when an action is
+    done
+-   scenario: Add a "run-command" action type
+-   scenario: Allow forcing running action on idle from scenario file
+-   scenario: Allow iterating over arrays in foreach
+-   scenario: Rename ‘interlaced’ action to ‘non-blocking’
+-   scenario: Add a non-blocking flag to the wait signal
 
 GStreamer Python Bindings
 
--   this section will be filled in in due course
+-   Fixes for Python 3.10
+-   Various build fixes
+-   at least one known breaking change caused by g-i annotation changes
+    (see below)
 
 GStreamer C# Bindings
 
--   this section will be filled in in due course
+-   Fix GstDebugGraphDetails enum
+-   Updated to latests GtkSharp
+-   Updated to include GStreamer 1.20 API
 
 GStreamer Rust Bindings and Rust Plugins
 
-The GStreamer Rust bindings are released separately with a different
-release cadence that’s tied to gtk-rs, but the latest release has
-already been updated for the upcoming new GStreamer 1.20 API.
-
-gst-plugins-rs, the module containing GStreamer plugins written in Rust,
-has also seen lots of activity with many new elements and plugins.
-
-What follows is a list of elements and plugins available in
-gst-plugins-rs, so people don’t miss out on all those potentially useful
-elements that have no C equivalent.
-
--   FIXME: add new elements
-
-Rust audio plugins
-
--   audiornnoise: New element for audio denoising which implements the
-    noise removal algorithm of the Xiph RNNoise library, in Rust
--   rsaudioecho: Port of the audioecho element from gst-plugins-good
-    rsaudioloudnorm: Live audio loudness normalization element based on
-    the FFmpeg af_loudnorm filter
--   claxondec: FLAC lossless audio codec decoder element based on the
-    pure-Rust claxon implementation
--   csoundfilter: Audio filter that can use any filter defined via the
-    Csound audio programming language
--   lewtondec: Vorbis audio decoder element based on the pure-Rust
-    lewton implementation
-
-Rust video plugins
-
--   cdgdec/cdgparse: Decoder and parser for the CD+G video codec based
-    on a pure-Rust CD+G implementation, used for example by karaoke CDs
--   cea608overlay: CEA-608 Closed Captions overlay element
--   cea608tott: CEA-608 Closed Captions to timed-text (e.g. VTT or SRT
-    subtitles) converter
--   tttocea608: CEA-608 Closed Captions from timed-text converter
--   mccenc/mccparse: MacCaption Closed Caption format encoder and parser
--   sccenc/sccparse: Scenarist Closed Caption format encoder and parser
--   dav1dec: AV1 video decoder based on the dav1d decoder implementation
-    by the VLC project
--   rav1enc: AV1 video encoder based on the fast and pure-Rust rav1e
-    encoder implementation
--   rsflvdemux: Alternative to the flvdemux FLV demuxer element from
-    gst-plugins-good, not feature-equivalent yet
--   rsgifenc/rspngenc: GIF/PNG encoder elements based on the pure-Rust
-    implementations by the image-rs project
-
-Rust text plugins
-
--   textwrap: Element for line-wrapping timed text (e.g. subtitles) for
-    better screen-fitting, including hyphenation support for some
-    languages
-
-Rust network plugins
-
--   reqwesthttpsrc: HTTP(S) source element based on the Rust
-    reqwest/hyper HTTP implementations and almost feature-equivalent
-    with the main GStreamer HTTP source souphttpsrc
--   s3src/s3sink: Source/sink element for the Amazon S3 cloud storage
--   awstranscriber: Live audio to timed text transcription element using
-    the Amazon AWS Transcribe API
-
-Generic Rust plugins
-
--   sodiumencrypter/sodiumdecrypter: Encryption/decryption element based
-    on libsodium/NaCl
--   togglerecord: Recording element that allows to pause/resume
-    recordings easily and considers keyframe boundaries
--   fallbackswitch/fallbacksrc: Elements for handling potentially
-    failing (network) sources, restarting them on errors/timeout and
-    showing a fallback stream instead
--   threadshare: Set of elements that provide alternatives for various
-    existing GStreamer elements but allow to share the streaming threads
-    between each other to reduce the number of threads
--   rsfilesrc/rsfilesink: File source/sink elements as replacements for
-    the existing filesrc/filesink elements
+-   The GStreamer Rust bindings are released separately with a different
+    release cadence that’s tied to gtk-rs, but the latest release has
+    already been updated for the upcoming new GStreamer 1.20 API (v1_20
+    feature).
+
+-   gst-plugins-rs, the module containing GStreamer plugins written in
+    Rust, has also seen lots of activity with many new elements and
+    plugins. See the New Elements section above for a list of new Rust
+    elements.
 
 Build and Dependencies
 
 -   Meson 0.59 or newer is required to build GStreamer now.
 
--   FIXME: this section will be filled in in due course
+-   The GLib requirement has been bumped to GLib 2.56 or newer (from
+    March 2018).
+
+-   The wpe plugin now requires wpe >= 2.28 and wpebackend-fdo >= 1.8
 
 Explicit opt-in required for build of certain plugins with (A)GPL dependencies
 
@@ -211,11 +1591,9 @@ even if the required dependencies are available.
 See Building plugins with (A)GPL-licensed dependencies for more details
 and a non-exhaustive list of plugins affected.
 
-gst-build: replaced by Monorepo
+gst-build: replaced by mono repository
 
--   this section will be filled in in due course
-
--   FIXME: describe + link to Monorepo FAQ
+See mono repository section above and the GStreamer mono repository FAQ.
 
 Cerbero
 
@@ -223,76 +1601,254 @@ Cerbero is a meta build system used to build GStreamer plus dependencies
 on platforms where dependencies are not readily available, such as
 Windows, Android, iOS and macOS.
 
-General improvements
+General Cerbero improvements
 
--   this section will be filled in in due course
+-   Plugin removed: libvisual
+-   New plugins: rtpmanagerbad and rist
 
-macOS / iOS
+macOS / iOS specific Cerbero improvements
 
--   this section will be filled in in due course
+-   XCode 12 support
+-   macOS OS release support is now future-proof, similar to iOS
+-   macOS Apple Silicon (ARM64) cross-compile support has been added
+-   macOS Apple Silicon (ARM64) native support is currently experimental
 
-Windows
+Windows specific Cerbero improvements
 
--   this section will be filled in in due course
+-   Visual Studio 2022 support has been added
+-   bootstrap is faster since it requires building fewer build-tools
+    recipes on Windows
+-   package is faster due to better scheduling of recipe stages and
+    elimination of unnecessary autotools regeneration
+-   The following plugins are no longer built on Windows:
+    -   a52dec (another decoder is still available in libav)
+    -   dvdread
+    -   resindvd
 
 Windows MSI installer
 
--   this section will be filled in in due course
+-   no major changes
 
-Linux
+Linux specific Cerbero improvements
 
--   this section will be filled in in due course
+-   Fedora, Debian OS release support is now more future-proof
+-   Amazon Linux 2 support has been added
 
-Android
+Android specific Cerbero improvements
 
--   this section will be filled in in due course
+-   no major changes
 
 Platform-specific changes and improvements
 
 Android
 
--   this section will be filled in in due course
+-   No major changes
 
 macOS and iOS
 
--   this section will be filled in in due course
+-   applemedia: add ProRes support to vtenc and vtdec
 
 Windows
 
--   this section will be filled in in due course
+-   On Windows the high-resolution clock is enabled now in the
+    gst-launch-1.0 and gst-play-1.0 command line tools, which provides
+    better clock and timer performance on Windows, at the cost of higher
+    power consumption. By default, without the high-resolution clock
+    enabled, the timer precision on Windows is system-dependent and may
+    be as bad as 15ms which is not good enough for many multimedia
+    applications. Developers may want to do the same in their Windows
+    applications if they think it’s a good idea for their application
+    use case, and depending on the Windows version they target. This is
+    not done automatically by GStreamer because on older Windows
+    versions (pre-Windows 10) this affects a global Windows setting and
+    also there’s a power consumption vs. performance trade-off that may
+    differ from application to application.
+
+-   dxgiscreencapsrc now supports resolution changes
+
+-   The wasapi2 audio plugin was rewritten and now has a higher rank
+    than the old wasapi plugin since it has a number of additional
+    features such as automatic stream routing, and no
+    known-but-hard-to-fix issues. The plugin is always built if the
+    Windows 10 SDK is available now.
+
+-   The wasapi device providers now detect and notify dynamic device
+    additions/removals
+
+-   d3d11screencapturesrc: new desktop capture element, including
+    GstDeviceProvider implementation to enumerate/select target monitors
+    for capture.
+
+-   Direct3D11/DXVA decoder now supports AV1 and MPEG2 codecs
+    (d3d11av1dec, d3d11mpeg2dec)
+
+-   VP9 decoding got more reliable and stable thanks to a newly written
+    codec parser
+
+-   Support for decoding interlaced H.264/AVC streams
+
+-   Hardware-accelerated video deinterlacing (d3d11deinterlace) and
+    video mixing (d3d11compositor)
+
+-   Video mixing with the Direct3D11 API (d3d11compositor)
+
+-   MediaFoundation API based hardware encoders gained the ability to
+    receive Direct3D11 textures as an input
+
+-   Seungha’s blog post “GStreamer ❤ Windows: A primer on the cool stuff
+    you’ll find in the 1.20 release” describes many of the
+    Windows-related improvements in more detail
 
 Linux
 
--   this section will be filled in in due course
+-   bluez: LDAC Bluetooth audio codec support in a2dpsink and avdtpsink,
+    as well as an LDAC RTP payloader (rtpldacpay) and an LDAC audio
+    encoder (ldacenc)
+
+-   kmssink: gained support for NV24, NV61, RGB16/BGR16 formats;
+    auto-detect NVIDIA Tegra driver
 
 Documentation improvements
 
--   this section will be filled in in due course
+-   hardware-accelerated GPU plugins will now no longer always list all
+    the element variants for all available GPUs, since those are
+    system-dependent and it’s confusing for users to see those in the
+    documentation just because the GStreamer developer who generated the
+    docs had multiple GPUs to play with at the time. Instead just show
+    the default elements.
 
-Possibly Breaking Changes
+Possibly Breaking and Other Noteworthy Behavioural Changes
+
+-   gst_parse_launch(), gst_parse_bin_from_description() and friends
+    will now error out when setting properties that don’t exist on
+    top-level bins. They were silently ignored before.
+
+-   The GstWebRTC library does not expose any objects anymore with
+    public fields. Instead properties have been added to replace that
+    functionality. If you are accessing such fields in your application,
+    switch to the corresponding properties.
 
--   FIXME: this section will be filled in in due course
--   MPEG-TS SCTE-35 API changes (FIXME: flesh out)
--   gst_parse_launch() and friends now error out on non-existing
-    properties on top-level bins where they would silently fail and
-    ignore those before.
 -   playbin and uridecodebin now emit the source-setup signal before the
     element is added to the bin and linked so that the source element is
     already configured before any scheduling query comes in, which is
-    useful for elements such as appsrc or giostreamsrc. (Merge Request)
+    useful for elements such as appsrc or giostreamsrc.
+
+-   The source element inside urisourcebin (used inside uridecodebin3
+    which is used inside playbin3) is no longer called "source". This
+    shouldn’t affect anyone hopefully, because there’s a "setup-source"
+    signal to configure the source element and no one should rely on
+    names of internal elements anyway.
+
+-   The vp8enc element now expects bps (bits per second) for the
+    "temporal-scalability-target-bitrate" property, which is consistent
+    with the "target-bitrate" property. Since additional configuration
+    is required with modern libvpx to make temporal scaling work anyway,
+    chances are that very few people will have been using this property
+
+-   vp8enc and vp9enc now default to “good quality” for the "deadline"
+    property rather then “best quality”. Having the deadline set to best
+    quality causes the encoder to be absurdly slow, most real-life users
+    will want the good quality tradeoff instead.
+
+-   The experimental GstTranscoder library API in gst-plugins-bad was
+    changed from a GObject signal-based notification mechanism to a
+    GstBus/message-based mechanism akin to GstPlayer/GstPlay.
+
+-   MPEG-TS SCTE-35 API: semantic change for SCTE-35 splice commands:
+    timestamps passed by the application should be in running time now,
+    since users of the API can’t really be expected to predict the local
+    PTS of the muxer.
+
+-   The GstContext used by souphttpsrc to share the session between
+    multiple element instances has changed. Previously it provided
+    direct access to the internal SoupSession object, now it only
+    provides access to an opaque, internal type. This change is
+    necessary because SoupSession is not thread-safe at all and can’t be
+    shared safely between arbitrary external code and souphttpsrc.
+
+-   Python bindings: GObject-introspection related Annotation fixes have
+    led to a case of a GstVideo.VideoInfo-related function signature
+    changing in the Python bindings (possibly one or two other cases
+    too). This is for a function that should never have been exposed in
+    the first place though, so the bindings are being updated to throw
+    an exception in that case, and the correct replacement API has been
+    added in form of an override.
 
 Known Issues
 
--   this section will be filled in in due course
-
--   There are a couple of known WebRTC-related regressions/blockers:
-
-    -   webrtc: DTLS setup with Chrome is broken
-    -   webrtcbin: First keyframe is usually lost
+-   nothing in particular at this point (but also see possibly breaking
+    changes section above)
 
 Contributors
 
--   this section will be filled in in due course
+Aaron Boxer, Adam Leppky, Adam Williamson, Alba Mendez, Alejandro
+González, Aleksandr Slobodeniuk, Alexander Vandenbulcke, Alex Ashley,
+Alicia Boya García, Andika Triwidada, Andoni Morales Alastruey, Andrew
+Wesie, Andrey Moiseev, Antonio Ospite, Antonio Rojas, Arthur Crippa
+Búrigo, Arun Raghavan, Ashley Brighthope, Axel Kellermann, Baek, Bastien
+Nocera, Bastien Reboulet, Benjamin Gaignard, Bing Song, Binh Truong,
+Biswapriyo Nath, Brad Hards, Brad Smith, Brady J. Garvin, Branko
+Subasic, Camilo Celis Guzman, Chris Bass, ChrisDuncanAnyvision, Chris
+White, Corentin Damman, Daniel Almeida, Daniel Knobe, Daniel Stone,
+david, David Fernandez, David Keijser, David Phung, Devarsh Thakkar,
+Dinesh Manajipet, Dmitry Samoylov, Dmitry Shusharin, Dominique Martinet,
+Doug Nazar, Ederson de Souza, Edward Hervey, Emmanuel Gil Peyrot,
+Enrique Ocaña González, Ezequiel Garcia, Fabian Orccon, Fabrice
+Fontaine, Fernando Jimenez Moreno, Florian Karydes, Francisco Javier
+Velázquez-García, François Laignel, Frederich Munch, Fredrik Pålsson,
+George Kiagiadakis, Georg Lippitsch, Göran Jönsson, Guido Günther,
+Guillaume Desmottes, Guiqin Zou, Haakon Sporsheim, Haelwenn (lanodan)
+Monnier, Haihao Xiang, Haihua Hu, Havard Graff, He Junyan, Helmut
+Januschka, Henry Wilkes, Hosang Lee, Hou Qi, Ignacio Casal Quinteiro,
+Igor Kovalenko, Ilya Kreymer, Imanol Fernandez, Jacek Tomaszewski, Jade
+Macho, Jakub Adam, Jakub Janků, Jan Alexander Steffens (heftig), Jan
+Schmidt, Jason Carrete, Jason Pereira, Jay Douglass, Jeongki Kim, Jérôme
+Laheurte, Jimmi Holst Christensen, Johan Sternerup, John Hassell, John
+Lindgren, John-Mark Bell, Jonathan Matthew, Jordan Petridis, Jose
+Quaresma, Julian Bouzas, Julien, Kai Uwe Broulik, Kasper Steensig
+Jensen, Kellermann Axel, Kevin Song, Khem Raj, Knut Inge Hvidsten, Knut
+Saastad, Kristofer Björkström, Lars Lundqvist, Lawrence Troup, Lim Siew
+Hoon, Lucas Stach, Ludvig Rappe, Luis Paulo Fernandes de Barros, Luke
+Yelavich, Mads Buvik Sandvei, Marc Leeman, Marco Felsch, Marek Vasut,
+Marian Cichy, Marijn Suijten, Marius Vlad, Markus Ebner, Mart Raudsepp,
+Matej Knopp, Mathieu Duponchelle, Matthew Waters, Matthieu De Beule,
+Mengkejiergeli Ba, Michael de Gans, Michael Olbrich, Michael Tretter,
+Michal Dzik, Miguel Paris, Mikhail Fludkov, mkba, Nazar Mokrynskyi,
+Nicholas Jackson, Nicola Murino, Nicolas Dufresne, Niklas Hambüchen,
+Nikolay Sivov, Nirbheek Chauhan, Olivier Blin, Olivier Crete, Olivier
+Crête, Paul Goulpié, Per Förlin, Peter Boba, P H, Philippe Normand,
+Philipp Zabel, Pieter Willem Jordaan, Piotrek Brzeziński, Rafał
+Dzięgiel, Rafostar, raghavendra, Raghavendra, Raju Babannavar, Raleigh
+Littles III, Randy Li, Randy Li (ayaka), Ratchanan Srirattanamet, Raul
+Tambre, reed.lawrence, Ricky Tang, Robert Rosengren, Robert Swain, Robin
+Burchell, Roman Sivriver, R S Nikhil Krishna, Ruben Gonzalez, Ruslan
+Khamidullin, Sanchayan Maity, Scott Moreau, Sebastian Dröge, Sergei
+Kovalev, Seungha Yang, Sid Sethupathi, sohwan.park, Sonny Piers, Staz M,
+Stefan Brüns, Stéphane Cerveau, Stephan Hesse, Stian Selnes, Stirling
+Westrup, Théo MAILLART, Thibault Saunier, Tim, Timo Wischer, Tim-Philipp
+Müller, Tim Schneider, Tobias Ronge, Tom Schoonjans, Tulio Beloqui,
+tyler-aicradle, U. Artie Eoff, Ung, Val Doroshchuk, VaL Doroshchuk,
+Víctor Manuel Jáquez Leal, Vivek R, Vivia Nikolaidou, Vivienne
+Watermeier, Vladimir Menshakov, Will Miller, Wim Taymans, Xabier
+Rodriguez Calvar, Xavier Claessens, Xℹ Ruoyao, Yacine Bandou, Yinhang
+Liu, youngh.lee, youngsoo.lee, yychao, Zebediah Figura, Zhang yuankun,
+Zhang Yuankun, Zhao, Zhao Zhili, , Aleksandar Topic, Antonio Ospite,
+Bastien Nocera, Benjamin Gaignard, Brad Hards, Carlos Falgueras García,
+Célestin Marot, Corentin Damman, Corentin Noël, Daniel Almeida, Daniel
+Knobe, Danny Smith, Dave Piché, Dmitry Osipenko, Fabrice Fontaine,
+fjmax, Florian Zwoch, Guillaume Desmottes, Haihua Hu, Heinrich Kruger,
+He Junyan, Jakub Adam, James Cowgill, Jan Alexander Steffens (heftig),
+Jean Felder, Jeongki Kim, Jiri Uncovsky, Joe Todd, Jordan Petridis,
+Krystian Wojtas, Marc-André Lureau, Marcin Kolny, Marc Leeman, Mark
+Nauwelaerts, Martin Reboredo, Mathieu Duponchelle, Matthew Waters,
+Mengkejiergeli Ba, Michael Gruner, Nicolas Dufresne, Nirbheek Chauhan,
+Olivier Crête, Philippe Normand, Rafał Dzięgiel, Ralf Sippl, Robert
+Mader, Sanchayan Maity, Sangchul Lee, Sebastian Dröge, Seungha Yang,
+Stéphane Cerveau, Teh Yule Kim, Thibault Saunier, Thomas Klausner, Timo
+Wischer, Tim-Philipp Müller, Tobias Reineke, Tomasz Andrzejak, Trung Do,
+Tyler Compton, Ung, Víctor Manuel Jáquez Leal, Vivia Nikolaidou, Wim
+Taymans, wngecn, Wonchul Lee, wuchang li, Xavier Claessens, Xi Ruoyao,
+Yoshiharu Hirose, Zhao,
 
 … and many others who have contributed bug reports, translations, sent
 suggestions or helped testing.
@@ -307,7 +1863,7 @@ the git 1.20 branch, which will be a stable branch.
 
 1.20.0
 
-1.20.0 is scheduled to be released around October/November 2021.
+1.20.0 is scheduled to be released around early February 2022.
 
 Schedule for 1.22
 
@@ -315,7 +1871,9 @@ Our next major feature release will be 1.22, and 1.21 will be the
 unstable development version leading up to the stable 1.22 release. The
 development of 1.21/1.22 will happen in the git main branch.
 
-The plan for the 1.22 development cycle is yet to be confirmed.
+The plan for the 1.22 development cycle is yet to be confirmed. Assuming
+no major project-wide reorganisations in the 1.22 cycle we might try and
+aim for a release around August 2022.
 
 1.22 will be backwards-compatible to the stable 1.20, 1.18, 1.16, 1.14,
 1.12, 1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series.
@@ -323,6 +1881,7 @@ The plan for the 1.22 development cycle is yet to be confirmed.
 ------------------------------------------------------------------------
 
 These release notes have been prepared by Tim-Philipp Müller with
-contributions from …
+contributions from Matthew Waters, Nicolas Dufresne, Nirbheek Chauhan,
+Sebastian Dröge and Seungha Yang.
 
 License: CC BY-SA 4.0
diff --git a/subprojects/gst-python/RELEASE b/subprojects/gst-python/RELEASE
index c2ceb77fe2..1d90472811 100644
--- a/subprojects/gst-python/RELEASE
+++ b/subprojects/gst-python/RELEASE
@@ -1,4 +1,4 @@
-This is GStreamer gst-python 1.19.3.
+This is GStreamer gst-python 1.19.90.
 
 GStreamer 1.19 is the development branch leading up to the next major
 stable version which will be 1.20.
diff --git a/subprojects/gst-python/gst-python.doap b/subprojects/gst-python/gst-python.doap
index 14bf441e4c..39cdcbb3d8 100644
--- a/subprojects/gst-python/gst-python.doap
+++ b/subprojects/gst-python/gst-python.doap
@@ -32,6 +32,16 @@ GStreamer Python Bindings is a set of overrides and Gst fundamental types handli
 
  <release>
   <Version>
+   <revision>1.19.90</revision>
+   <branch>main</branch>
+   <name></name>
+   <created>2022-01-28</created>
+   <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-python/gst-python-1.19.90.tar.xz" />
+  </Version>
+ </release>
+
+ <release>
+  <Version>
    <revision>1.19.3</revision>
    <branch>main</branch>
    <name></name>
diff --git a/subprojects/gst-python/meson.build b/subprojects/gst-python/meson.build
index 7d1f45356e..c197664dec 100644
--- a/subprojects/gst-python/meson.build
+++ b/subprojects/gst-python/meson.build
@@ -1,5 +1,5 @@
 project('gst-python', 'c', 'cpp',
-  version : '1.19.3.1',
+  version : '1.19.90',
   meson_version : '>= 0.59',
   default_options : [ 'warning_level=1',
                       'c_std=gnu99',
diff --git a/subprojects/gst-rtsp-server/ChangeLog b/subprojects/gst-rtsp-server/ChangeLog
index cb38ffce71..0cba8535db 100644
--- a/subprojects/gst-rtsp-server/ChangeLog
+++ b/subprojects/gst-rtsp-server/ChangeLog
@@ -1,3 +1,19 @@
+=== release 1.19.90 ===
+
+2022-01-28 14:28:35 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* NEWS:
+	* RELEASE:
+	* docs/gst_plugins_cache.json:
+	* gst-rtsp-server.doap:
+	* meson.build:
+	  Release 1.19.90
+
+2022-01-28 14:28:28 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* ChangeLog:
+	  Update ChangeLogs for 1.19.90
+
 2022-01-20 17:13:36 -0600  Michael Gruner <michael.gruner@ridgerun.com>
 
 	* examples/test-appsrc2.c:
diff --git a/subprojects/gst-rtsp-server/NEWS b/subprojects/gst-rtsp-server/NEWS
index e9f5227c15..1a512c1848 100644
--- a/subprojects/gst-rtsp-server/NEWS
+++ b/subprojects/gst-rtsp-server/NEWS
@@ -1,23 +1,19 @@
 GStreamer 1.20 Release Notes
 
 GStreamer 1.20 has not been released yet. It is scheduled for release in
-November 2021.
+late January / early February 2022.
 
 1.19.x is the unstable development version that is being developed in
-the git main branch and which will eventually result in 1.20, and 1.19.3
-is the current development release in that series
-
-It is expected that feature freeze will be in early October 2021,
-followed by one or two 1.19.9x pre-releases and the new 1.20 stable
-release around October/November 2021.
+the git main branch and which will eventually result in 1.20, and
+1.19.90 is the first release candidate in that series (1.20rc1).
 
 1.20 will be backwards-compatible to the stable 1.18, 1.16, 1.14, 1.12,
-1.10, 1.8, 1.6,, 1.4, 1.2 and 1.0 release series.
+1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series.
 
 See https://gstreamer.freedesktop.org/releases/1.20/ for the latest
 version of this document.
 
-Last updated: Monday 1 November 2021, 01:00 UTC (log)
+Last updated: Wednesday 26 January 2022, 01:00 UTC (log)
 
 Introduction
 
@@ -30,25 +26,971 @@ fixes and other improvements.
 
 Highlights
 
--   this section will be completed in due course
+-   Development in GitLab was switched to a single git repository
+    containing all the modules
+-   GstPlay: new high-level playback library, replaces GstPlayer
+-   WebM Alpha decoding support
+-   Encoding profiles can now be tweaked with additional
+    application-specified element properties
+-   Compositor: multi-threaded video conversion and mixing
+-   RTP header extensions: unified support in RTP depayloader and
+    payloader base classes
+-   SMPTE 2022-1 2-D Forward Error Correction support
+-   Smart encoding (passthrough) support for VP8, VP9, H.265 in
+    encodebin and transcodebin
+-   Runtime compatibility support for libsoup2 and libsoup3 (libsoup3
+    support experimental)
+-   Video decoder subframe support
+-   Video decoder automatic packet-loss, data corruption, and keyframe
+    request handling for RTP / WebRTC / RTSP
+-   MP4 and Matroska muxers now support profile/level/resolution changes
+    for H264/H265 input streams (i.e. codec data changing on the fly)
+-   MP4 muxing mode that initially creates a fragmented mp4 which is
+    converted to a regular mp4 on EOS
+-   Audio support for the WebKit Port for Embedded (WPE) web page source
+    element
+-   CUDA based video color space convert and rescale elements and
+    upload/download elements
+-   NVIDIA memory:NVMM support for OpenGL glupload and gldownload
+    elements
+-   Many WebRTC improvements
+-   The new VA-API plugin implemention fleshed out with more decoders
+    and new postproc elements
+-   AppSink API to retrieve events in addition to buffers and buffer
+    lists
+-   AppSrc gained more configuration options for the internal queue
+    (leakiness, limits in buffers and time, getters to read current
+    levels)
+-   Updated Rust bindings and many new Rust plugins
+-   Improved support for custom minimal GStreamer builds
+-   Support build against FFmpeg 5.0
+-   Linux Stateless CODEC support gained MPEG2 and VP9
+-   Windows Direct3D11/DXVA decoder gained AV1 and MPEG2 support
+-   Lots of new plugins, features, performance improvements and bug
+    fixes
 
 Major new features and changes
 
 Noteworthy new features and API
 
--   this section will be filled in in due course
+-   gst_element_get_request_pad() has been deprecated in favour of the
+    newly-added gst_element_request_pad_simple() which does the exact
+    same thing but has a less confusing name that hopefully makes clear
+    that the function request a new pad rather than just retrieves an
+    already-existing request pad.
+
+Development in GitLab was switched to a single git repository containing all the modules
+
+The GStreamer multimedia framework is a set of libraries and plugins
+split into a number of distinct modules which are released independently
+and which have so far been developed in separate git repositories in
+freedesktop.org GitLab.
+
+In addition to these separate git repositories there was a gst-build
+module that would use the Meson build systems’s subproject feature to
+download each individual module and then build everything in one go. It
+would also provide an uninstalled development environment that made it
+easy to work on GStreamer and use or test versions other than the
+system-installed GStreamer version.
+
+All of these modules have now (as of 28 September 2021) been merged into
+a single git repository (“Mono repository” or “monorepo”) which should
+simplify development workflows and continuous integration, especially
+where changes need to be made to multiple modules at once.
+
+This mono repository merge will primarily affect GStreamer developers
+and contributors and anyone who has workflows based on the GStreamer git
+repositories.
+
+The Rust bindings and Rust plugins modules have not been merged into the
+mono repository at this time because they follow a different release
+cycle.
+
+The mono repository lives in the existing GStreamer core git repository
+in GitLab in the new main branch and all future development will happen
+on this branch.
+
+Modules will continue to be released as separate tarballs.
+
+For more details, please see the GStreamer mono repository FAQ.
+
+GstPlay: new high-level playback library replacing GstPlayer
+
+-   GstPlay is a new high-level playback library that replaces the older
+    GstPlayer API. It is basically the same API as GstPlayer but
+    refactored to use bus messages for application notifications instead
+    of GObject signals. There is still a signal adapter object for those
+    who prefer signals. Since the existing GstPlayer API is already in
+    use in various applications, it didn’t seem like a good idea to
+    break it entirely. Instead a new API was added, and it is expected
+    that this new GstPlay API will be moved to gst-plugins-base in
+    future.
+
+-   The existing GstPlayer API is scheduled for deprecation and will be
+    removed at some point in the future (e.g. in GStreamer 1.24), so
+    application developers are urged to migrate to the new GstPlay API
+    at their earliest convenience.
+
+WebM alpha decoding
+
+-   Implement WebM alpha decoding (VP8/VP9 with alpha), which required
+    support and additions in various places. This is supported both with
+    software decoders and hardware-accelerated decoders.
+
+-   VP8/VP9 don’t support alpha components natively in the codec, so the
+    way this is implemented in WebM is by encoding the alpha plane with
+    transparency data as a separate VP8/VP9 stream. Inside the WebM
+    container (a variant of Matroska) this is coded as a single video
+    track with the “normal” VP8/VP9 video data making up the main video
+    data and each frame of video having an encoded alpha frame attached
+    to it as extra data ("BlockAdditional").
+
+-   matroskademux has been extended extract this per-frame alpha side
+    data and attach it in form of a GstVideoCodecAlphaMeta to the
+    regular video buffers. Note that this new meta is specific to this
+    VP8/VP9 alpha support and can’t be used to just add alpha support to
+    other codecs that don’t support it. Lastly, matroskademux also
+    advertises the fact that the streams contain alpha in the caps.
+
+-   The new codecalpha plugin contains various bits of infrastructure to
+    support autoplugging and debugging:
+
+    -   codecalphademux splits out the alpha stream from the metas on
+        the regular VP8/VP9 buffers
+    -   alphacombine takes two decoded raw video streams (one alpha, one
+        the regular video) and combines it into a video stream with
+        alpha
+    -   vp8alphadecodebin + vp9alphadecodebin are wrapper bins that use
+        the regular vp8dec and vp9dec software decoders to decode
+        regular and alpha streams and combine them again. To decodebin
+        these look like regular decoders which ju
+    -   The V4L2 CODEC plugin has stateless VP8/VP9 decoders that can
+        decode both alpha and non-alpha stream with a single decoder
+        instance
+
+-   A new AV12 video format was added which is basically NV12 with an
+    alpha plane, which is more convenient for many hardware-accelerated
+    decoders.
+
+-   Watch Nicolas Dufresne’s LCA 2022 talk “Bringing WebM Alpha support
+    to GStreamer” for all the details and a demo.
+
+RTP Header Extensions Base Class and Automatic Header Extension Handling in RTP Payloaders and Depayloaders
+
+-   RTP Header Extensions are specified in RFC 5285 and provide a way to
+    add small pieces of data to RTP packets in between the RTP header
+    and the RTP payload. This is often used for per-frame metadata,
+    extended timestamps or other application-specific extra data. There
+    are several commonly-used extensions specified in various RFCs, but
+    senders are free to put any kind of data in there, as long as sender
+    and receiver both know what that data is. Receivers that don’t know
+    about the header extensions will just skip the extra data without
+    ever looking at it. These header extensions can often be combined
+    with any kind of payload format, so may need to be supported by many
+    RTP payloader and depayloader elements.
+
+-   Inserting and extracting RTP header extension data has so far been a
+    bit inconvenient in GStreamer: There are functions to add and
+    retrieve RTP header extension data from RTP packets, but nothing
+    works automatically, even for common extensions. People would have
+    to do the insertion/extraction either in custom elements
+    before/after the RTP payloader/depayloader, or inside pad probes,
+    which isn’t very nice.
+
+-   This release adds various pieces of new infrastructure for generic
+    RTP header extension handling, as well as some implementations for
+    common extensions:
+
+    -   GstRTPHeaderExtension is a new helper base class for reading and
+        writing RTP header extensions. Nominally this subclasses
+        GstElement, but only so these extensions are stored in the
+        registry where they can be looked up by URI or name. They don’t
+        have pads and don’t get added to the pipeline graph as an
+        element.
+
+    -   "add-extension" and "clear-extension" action signals on RTP
+        payloaders and depayloaders for manual extension management
+
+    -   The "request-extension" signal will be emitted if an extension
+        is encountered that requires explicit mapping by the application
+
+    -   new "auto-header-extension" property on RTP payloaders and
+        depayloaders for automatic handling of known header extensions.
+        This is enabled by default. The extensions must be signalled via
+        caps / SDP.
+
+    -   RTP header extension implementations:
+
+        -   rtphdrextclientaudiolevel: Client-to-Mixer Audio Level
+            Indication (RFC 6464) (also see below)
+        -   rtphdrextcolorspace: Color Space extension, extends RTP
+            packets with color space and high dynamic range (HDR)
+            information
+        -   rtphdrexttwcc: Transport Wide Congestion Control support
+
+-   gst_rtp_buffer_remove_extension_data() is a new helper function to
+    remove an RTP header extension from an RTP buffer
+
+-   The existing gst_rtp_buffer_set_extension_data() now also supports
+    shrinking the extension data in size
+
+AppSink and AppSrc improvements
+
+-   appsink: new API to pull events out of appsink in addition to
+    buffers and buffer lists.
+
+    There was previously no way for users to receive incoming events
+    from appsink properly serialised with the data flow, even if they
+    are serialised events. The reason for that is that the only way to
+    intercept events was via a pad probe on the appsink sink pad, but
+    there is also internal queuing inside of appsink, so it’s difficult
+    to ascertain the right order of everything in all cases.
+
+    There is now a new "new-serialized-event" signal which will be
+    emitted when there’s a new event pending (just like the existing
+    "new-sample" signal). The "emit-signals" property must be set to
+    TRUE in order to activate this (but it’s also fine to just pull from
+    the application thread without using the signals).
+
+    gst_app_sink_pull_object() and gst_app_sink_try_pull_object() can be
+    used to pull out either an event or a new sample carrying a buffer
+    or buffer list, whatever is next in the queue.
+
+    EOS events will be filtered and will not be returned. EOS handling
+    can be done the ususal way, same as with _pull_sample().
+
+-   appsrc: allow configuration of internal queue limits in time and
+    buffers and add leaky mode.
+
+    There is internal queuing inside appsrc so the application thread
+    can push data into the element which will then be picked up by the
+    source element’s streaming thread and pushed into the pipeline from
+    that streaming thread. This queue is unlimited by default and until
+    now it was only possible to set a maximum size limit in bytes. When
+    that byte limit is reached, the pushing thread (application thread)
+    would be blocked until more space becomes available.
+
+    A limit in bytes is not particularly useful for many use cases, so
+    now it is possible to also configure limits in time and buffers
+    using the new "max-time" and "max-buffers" properties. Of course
+    there are also matching new read-only"current-level-buffers" and
+    "current-level-time properties" properties to query the current fill
+    level of the internal queue in time and buffers.
+
+    And as if that wasn’t enough the internal queue can also be
+    configured as leaky using the new "leaky-type" property. That way
+    when the queue is full the application thread won’t be blocked when
+    it tries to push in more data, but instead either the new buffer
+    will be dropped or the oldest data in the queue will be dropped.
+
+Better string serialization of nested GstCaps and GstStructures
+
+-   New string serialisation format for structs and caps that can handle
+    nested structs and caps properly by using brackets to delimit nested
+    items (e.g. some-struct, some-field=[nested-struct, nested=true]).
+    Unlike the default format the new variant can also support more than
+    one level of nesting. For backwards-compatibility reasons the old
+    format is still output by default when serialising caps and structs
+    using the existing API. The new functions gst_caps_serialize() and
+    gst_structure_serialize() can be used to output strings in the new
+    format.
+
+Convenience API for custom GstMetas
+
+-   New convenience API to register and create custom GstMetas:
+    gst_meta_register_custom() and gst_buffer_add_custom_meta(). Such
+    custom meta is backed by a GstStructure and does not require that
+    users of the API expose their GstMeta implementation as public API
+    for other components to make use of it. In addition, it provides a
+    simpler interface by ignoring the impl vs. api distinction that the
+    regular API exposes. This new API is meant to be the meta
+    counterpart to custom events and messages, and to be more convenient
+    than the lower-level API when the absolute best performance isn’t a
+    requirement. The reason it’s less performant than a “proper” meta is
+    that a proper meta is just a C struct in the end whereas this goes
+    through the GstStructure API which has a bit more overhead, which
+    for most scenarios is negligible however. This new API is useful for
+    experimentation or proprietary metas, but also has some limitations:
+    it can only be used if there’s a single producer of these metas;
+    it’s not allowed to register the same custom meta multiple times or
+    from multiple places.
+
+Additional Element Properties on Encoding Profiles
+
+-   GstEncodingProfile: The new "element-properties" and
+    gst_encoding_profile_set_element_properties() API allows
+    applications to set additional element properties on encoding
+    profiles to configure muxers and encoders. So far the encoding
+    profile template was the only place where this could be specified,
+    but often what applications want to do is take a ready-made encoding
+    profile shipped by GStreamer or the application and then tweak the
+    settings on top of that, which is now possible with this API. Since
+    applications can’t always know in advance what encoder element will
+    be used in the end, it’s even possible to specify properties on a
+    per-element basis.
+
+    Encoding Profiles are used in the encodebin, transcodebin and
+    camerabin elements and APIs to configure output formats (containers
+    and elementary streams).
+
+Audio Level Indication Meta for RFC 6464
+
+-   New GstAudioLevelMeta containing Audio Level Indication as per RFC
+    6464
+
+-   The level element has been updated to add GstAudioLevelMeta on
+    buffers if the "audio-level-meta" property is set to TRUE. This can
+    then in turn be picked up by RTP payloaders to signal the audio
+    level to receivers through RTP header extensions (see above).
+
+-   New Client-to-Mixer Audio Level Indication (RFC6464) RTP Header
+    Extension which should be automatically created and used by RTP
+    payloaders and depayloaders if their "auto-header-extension"
+    property is enabled and if the extension is part of the RTP caps.
+
+Automatic packet loss, data corruption and keyframe request handling for video decoders
+
+-   The GstVideoDecoder base class has gained various new APIs to
+    automatically handle packet loss and data corruption better by
+    default, especially in RTP, RTSP and WebRTC streaming scenarios, and
+    to give subclasses more control about how they want to handle
+    missing data:
+
+    -   Video decoder subclasses can mark output frames as corrupted via
+        the new GST_VIDEO_CODEC_FRAME_FLAG_CORRUPTED flag
+
+    -   A new "discard-corrupted-frames" property allows applications to
+        configure decoders so that corrupted frames are directly
+        discarded instead of being forwarded inside the pipeline. This
+        is a replacement for the "output-corrupt" property of the FFmpeg
+        decoders.
+
+    -   RTP depayloaders can now signal to decoders that data is missing
+        when sending GAP events for lost packets. GAP events can be sent
+        for various reason in a GStreamer pipeline. Often they are just
+        used to let downstream elements know that there isn’t a buffer
+        available at the moment, so downstream elements can move on
+        instead of waiting for one. They are also sent by RTP
+        depayloaders in the case that packets are missing, however, and
+        so far a decoder was not able to differentiate the two cases.
+        This has been remedied now: GAP events can be decorated with
+        gst_event_set_gap_flags() and GST_GAP_FLAG_MISSING_DATA to let
+        decoders now what happened, and decoders can then use that in
+        some cases to handle missing data better.
+
+    -   The GstVideoDecoder::handle_missing_data vfunc was added to
+        inform subclasses about packet loss or missing data and let them
+        handle it in their own way if they like.
+
+    -   gst_video_decoder_set_needs_sync_point() lets subclasses signal
+        that they need the stream to start with a sync point. If
+        enabled, the base class will discard all non-sync point frames
+        in the beginning and after a flush and does not pass them to the
+        subclass. Furthermore, if the first frame is not a sync point,
+        the base class will try and request a sync frame from upstream
+        by sending a force-key-unit event (see next items).
+
+    -   New "automatic-request-sync-points" and
+        "automatic-request-sync-point-flags" properties to automatically
+        request sync points when needed, e.g. on packet loss or if the
+        first frame is not a keyframe. Applications may want to enable
+        this on decoders operating in e.g. RTP/WebRTC/RTSP receiver
+        pipelines.
+
+    -   The new "min-force-key-unit-interval" property can be used to
+        ensure there’s a minimal interval between keyframe requests to
+        upstream (and/or the sender) and we’re not flooding the sender
+        with key unit requests.
+
+    -   gst_video_decoder_request_sync_point() allows subclasses to
+        request a new sync point (e.g. if they choose to do their own
+        missing data handling). This will still honour the
+        "min-force-key-unit-interval" property if set.
+
+Improved support for custom minimal GStreamer builds
+
+-   Element registration and registration of other plugin features
+    inside plugin init functions has been improved in order to
+    facilitate minimal custom GStreamer builds.
+
+-   A number of new macros have been added to declare and create
+    per-element and per-pluginfeature register functions in all plugins,
+    and then call those from the per-plugin plugin_init functions:
+
+    -   GST_ELEMENT_REGISTER_DEFINE,
+        GST_DEVICE_PROVIDER_REGISTER_DEFINE,
+        GST_DYNAMIC_TYPE_REGISTER_DEFINE, GST_TYPE_FIND_REGISTER_DEFINE
+        for the actual registration call with GStreamer
+    -   GST_ELEMENT_REGISTER, GST_DEVICE_PROVIDER_REGISTER,
+        GST_DYNAMIC_TYPE_REGISTER, GST_PLUGIN_STATIC_REGISTER,
+        GST_TYPE_FIND_REGISTER to call the registration function defined
+        by the REGISTER_DEFINE macro
+    -   GST_ELEMENT_REGISTER_DECLARE,
+        GST_DEVICE_PROVIDER_REGISTER_DECLARE,
+        GST_DYNAMIC_TYPE_REGISTER_DECLARE,
+        GST_TYPE_FIND_REGISTER_DECLARE to declare the registration
+        function defined by the REGISTER_DEFINE macro
+    -   and various variants for advanced use cases.
+
+-   This means that applications can call the per-element and
+    per-pluginfeature registration functions for only the elements they
+    need instead of registering plugins as a whole with all kinds of
+    elements that may not be required (e.g. encoder and decoder instead
+    of just decoder). In case of static linking all unused functions and
+    their dependencies would be removed in this case by the linker,
+    which helps minimise binary size for custom builds.
+
+-   gst_init() will automatically call a gst_init_static_plugins()
+    function if one exists.
+
+-   See the GStreamer static build documentation and Stéphane’s blog
+    post Generate a minimal GStreamer build, tailored to your needs for
+    more details.
 
 New elements
 
--   this section will be filled in in due course
+-   New aesdec and aesenc elements for AES encryption and decryption in
+    a custom format.
+
+-   New encodebin2 element with dynamic/sometimes source pads in order
+    to support the option of doing the muxing outside of encodebin,
+    e.g. in combination with a splitmuxsink.
+
+-   New fakeaudiosink and videocodectestsink elements for testing and
+    debugging (see below for more details)
+
+-   rtpisacpay, rtpisacdepay: new RTP payloader and depayloader for iSAC
+    audio codec
+
+-   rtpst2022-1-fecdec, rtpst2022-1-fecenc: new elements providing SMPTE
+    2022-1 2-D Forward Error Correction. More details in Mathieu’s blog
+    post.
+
+-   isac: new plugin wrapping the Internet Speech Audio Codec reference
+    encoder and decoder from the WebRTC project.
+
+-   asio: plugin for Steinberg ASIO (Audio Streaming Input/Output) API
+
+-   gssrc, gssink: add source and sink for Google Cloud Storage
+
+-   onnx: new plugin to apply ONNX neural network models to video
+
+-   openaptx: aptX and aptX-HD codecs using libopenaptx (v0.2.0)
+
+-   qroverlay, debugqroverlay: new elements that allows overlaying data
+    on top of video in form of a QR code
+
+-   cvtracker: new OpenCV-based tracker element
+
+-   av1parse, vp9parse: new parsers for AV1 and VP9 video
+
+-   va: work on the new VA-API plugin implementation for
+    hardware-accelerated video decoding and encoding has continued at
+    pace, with various new decoders and filters having joined the
+    initial vah264dec:
+
+    -   vah265dec: VA-API H.265 decoder
+    -   vavp8dec: VA-API VP8 decoder
+    -   vavp9dec: VA-API VP9 decoder
+    -   vaav1dec: VA-API AV1 decoder
+    -   vampeg2dec: VA-API MPEG-2 decoder
+    -   vadeinterlace: : VA-API deinterlace filter
+    -   vapostproc: : VA-API postproc filter (color conversion,
+        resizing, cropping, color balance, video rotation, skin tone
+        enhancement, denoise, sharpen)
+
+    See Víctor’s blog post “GstVA in GStreamer 1.20” for more details
+    and what’s coming up next.
+
+-   vaapiav1dec: new AV1 decoder element (in gstreamer-vaapi)
+
+-   msdkav1dec: hardware-accelerated AV1 decoder using the Intel Media
+    SDK / oneVPL
+
+-   nvcodec plugin for NVIDIA NVCODEC API for hardware-accelerated video
+    encoding and decoding:
+
+    -   cudaconvert, cudascale: new CUDA based video color space convert
+        and rescale elements
+    -   cudaupload, cudadownload: new helper elements for memory
+        transfer between CUDA and system memory spaces
+    -   nvvp8sldec, nvvp9sldec: new GstCodecs-based VP8/VP9 decoders
+
+-   Various new hardware-accelerated elements for Windows:
+
+    -   d3d11screencapturesrc: new desktop capture element, including a
+        GstDeviceProvider implementation to enumerate/select target
+        monitors for capture.
+    -   d3d11av1dec and d3d11mpeg2dec: AV1 and MPEG-2 decoders
+    -   d3d11deinterlace: deinterlacing filter
+    -   d3d11compositor: video composing element
+    -   see Windows section below for more details
+
+-   new Rust plugins:
+
+    -   audiornnoise: Removes noise from an audio stream
+    -   awstranscribeparse: Parses AWS audio transcripts into timed text
+        buffers
+    -   ccdetect: Detects if valid closed captions are present in a
+        closed captions stream
+    -   cea608tojson: Converts CEA-608 Closed Captions to a JSON
+        representation
+    -   cmafmux: CMAF fragmented MP4 muxer
+    -   dashmp4mux: DASH fragmented MP4 muxer
+    -   isofmp4mux: ISO fragmented MP4 muxer
+    -   ebur128level: EBU R128 Loudness Level Measurement
+    -   ffv1dec: FFV1 video decoder
+    -   gtk4paintablesink: GTK4 video sink, which provides a
+        GdkPaintable that can be rendered in various widgets
+    -   hlssink3: HTTP Live Streaming sink
+    -   hrtfrender: Head-Related Transfer Function (HRTF) renderer
+    -   hsvdetector: HSV colorspace detector
+    -   hsvfilter: HSV colorspace filter
+    -   jsongstenc: Wraps buffers containing any valid top-level JSON
+        structures into higher level JSON objects, and outputs those as
+        ndjson
+    -   jsongstparse: Parses ndjson as output by jsongstenc
+    -   jsontovtt: converts JSON to WebVTT subtitles
+    -   regex: Applies regular expression operations on text
+    -   roundedcorners: Adds rounded corners to video
+    -   spotifyaudiosrc: Spotify source
+    -   textahead: Display upcoming text buffers ahead (e.g. for
+        Karaoke)
+    -   transcriberbin: passthrough bin that transcribes raw audio to
+        closed captions using awstranscriber and puts the captions as
+        metas onto the video
+    -   tttojson: Converts timed text to a JSON representation
+    -   uriplaylistbin: Playlist source bin
+    -   webpdec-rs: WebP image decoder with animation support
+
+-   New plugin codecalpha with elements to assist with WebM Alpha
+    decoding
+
+    -   codecalphademux: Split stream with GstVideoCodecAlphaMeta into
+        two streams
+    -   alphacombine: Combine two raw video stream (I420 or NV12) as one
+        stream with alpha channel (A420 or AV12)
+    -   vp8alphadecodebin: A bin to handle software decoding of VP8 with
+        alpha
+    -   vp9alphadecodebin: A bin to handle software decoding of VP9 with
+        alpha
+
+-   New hardware accelerated elements for Linux:
+
+    -   v4l2slmpeg2dec: Support for Linux Stateless MPEG2 decoders
+    -   v4l2slvp9dec: Support for Linux Stateless VP9 decoders
+    -   v4l2slvp8alphadecodebin: Support HW accelerated VP8 with alpha
+        layer decoding
+    -   v4l2slvp9alphadecodebin: Support HW accelerated VP9 with alpha
+        layer decoding
 
 New element features and additions
 
--   this section will be filled in in due course
+-   assrender: handle more font mime types; better interaction with
+    matroskademux for embedded fonts
+
+-   audiobuffersplit: Add support for specifying output buffer size in
+    bytes (not just duration)
+
+-   audiolatency: new "samplesperbuffer" property so users can configure
+    the number of samples per buffer. The default value is 240 samples
+    which is equivalent to 5ms latency with a sample rate of 48000,
+    which might be larger than actual buffer size of audio capture
+    device.
+
+-   audiomixer, audiointerleave, GstAudioAggregator: now keep a count of
+    samples that are dropped or processed as statistic and can be made
+    to post QoS messages on the bus whenever samples are dropped by
+    setting the "qos-messages" property on input pads.
+
+-   audiomixer, compositor: improved handling of new inputs added at
+    runtime. New API was added to the GstAggregator base class to allow
+    subclasses to opt into an aggregation mode where inactive pads are
+    ignored when processing input buffers
+    (gst_aggregator_set_ignore_inactive_pads(),
+    gst_aggregator_pad_is_inactive()). An “inactive pad” in this context
+    is a pad which, in live mode, hasn’t yet received a first buffer,
+    but has been waited on at least once. What would happen usually in
+    this case is that the aggregator would wait for data on this pad
+    every time, up to the maximum configured latency. This would
+    inadvertently push mixer elements in live mode to the configured
+    latency envelope and delay processing when new inputs are added at
+    runtime until these inputs have actually produced data. This is
+    usually undesirable. With this new API, new inputs can be added
+    (requested) and configured and they won’t delay the data processing.
+    Applications can opt into this new behaviour by setting the
+    "ignore-inactive-pads" property on compositor, audiomixer or other
+    GstAudioAggregator-based elements.
+
+-   cccombiner: implement “scheduling” of captions. So far cccombiner’s
+    behaviour was essentially that of a funnel: it strictly looked at
+    input timestamps to associate together video and caption buffers.
+    Now it will try to smoothly schedule caption buffers in order to
+    have exactly one per output video buffer. This might involve
+    rewriting input captions, for example when the input is CDP then
+    sequence counters are rewritten, time codes are dropped and
+    potentially re-injected if the input video frame had a time code
+    meta. This can also lead to the input drifting from synchronization,
+    when there isn’t enough padding in the input stream to catch up. In
+    that case the element will start dropping old caption buffers once
+    the number of buffers in its internal queue reaches a certain limit
+    (configurable via the "max-scheduled" property). The new original
+    funnel-like behaviour can be restored by setting the "scheduling"
+    property to FALSE.
+
+-   ccconverter: new "cdp-mode" property to specify which sections to
+    include in CDP packets (timecode, CC data, service info). Various
+    software, including ffmpeg’s Decklink support, fails parsing CDP
+    packets that contain anything but CC data in the CDP packets.
+
+-   clocksync: new "sync-to-first" property for automatic timestamp
+    offset setup: if set clocksync will set up the "ts-offset" value
+    based on the first buffer and the pipeline’s running time when the
+    first buffer arrived. The newly configured "ts-offset" in this case
+    would be the value that allows outputting the first buffer without
+    waiting on the clock. This is useful for example to feed a non-live
+    input into an already-running pipeline.
+
+-   compositor:
+
+    -   multi-threaded input conversion and compositing. Set the
+        "max-threads" property to activate this.
+    -   new "sizing-policy" property to support display aspect ratio
+        (DAR)-aware scaling. By default the image is scaled to fill the
+        configured destination rectangle without padding and without
+        keeping the aspect ratio. With sizing-policy=keep-aspect-ratio
+        the input image is scaled to fit the destination rectangle
+        specified by GstCompositorPad:{xpos, ypos, width, height}
+        properties preserving the aspect ratio. As a result, the image
+        will be centered in the destination rectangle with padding if
+        necessary.
+    -   new "zero-size-is-unscaled" property on input pads. By default
+        pad width=0 or pad height=0 mean that the stream should not be
+        scaled in that dimension. But if the "zero-size-is-unscaled"
+        property is set to FALSE a width or height of 0 is instead
+        interpreted to mean that the input image on that pad should not
+        be composited, which is useful when creating animations where an
+        input image is made smaller and smaller until it disappears.
+    -   improved handling of new inputs at runtime via
+        "ignore-inactive-pads"property (see above for details)
+    -   allow output format with alpha even if none of the inputs have
+        alpha (also glvideomixer and other GstVideoAggregator
+        subclasses)
+
+-   dashsink: add h265 codec support and signals for allowing custom
+    playlist/fragment output
+
+-   decodebin3:
+
+    -   improved decoder selection, especially for hardware decoders
+    -   make input activation “atomic” when adding inputs dynamically
+    -   better interleave handling: take into account decoder latency
+        for interleave size
+
+-   decklink:
+
+    -   Updated DeckLink SDK to 11.2 to support DeckLink 8K Pro
+    -   decklinkvideosrc:
+        -   More accurate and stable capture timestamps: use the
+            hardware reference clock time when the frame was finished
+            being captured instead of a clock time much further down the
+            road.
+        -   Automatically detect widescreen vs. normal NTSC/PAL
+
+-   encodebin:
+
+    -   add “smart encoding” support for H.265, VP8 and VP9 (i.e. only
+        re-encode where needed and otherwise pass through encoded video
+        as-is).
+    -   H264/H265 smart encoding improvements: respect user-specified
+        stream-format, but if not specified default to avc3/hvc1 with
+        in-band SPS/PPS/VPS signalling for more flexibility.
+    -   new encodebin2 element with dynamic/sometimes source pads in
+        order to support the option of doing the muxing outside of
+        encodebin, e.g. in combination with splitmuxsink.
+    -   add APIs to set element properties on encoding profiles (see
+        below)
+
+-   errorignore: new "ignore-eos" property to also ignore FLOW_EOS from
+    downstream elements
+
+-   giosrc: add support for growing source files: applications can
+    specify that the underlying file being read is growing by setting
+    the "is-growing" property. If set, the source won’t EOS when it
+    reaches the end of the file, but will instead start monitoring it
+    and will start reading data again whenever a change is detected. The
+    new "waiting-data" and "done-waiting-data" signals keep the
+    application informed about the current state.
+
+-   gtksink, gtkglsink:
+
+    -   scroll event support: forwarded as navigation events into the
+        pipeline
+    -   "video-aspect-ratio-override" property to force a specific
+        aspect ratio
+    -   "rotate-method" property and support automatic rotation based on
+        image tags
+
+-   identity: new "stats" property allows applications to retrieve the
+    number of bytes and buffers that have passed through so far.
+
+-   interlace: add support for more formats, esp 10-bit, 12-bit and
+    16-bit ones
+
+-   jack: new "low-latency" property for automatic latency-optimized
+    setting and "port-names" property to select ports explicitly
+
+-   jpegdec: support output conversion to RGB using libjpeg-turbo (for
+    certain input files)
+
+-   line21dec:
+
+    -   "mode" property to control whether and how detected closed
+        captions should be inserted in the list of existing close
+        caption metas on the input frame (if any): add, drop, or
+        replace.
+    -   "ntsc-only" property to only look for captions if video has NTSC
+        resolution
+
+-   line21enc: new "remove-caption-meta" to remove metas from output
+    buffers after encoding the captions into the video data; support for
+    CDP closed captions
+
+-   matroskademux, matroskamux: Add support for ffv1, a lossless
+    intra-frame video coding format.
+
+-   matroskamux: accept in-band SPS/PPS/VPS for H264 and H265
+    (i.e. stream-format avc3 and hev1) which allows on-the-fly
+    profile/level/resolution changes.
+
+-   matroskamux: new "cluster-timestamp-offset" property, useful for use
+    cases where the container timestamps should map to some absolute
+    wall clock time, for example.
+
+-   rtpsrc: add "caps" property to allow explicit setting of the caps
+    where needed
+
+-   mpegts: support SCTE-35 passthrough via new "send-scte35-events"
+    property on MPEG-TS demuxer tsdemux. When enabled, SCTE 35 sections
+    (eg ad placement opportunities) are forwarded as events donwstream
+    where they can be picked up again by mpegtsmux. This required a
+    semantic change in the SCTE-35 section API: timestamps are now in
+    running time instead of muxer pts.
+
+-   tsdemux: Handle PCR-less MPEG-TS streams; more robust timestamp
+    handling in certain corner cases and for poorly muxed streams.
+
+-   mpegtsmux:
+
+    -   More conformance improvements to make MPEG-TS analyzers happy:
+        -   PCR timing accuracy: Improvements to the way mpegtsmux
+            outputs PCR observations in CBR mode, so that a PCR
+            observation is always inserted when needed, so that we never
+            miss the configured pcr-interval, as that triggers various
+            MPEG-TS analyser errors.
+        -   Improved PCR/SI scheduling
+    -   Don’t write PCR until PAT/PMT are output to make sure streams
+        start cleanly with a PAT/PMT.
+    -   Allow overriding the automatic PMT PID selection via
+        application-supplied PMT_%d fields in the prog-map
+        structure/property.
+
+-   mp4mux:
+
+    -   new "first-moov-then-finalise" mode for fragmented output where
+        the output will start with a self-contained moov atom for the
+        first fragment, and then produce regular fragments. Then at the
+        end when the file is finalised, the initial moov is invalidated
+        and a new moov is written covering the entire file. This way the
+        file is a “fragmented mp4” file while it is still being written
+        out, and remains playable at all times, but at the end it is
+        turned into a regular mp4 file (with former fragment headers
+        remaining as unused junk data in the file).
+    -   support H.264 avc3 and H.265 hvc1 stream formats as input where
+        the codec data is signalled in-band inside the bitstream instead
+        of caps/file headers.
+    -   support profile/level/resolution changes for H264/H265 input
+        streams (i.e. codec data changing on the fly). Each codec_data
+        is put into its own SampleTableEntry inside the stsd, unless the
+        input is in avc3 stream format in which case it’s written
+        in-band an not in the headers.
+
+-   multifilesink: new ""min-keyframe-distance"" property to make
+    minimum distance between keyframes in next-file=key-frame mode
+    configurable instead of hard-coding it to 10 seconds.
+
+-   mxfdemux has seen a big refactoring to support non-frame wrappings
+    and more accurate timestamp/seek handling for some formats
+
+-   msdk plugin for hardware-accelerated video encoding and decoding
+    using the Intel Media SDK:
+
+    -   oneVPL support (Intel oneAPI Video Processing Library)
+    -   AV1 decoding support
+    -   H.264 decoder now supports constrained-high and progressive-high
+        profiles
+    -   H.264 encoder:
+        -   more configuration options (properties):
+            "intra-refresh-type", "min-qp" , "max-qp", "p-pyramid",
+            "dblk-idc"
+    -   H.265 encoder:
+        -   can output main-still-picture profile
+        -   now inserts HDR SEIs (mastering display colour volume and
+            content light level)
+        -   more configuration options (properties):
+            "intra-refresh-type", "min-qp" , "max-qp", "p-pyramid",
+            "b-pyramid", "dblk-idc", "transform-skip"
+        -   support for RGB 10bit format
+    -   External bitrate control in encoders
+    -   Video post proc element msdkvpp gained support for 12-bit pixel
+        formats P012_LE, Y212_LE and Y412_LE
+
+-   nvh264sldec: interlaced stream support
+
+-   openh264enc: support main, high, constrained-high and
+    progressive-high profiles
+
+-   openjpeg: support for multithreaded decoding and encoding
+
+-   rtspsrc: now supports IPv6 also for tunneled mode (RTSP-over-HTTP);
+    new "ignore-x-server-reply" property to ignore the
+    x-server-ip-address server header reply in case of HTTP tunneling,
+    as it is often broken.
+
+-   souphttpsrc: Runtime compatibility support for libsoup2 and
+    libsoup3. libsoup3 is the latest major version of libsoup, but
+    libsoup2 and libsoup3 can’t co-exist in the same process because
+    there is no namespacing or versioning for GObject types. As a
+    result, it would be awkward if the GStreamer souphttpsrc plugin
+    linked to a specific version of libsoup, because it would only work
+    with applications that use the same version of libsoup. To make this
+    work, the soup plugin now tries to determine the libsoup version
+    used by the application (and its other dependencies) at runtime on
+    systems where GStreamer is linked dynamically. libsoup3 support is
+    still considered somewhat experimental at this point.
+
+-   srtsrc, srtsink: add signals for the application to accept/reject
+    incoming connections
+
+-   timeoverlay: new elapsed-running-time time mode which shows the
+    running time since the first running time (and each flush-stop).
+
+-   udpsrc: new timestamping mode to retrieve packet receive timestamps
+    from the kernel via socket control messages (SO_TIMESTAMPNS) on
+    supported platforms
+
+-   uritranscodebin: new setup-source and element-setup signals for
+    applications to configure elements used
+
+-   v4l2codecs plugin gained support for 4x4 and 32x32 tile formats
+    enabling some platforms or direct renders. Important memory usage
+    improvement.
+
+-   v4l2slh264dec now implements the final Linux uAPI as shipped on
+    Linux 5.11 and later.
+
+-   valve: add "drop-mode" property and provide two new modes of
+    operation: in drop-mode=forward-sticky-events sticky events
+    (stream-start, segment, tags, caps, etc.) are forwarded downstream
+    even when dropping is enabled; drop-mode=transform-to-gap will in
+    addition also convert buffers into gap events when dropping is
+    enabled, which lets downstream elements know that time is advancing
+    and might allow for preroll in many scenarios. By default all events
+    and all buffers are dropped when dropping is enabled, which can
+    cause problems with caps negotiation not progressing or branches not
+    prerolling when dropping is enabled.
+
+-   videocrop: support for many more pixel formats, e.g. planar YUV
+    formats with > 8bits and GBR* video formats; can now also accept
+    video not backed by system memory as long as downstream supports the
+    GstCropMeta
+
+-   videotestsrc: new smpte-rp-219 pattern for SMPTE75 RP-219 conformant
+    color bars
+
+-   vp8enc: finish support for temporal scalability: two new properties
+    ("temporal-scalability-layer-flags",
+    "temporal-scalability-layer-sync-flags") and a unit change on the
+    "temporal-scalability-target-bitrate" property (now expects bps);
+    also make temporal scalability details available to RTP payloaders
+    as buffer metadata.
+
+-   vp9enc: new properties to tweak encoder performance:
+
+    -   "aq-mode" to configure adaptive quantization modes
+    -   "frame-parallel-decoding" to configure whether to create a
+        bitstream that reduces decoding dependencies between frames
+        which allows staged parallel processing of more than one video
+        frames in the decoder. (Defaults to TRUE)
+    -   "row-mt", "tile-columns" and "tile-rows" so multithreading can
+        be enabled on a per-tile basis, instead of on a per tile-column
+        basis. In combination with the new "tile-rows" property, this
+        allows the encoder to make much better use of the available CPU
+        power.
+
+-   vp9dec, vp9enc: add support for 10-bit 4:2:0 and 4:2:2 YUV, as well
+    as 8-bit 4:4:4
+
+-   vp8enc, vp9enc now default to “good quality” for the deadline
+    property rather then “best quality”. Having the deadline set to best
+    quality causes the encoder to be absurdly slow, most real-life users
+    will prefer good-enough quality with better performance instead.
+
+-   wpesrc:
+
+    -   implement audio support: a new sometimes source pad will be
+        created for each audio stream created by the web engine.
+    -   move wpesrc to wpevideosrc and add a wrapper bin wpesrc to also
+        support audio
+    -   also handles web:// URIs now (same as cefsrc)
+    -   post messages with the estimated load progress on the bus
 
-Plugin and library moves
+-   x265enc: add negative DTS support, which means timestamps are now
+    offset by 1h same as with x264enc
+
+RTP Payloaders and Depayloaders
+
+-   rtpisacpay, rtpisacdepay: new RTP payloader and depayloader for iSAC
+    audio codec
+
+-   rtph264depay:
+
+    -   new "request-keyframe" property to make the depayloader
+        automatically request a new keyframe from the sender on packet
+        loss, consistent with the new property on rtpvp8depay.
+    -   new "wait-for-keyframe" property to make depayloader wait for a
+        new keyframe at the beginning and after packet loss (only
+        effective if the depayloader outputs AUs), consistent with the
+        existing property on rtpvp8depay.
+
+-   rtpopuspay, rtpopusdepay: support libwebrtc-compatible multichannel
+    audio in addition to the previously supported multichannel audio
+    modes
+
+-   rtpopuspay: add DTX (Discontinuous Transmission) support
+
+-   rtpvp8depay: new "request-keyframe" property to make the depayloader
+    automatically request a new keyframe from the sender on packet loss.
+
+-   rtpvp8pay: temporal scaling support
 
--   this section will be filled in in due course
+-   rtpvp9depay: Improved SVC handling (aggregate all layers)
+
+RTP Infrastructure
+
+-   rtpst2022-1-fecdec, rtpst2022-1-fecenc: new elements providing SMPTE
+    2022-1 2-D Forward Error Correction. More details in Mathieu’s blog
+    post.
+
+-   rtpreddec: BUNDLE support
+
+-   rtpredenc, rtpulpfecenc: add support for Transport-wide Congestion
+    Control (TWCC)
+
+-   rtpsession: new "twcc-feedback-interval" property to allow RTCP TWCC
+    reports to be scheduled on a timer instead of per marker-bit.
+
+Plugin and library moves
 
 -   There were no plugin moves or library moves in this cycle.
 
@@ -56,8 +998,6 @@ Plugin removals
 
 The following elements or plugins have been removed:
 
--   this section will be filled in in due course
-
 -   The ofa audio fingerprinting plugin has been removed. The MusicIP
     database has been defunct for years so this plugin is likely neither
     useful nor used by anyone.
@@ -71,135 +1011,575 @@ The following elements or plugins have been removed:
 
 Miscellaneous API additions
 
--   this section will be filled in in due course
+Core
+
+-   gst_buffer_new_memdup() is a convenience function for the
+    widely-used gst_buffer_new_wrapped(g_memdup(data,size),size)
+    pattern.
+
+-   gst_caps_features_new_single() creates a new single GstCapsFeatures,
+    avoiding the need to use the vararg function with NULL terminator
+    for simple cases.
+
+-   gst_element_type_set_skip_documentation() can be used by plugins to
+    signal that certain elements should not be included in the GStreamer
+    plugin documentation. This is useful for plugins where elements are
+    registered dynamically based on hardware capabilities and/or where
+    the available plugins and properties vary from system to system.
+    This is used in the d3d11 plugin for example to ensure that only the
+    list of default elements is advertised in the documentation.
+
+-   gst_type_find_suggest_empty_simple() is a new convenience function
+    for typefinders for cases where there’s only a media type and no
+    other fields.
+
+-   New API to create elements and set properties at construction time,
+    which is not only convenient, but also allows GStreamer elements to
+    have construct-only properties: gst_element_factory_make_full(),
+    gst_element_factory_make_valist(),
+    gst_element_factory_make_with_properties(),
+    gst_element_factory_create_full(),
+    gst_element_factory_create_valist(),
+    gst_element_factory_create_with_properties().
+
+-   GstSharedTaskPool: new “shared” task pool subclass with slightly
+    different default behaviour than the existing GstTaskPool which
+    would create unlimited number of threads for new tasks. The shared
+    taskpool creates up to N threads (default: 1) and then distributes
+    pending tasks to those threads round-robin style, and blocks if no
+    thread is available. It is possible to join tasks. This can be used
+    by plugins to implement simple multi-threaded processing and is used
+    for the new multi-threaded video conversion and compositing done in
+    GstVideoAggregator, videoconverter and compositor.
+
+Plugins Base Utils library
+
+-   GstDiscoverer:
+
+    -   gst_discoverer_container_info_get_tags() was added to retrieve
+        global/container tags (vs. per-stream tags). Per-Stream tags can
+        be retrieved via the existing
+        gst_discoverer_stream_info_get_tags().
+        gst_discoverer_info_get_tags(), which for many files returns a
+        confusing mix of stream and container tags, has been deprecated
+        in favour of the container/stream-specific functions.
+    -   gst_discoverer_stream_info_get_stream_number() returns a unique
+        integer identifier for a given stream within the given
+        GstDiscoverer context. (If this matches the stream number inside
+        the container bitstream that’s by coincidence and not by
+        design.)
+
+-   gst_pb_utils_get_caps_description_flags() can be used to query
+    whether certain caps represent a container, audio, video, image,
+    subtitles, tags, or something else. This only works for formats
+    known to GStreamer.
+
+-   gst_pb_utils_get_file_extension_from_caps() returns a possible file
+    extension for given caps.
+
+-   gst_codec_utils_h264_get_profile_flags_level(): Parses profile,
+    flags, and level from H264 AvcC codec_data. The format of H264 AVCC
+    extradata/sequence_header is documented in the ITU-T H.264
+    specification section 7.3.2.1.1 as well as in ISO/IEC 14496-15
+    section 5.3.3.1.2.
+
+-   gst_codec_utils_caps_get_mime_codec() to convert caps to a RFC 6381
+    compatible MIME codec string codec. Useful for providing the codecs
+    field inside the Content-Type HTTP header for containerized formats,
+    such as mp4 or matroska.
+
+GStreamer OpenGL integration library and plugins
+
+-   glcolorconvert: added suppport for converting the video formats
+    A420, AV12, BGR, BGRA, RGBP and BGRP.
+
+-   Added support to GstGLBuffer for persistent buffer mappings where a
+    Pixel Buffer Object (PBO) can be mapped by both the CPU and the GPU.
+    This removes a memcpy() when uploading textures or vertices
+    particularly when software decoders (e.g. libav) are direct
+    rendering into our memory. Improves transfer performance
+    significantly. Requires OpenGL 4.4, GL_ARB_buffer_storage or
+    GL_EXT_buffer_storage
+
+-   Added various helper functions for handling 4x4 matrices of affine
+    transformations as used by GstVideoAffineTransformationMeta.
+
+-   Add support to GstGLContext for allowing the application to control
+    the config (EGLConfig, GLXConfig, etc) used when creating the OpenGL
+    context. This allows the ability to choose between RGB16 or RGB10A2
+    or RGBA8 back/front buffer configurations that were previously
+    hardcoded. GstGLContext also supports retrieving the configuration
+    it was created with or from an externally provide OpenGL context
+    handle. This infrastructure is also used to create a compatible
+    config from an application/externally provided OpenGL context in
+    order to improve compatibility with other OpenGL frameworks and GUI
+    toolkits. A new environment variable GST_GL_CONFIG was also added to
+    be able to request a specific configuration from the command line.
+    Note: different platforms will have different functionality
+    available.
+
+-   Add support for choosing between EGL and WGL at runtime when running
+    on Windows. Previously this was a build-time switch. Allows use in
+    e.g. Gtk applications on Windows that target EGL/ANGLE without
+    recompiling GStreamer. gst_gl_display_new_with_type() can be used by
+    applications to choose a specific display type to use.
+
+-   Build fixes to explicitly check for Broadcom-specific libraries on
+    older versions of the Raspberry Pi platform. The Broadcom OpenGL ES
+    and EGL libraries have different filenames. Using the vc4 Mesa
+    driver on the Raspberry Pi is not affected.
+
+-   Added support to glupload and gldownload for transferring RGBA
+    buffers using the memory:NVMM available on the Nvidia Tegra family
+    of embedded devices.
+
+-   Added support for choosing libOpenGL and libGLX as used in a GLVND
+    environment on unix-based platforms. This allows using desktop
+    OpenGL and EGL without pulling in any GLX symbols as would be
+    required with libGL.
+
+Video library
+
+-   New raw video formats:
+
+    -   AV12 (NV12 with alpha plane)
+    -   RGBP and BGRP (planar RGB formats)
+    -   ARGB64 variants with specified endianness instead of host
+        endianness:
+        -   ARGB64_LE, ARGB64_BE
+        -   RGBA64_BE, RGBA64_LE
+        -   BGRA64_BE, BGRA64_LE
+        -   ABGR64_BE, ABGR64_LE
+
+-   gst_video_orientation_from_tag() is new convenience API to parse the
+    image orientation from a GstTagList.
+
+-   GstVideoDecoder subframe support (see below)
+
+-   GstVideoCodecState now also carries some HDR metadata
+
+-   Ancillary video data: implement transform functions for AFD/Bar
+    metas, so they will be forwarded in more cases
+
+MPEG-TS library
+
+This library only handles section parsing and such, see above for
+changes to the actual mpegtsmux and mpegtsdemux elements.
+
+-   many additions and improvements to SCTE-35 section parsing
+-   new API for fetching extended descriptors:
+    gst_mpegts_find_descriptor_with_extension()
+-   add support for SIT sections (Selection Information Tables)
+-   expose event-from-section constructor gst_event_new_mpegts_section()
+-   parse Audio Preselection Descriptor needed for Dolby AC-4
+
+GstWebRTC library + webrtcbin
+
+-   Change the way in which sink pads and transceivers are matched
+    together to support easier usage. If a pad is created without a
+    specific index (i.e. using sink_%u as the pad template), then an
+    available compatible transceiver will be searched for. If a specific
+    index is requested (i.e. sink_1) then if a transceiver for that
+    m-line already exists, that transceiver must match the new sink pad
+    request. If there is no transceiver available in either scenario, a
+    new transceiver is created. If a mixture of both sink_1 and sink_%u
+    requests result in an impossible situation, an error will be
+    produced at pad request time or from create offer/answer.
+
+-   webrtcbin now uses regular ICE nomination instead of libnice’s
+    default of aggressive ICE nomination. Regular ICE nomination is the
+    default recommended by various relevant standards and improves
+    connectivity in specific network scenarios.
+
+-   Add support for limiting the port range used for RTP with the
+    addition of the min-rtp-port and max-rtp-port properties on the ICE
+    object.
+
+-   Expose the SCTP transport as a property on webrtcbin to more closely
+    match the WebRTC specification.
+
+-   Added support for taking into account the data channel transport
+    state when determining the value of the "connection-state" property.
+    Previous versions of the WebRTC spec did not include the data
+    channel state when computing this value.
+
+-   Add configuration for choosing the size of the underlying sockets
+    used for transporting media data
+
+-   Always advertise support for the transport-cc RTCP feedback protocol
+    as rtpbin supports it. For full support, the configured caps (input
+    or through codec-preferences) need to include the relevant RTP
+    header extension.
+
+-   Numerous fixes to caps and media handling to fail-fast when an
+    incompatible situation is detected.
+
+-   Improved support for attaching the required media after a remote
+    offer has been set.
+
+-   Add support for dynamically changing the amount of FEC used for a
+    particular stream.
+
+-   webrtcbin now stops further SDP processing at the first error it
+    encounters.
+
+-   Completed support for either local or the remote closing a data
+    channel.
+
+-   Various fixes when performing BUNDLEing of the media streams in
+    relation to RTX and FEC usage.
+
+-   Add support for writing out QoS DSCP marking on outgoing packets to
+    improve reliability in some network scenarios.
+
+-   Improvements to the statistics returned by the get-stats signal
+    including the addition of the raw statistics from the internal
+    RTPSource, the TWCC stats when available.
+
+-   The webrtc library does not expose any objects anymore with public
+    fields. Instead properties have been added to replace that
+    functionality. If you are accessing such fields in your application,
+    switch to the corresponding properties.
+
+GstCodecs and Video Parsers
+
+-   Support for render delays to improve throughput across all CODECs
+    (used with NVDEC and V4L2).
+-   lots of improvements to parsers and the codec parsing decoder base
+    classes (H264, H265, VP8, VP9, AV1, MPEG-2) used for various
+    hardware-accelerate decoder APIs.
+
+Bindings support
+
+-   gst_allocation_params_new() allocates a GstAllocationParams struct
+    on the heap. This should only be used by bindings (and freed via
+    gst_allocation_params_free() then). In C code you would allocate
+    this on the stack and only init it in place.
+
+-   gst_debug_log_literal() can be used to log a string to the debug log
+    without going through any printf format expansion and associated
+    overhead. This is mostly useful for bindings such as the Rust
+    bindings which may have done their own formatting already .
+
+-   Provide non-inlined versions of refcounting APIs for various
+    GStreamer mini objects, so that they can be consumed by bindings
+    (e.g. gstreamer-sharp): gst_buffer_ref, gst_buffer_unref,
+    gst_clear_buffer, gst_buffer_copy, gst_buffer_replace,
+    gst_buffer_list_ref, gst_buffer_list_unref, gst_clear_buffer_list,
+    gst_buffer_list_copy, gst_buffer_list_replace, gst_buffer_list_take,
+    gst_caps_ref, gst_caps_unref, gst_clear_caps, gst_caps_replace,
+    gst_caps_take, gst_context_ref, gst_context_unref, gst_context_copy,
+    gst_context_replace, gst_event_replace, gst_event_steal,
+    gst_event_take, gst_event_ref, gst_event_unref, gst_clear_event,
+    gst_event_copy, gst_memory_ref, gst_memory_unref, gst_message_ref,
+    gst_message_unref, gst_clear_message, gst_message_copy,
+    gst_message_replace, gst_message_take, gst_promise_ref,
+    gst_promise_unref, gst_query_ref, gst_query_unref, gst_clear_query,
+    gst_query_copy, gst_query_replace, gst_query_take, gst_sample_ref,
+    gst_sample_unref, gst_sample_copy, gst_tag_list_ref,
+    gst_tag_list_unref, gst_clear_tag_list, gst_tag_list_replace,
+    gst_tag_list_take, gst_uri_copy, gst_uri_ref, gst_uri_unref,
+    gst_clear_uri.
+
+-   expose a GType for GstMiniObject
+
+-   gst_device_provider_probe() now returns non-floating device object
+
+API Deprecations
+
+-   gst_element_get_request_pad() has been deprecated in favour of the
+    newly-added gst_element_request_pad_simple() which does the exact
+    same thing but has a less confusing name that hopefully makes clear
+    that the function request a new pad rather than just retrieves an
+    already-existing request pad.
+
+-   gst_discoverer_info_get_tags(), which for many files returns a
+    confusing mix of stream and container tags, has been deprecated in
+    favour of the container-specific and stream-specific functions,
+    gst_discoverer_container_info_get_tags() and
+    gst_discoverer_stream_info_get_tags().
+
+-   gst_video_sink_center_rect() was deprecated in favour of the more
+    generic newly-added gst_video_center_rect().
+
+-   The GST_MEMORY_FLAG_NO_SHARE flag has been deprecated, as it tends
+    to cause problems and prevents sub-buffering. If pooling or lifetime
+    tracking is required, memories should be allocated through a custom
+    GstAllocator instead of relying on the lifetime of the buffers the
+    memories were originally attached to, which is fragile anyway.
+
+-   The GstPlayer high-level playback library is being replaced with the
+    new GstPlay library (see above). GstPlayer should be considered
+    deprecated at this point and will be marked as such in the next
+    development cycle. Applications should be ported to GstPlay.
+
+-   Gstreamer Editing Services: ges_video_transition_set_border(),
+    ges_video_transition_get_border()
+    ges_video_transition_set_inverted()
+    ges_video_transition_is_inverted() have been deprecated, use
+    ges_timeline_element_set_children_properties() instead.
 
 Miscellaneous performance, latency and memory optimisations
 
--   this section will be filled in in due course
+More video conversion fast paths
 
-Miscellaneous other changes and enhancements
+-   v210 ↔ I420, YV12, Y42B, UYVY and YUY2
+-   A420 → RGB
+
+Less jitter when waiting on the system clock
+
+-   Better system clock wait accuracy, less jitter: where available,
+    clock_nanosleep is used for higher accuracy for waits below 500
+    usecs, and waits below 2ms will first use the regular waiting system
+    and then clock_nanosleep for the remainder. The various wait
+    implementation have a latency ranging from 50 to 500+ microseconds.
+    While this is not a major issue when dealing with a low number of
+    waits per second (for ex: video), it does introduce a non-negligible
+    jitter for synchronisation of higher packet rate systems.
+
+Video decoder subframe support
 
--   this section will be filled in in due course
+-   The GstVideoDecoder base class gained API to process input at the
+    sub-frame level. That way video decoders can start decoding slices
+    before they have received the full input frame in its entirety (to
+    the extent this is supported by the codec, of course). This helps
+    with CPU utilisation and reduces latency.
 
-Tracing framework and debugging improvements
+-   This functionality is now being used in the OpenJPEG JPEG 2000
+    decoder, the FFmpeg H.264 decoder (in case of NAL-aligned input) and
+    the OpenMAX H.264/H.265 decoders (in case of NAL-aligned input).
 
--   this section will be filled in in due course
+Miscellaneous other changes and enhancements
+
+-   GstDeviceMonitor no longer fails to start just because one of the
+    device providers failed to start. That could happen for example on
+    systems where the pulseaudio device provider is installed, but
+    pulseaudio isn’t actually running but ALSA is used for audio
+    instead. In the same vein the device monitor now keeps track of
+    which providers have been started (via the new
+    gst_device_provider_is_started()) and only stops actually running
+    device providers when stopping the device monitor.
+
+-   On embedded systems it can be useful to create a registry that can
+    be shared and read by multiple processes running as different users.
+    It is now possible to set the new GST_REGISTRY_MODE environment
+    variable to specify the file mode for the registry file, which by
+    default is set to be only user readable/writable.
+
+-   GstNetClientClock will signal lost sync in case the remote time
+    resets (e.g. because device power cycles), by emitting the “synced”
+    signal with synced=FALSE parameter, so applications can take action.
+
+-   gst_value_deserialize_with_pspec() allows deserialization with a
+    hint for what the target GType should be. This allows for example
+    passing arrays of flags through the command line or
+    gst_util_set_object_arg(), eg: foo="<bar,bar+baz>".
+
+-   It’s now allowed to create an empty GstVideoOverlayComposition
+    without any rectangles by passing a NULL rectangle to
+    gst_video_overlay_composition_new(). This is useful for bindings and
+    simplifies application code in some places.
+
+Tracing framework, debugging and testing improvements
+
+-   New factories tracer to list loaded elements (and other plugin
+    features). This can be useful to collect a list of elements needed
+    for an application, which then in turn can be used to create a
+    tailored minimal GStreamer build that contains just the elements
+    needed and nothing else.
+-   New plugin-feature-loaded tracing hook for use by tracers like the
+    new factories tracer
+
+-   GstHarness: Add gst_harness_set_live() so that harnesses can be set
+    to non-live and return is-live=false in latency queries if needed.
+    Default behaviour is to always return is-live=true in latency
+    queries.
+
+-   navseek: new "hold-eos" property. When enabled, the element will
+    hold back an EOS event until the next keystroke (via navigation
+    events). This can be used to keep a video sink showing the last
+    frame of a video pipeline until a key is pressed instead of tearing
+    it down immediately on EOS.
+
+-   New fakeaudiosink element: mimics an audio sink and can be used for
+    testing and CI pipelines on systems where no audio system is
+    installed or running. It differs from fakesink in that it only
+    support audio caps and syncs to the clock by default like a normal
+    audio sink. It also implements the GstStreamVolume interface like
+    most audio sinks do.
+
+-   New videocodectestsink element for video codec conformance testing:
+    Calculates MD5 checksums for video frames and skips any padding
+    whilst doing so. Can optionally also write back the video data with
+    padding removed into a file for easy byte-by-byte comparison with
+    reference data.
 
 Tools
 
--   this section will be filled in in due course
+gst-inspect-1.0
+
+-   Can sort the list of plugins by passing --sort=name as command line
+    option
+
+gst-launch-1.0
+
+-   will now error out on top-level properties that don’t exist and
+    which were silently ignored before
+-   On Windows the high-resolution clock is enabled now, which provides
+    better clock and timer performance on Windows (see Windows section
+    below for more details).
+
+gst-play-1.0
+
+-   New --start-position command line argument to start playback from
+    the specified position
+-   Audio can be muted/unmuted in interactive mode by pressing the m
+    key.
+-   On Windows the high-resolution clock is enabled now (see Windows
+    section below for more details)
+
+gst-device-monitor-1.0
+
+-   New --include-hidden command line argument to also show “hidden”
+    device providers
+
+ges-launch-1.0
+
+-   New interactive mode that allows seeking and such. Can be disabled
+    by passing the --no-interactive argument on the command line.
+-   Option to forward tags
+-   Allow using an existing clip to determine the rendering format (both
+    topology and profile) via new --profile-from command line argument.
 
 GStreamer RTSP server
 
--   this section will be filled in in due course
+-   GstRTSPMediaFactory gained API to disable RTCP
+    (gst_rtsp_media_factory_set_enable_rtcp(), "enable-rtcp" property).
+    Previously RTCP was always allowed for all RTSP medias. With this
+    change it is possible to disable RTCP completely, no matter if the
+    client wants to do RTCP or not.
+
+-   Make a mount point of / work correctly. While not allowed by the
+    RTSP 2 spec, the RTSP 1 spec is silent on this and it is used in the
+    wild. It is now possible to use / as a mount path in
+    gst-rtsp-server, e.g. rtsp://example.com/ would work with this now.
+    Note that query/fragment parts of the URI are not necessarily
+    correctly handled, and behaviour will differ between various
+    client/server implementations; so use it if you must but don’t bug
+    us if it doesn’t work with third party clients as you’d hoped.
+
+-   multithreading fixes (races, refcounting issues, deadlocks)
+
+-   ONVIF audio backchannel fixes
+
+-   ONVIF trick mode optimisations
+
+-   rtspclientsink: new "update-sdp" signal that allows updating the SDP
+    before sending it to the server via ANNOUNCE. This can be used to
+    add additional metadata to the SDP, for example. The order and
+    number of medias must not be changed, however.
 
 GStreamer VAAPI
 
--   this section will be filled in in due course
+-   new AV1 decoder element (vaapiav1dec)
+
+-   H264 decoder: handle stereoscopic 3D video with frame packing
+    arrangement SEI messages
+
+-   H265 encoder: added Screen Content Coding extensions support
+
+-   H265 decoder: gained MAIN_444_12 profile support (decoded to
+    Y412_LE), and 4:2:2 12-bits support (decoded to Y212_LE)
+
+-   vaapipostproc: gained BT2020 color standard support
+
+-   vaapidecode: now generates caps templates dynamically at runtime in
+    order to advertise actually supported caps instead of all
+    theoretically supported caps.
+
+-   GST_VAAPI_DRM_DEVICE environment variable to force a specified DRM
+    device when a DRM display is used. It is ignored when other types of
+    displays are used. By default /dev/dri/renderD128 is used for DRM
+    display.
 
 GStreamer OMX
 
--   this section will be filled in in due course
+-   subframe support in H.264/H.265 decoders
 
 GStreamer Editing Services and NLE
 
--   this section will be filled in in due course
+-   framepositioner: new "operator" property to access blending modes in
+    the compositor
+-   timeline: Implement snapping to markers
+-   smart-mixer: Add support for d3d11compositor and glvideomixer
+-   titleclip: add "draw-shadow" child property
+-   ges:// URI support to define a timeline from a description.
+-   command-line-formatter
+    -   Add track management to timeline description
+    -   Add keyframe support
+-   ges-launch-1.0:
+    -   Add an interactive mode where we can seek etc…
+    -   Add option to forward tags
+    -   Allow using an existing clip to determine the rendering format
+        (both topology and profile) via new --profile-from command line
+        argument.
+-   Fix static build
 
 GStreamer validate
 
--   this section will be filled in in due course
+-   report: Add a way to force backtraces on reports even if not a
+    critical issue (GST_VALIDATE_ISSUE_FLAGS_FORCE_BACKTRACE)
+-   Add a flag to gst_validate_replace_variables_in_string() allow
+    defining how to resolve variables in structs
+-   Add gst_validate_bin_monitor_get_scenario() to get the bin monitor
+    scenario, which is useful for applications that use Validate
+    directly.
+-   Add an expected-values parameter to wait, message-type=XX allowing
+    more precise filtering of the message we are waiting for.
+-   Add config file support: each test can now use a config file for the
+    given media file used to test.
+-   Add support to check properties of object properties
+-   scenario: Add an "action-done" signal to signal when an action is
+    done
+-   scenario: Add a "run-command" action type
+-   scenario: Allow forcing running action on idle from scenario file
+-   scenario: Allow iterating over arrays in foreach
+-   scenario: Rename ‘interlaced’ action to ‘non-blocking’
+-   scenario: Add a non-blocking flag to the wait signal
 
 GStreamer Python Bindings
 
--   this section will be filled in in due course
+-   Fixes for Python 3.10
+-   Various build fixes
+-   at least one known breaking change caused by g-i annotation changes
+    (see below)
 
 GStreamer C# Bindings
 
--   this section will be filled in in due course
+-   Fix GstDebugGraphDetails enum
+-   Updated to latests GtkSharp
+-   Updated to include GStreamer 1.20 API
 
 GStreamer Rust Bindings and Rust Plugins
 
-The GStreamer Rust bindings are released separately with a different
-release cadence that’s tied to gtk-rs, but the latest release has
-already been updated for the upcoming new GStreamer 1.20 API.
-
-gst-plugins-rs, the module containing GStreamer plugins written in Rust,
-has also seen lots of activity with many new elements and plugins.
-
-What follows is a list of elements and plugins available in
-gst-plugins-rs, so people don’t miss out on all those potentially useful
-elements that have no C equivalent.
-
--   FIXME: add new elements
-
-Rust audio plugins
-
--   audiornnoise: New element for audio denoising which implements the
-    noise removal algorithm of the Xiph RNNoise library, in Rust
--   rsaudioecho: Port of the audioecho element from gst-plugins-good
-    rsaudioloudnorm: Live audio loudness normalization element based on
-    the FFmpeg af_loudnorm filter
--   claxondec: FLAC lossless audio codec decoder element based on the
-    pure-Rust claxon implementation
--   csoundfilter: Audio filter that can use any filter defined via the
-    Csound audio programming language
--   lewtondec: Vorbis audio decoder element based on the pure-Rust
-    lewton implementation
-
-Rust video plugins
-
--   cdgdec/cdgparse: Decoder and parser for the CD+G video codec based
-    on a pure-Rust CD+G implementation, used for example by karaoke CDs
--   cea608overlay: CEA-608 Closed Captions overlay element
--   cea608tott: CEA-608 Closed Captions to timed-text (e.g. VTT or SRT
-    subtitles) converter
--   tttocea608: CEA-608 Closed Captions from timed-text converter
--   mccenc/mccparse: MacCaption Closed Caption format encoder and parser
--   sccenc/sccparse: Scenarist Closed Caption format encoder and parser
--   dav1dec: AV1 video decoder based on the dav1d decoder implementation
-    by the VLC project
--   rav1enc: AV1 video encoder based on the fast and pure-Rust rav1e
-    encoder implementation
--   rsflvdemux: Alternative to the flvdemux FLV demuxer element from
-    gst-plugins-good, not feature-equivalent yet
--   rsgifenc/rspngenc: GIF/PNG encoder elements based on the pure-Rust
-    implementations by the image-rs project
-
-Rust text plugins
-
--   textwrap: Element for line-wrapping timed text (e.g. subtitles) for
-    better screen-fitting, including hyphenation support for some
-    languages
-
-Rust network plugins
-
--   reqwesthttpsrc: HTTP(S) source element based on the Rust
-    reqwest/hyper HTTP implementations and almost feature-equivalent
-    with the main GStreamer HTTP source souphttpsrc
--   s3src/s3sink: Source/sink element for the Amazon S3 cloud storage
--   awstranscriber: Live audio to timed text transcription element using
-    the Amazon AWS Transcribe API
-
-Generic Rust plugins
-
--   sodiumencrypter/sodiumdecrypter: Encryption/decryption element based
-    on libsodium/NaCl
--   togglerecord: Recording element that allows to pause/resume
-    recordings easily and considers keyframe boundaries
--   fallbackswitch/fallbacksrc: Elements for handling potentially
-    failing (network) sources, restarting them on errors/timeout and
-    showing a fallback stream instead
--   threadshare: Set of elements that provide alternatives for various
-    existing GStreamer elements but allow to share the streaming threads
-    between each other to reduce the number of threads
--   rsfilesrc/rsfilesink: File source/sink elements as replacements for
-    the existing filesrc/filesink elements
+-   The GStreamer Rust bindings are released separately with a different
+    release cadence that’s tied to gtk-rs, but the latest release has
+    already been updated for the upcoming new GStreamer 1.20 API (v1_20
+    feature).
+
+-   gst-plugins-rs, the module containing GStreamer plugins written in
+    Rust, has also seen lots of activity with many new elements and
+    plugins. See the New Elements section above for a list of new Rust
+    elements.
 
 Build and Dependencies
 
 -   Meson 0.59 or newer is required to build GStreamer now.
 
--   FIXME: this section will be filled in in due course
+-   The GLib requirement has been bumped to GLib 2.56 or newer (from
+    March 2018).
+
+-   The wpe plugin now requires wpe >= 2.28 and wpebackend-fdo >= 1.8
 
 Explicit opt-in required for build of certain plugins with (A)GPL dependencies
 
@@ -211,11 +1591,9 @@ even if the required dependencies are available.
 See Building plugins with (A)GPL-licensed dependencies for more details
 and a non-exhaustive list of plugins affected.
 
-gst-build: replaced by Monorepo
+gst-build: replaced by mono repository
 
--   this section will be filled in in due course
-
--   FIXME: describe + link to Monorepo FAQ
+See mono repository section above and the GStreamer mono repository FAQ.
 
 Cerbero
 
@@ -223,76 +1601,254 @@ Cerbero is a meta build system used to build GStreamer plus dependencies
 on platforms where dependencies are not readily available, such as
 Windows, Android, iOS and macOS.
 
-General improvements
+General Cerbero improvements
 
--   this section will be filled in in due course
+-   Plugin removed: libvisual
+-   New plugins: rtpmanagerbad and rist
 
-macOS / iOS
+macOS / iOS specific Cerbero improvements
 
--   this section will be filled in in due course
+-   XCode 12 support
+-   macOS OS release support is now future-proof, similar to iOS
+-   macOS Apple Silicon (ARM64) cross-compile support has been added
+-   macOS Apple Silicon (ARM64) native support is currently experimental
 
-Windows
+Windows specific Cerbero improvements
 
--   this section will be filled in in due course
+-   Visual Studio 2022 support has been added
+-   bootstrap is faster since it requires building fewer build-tools
+    recipes on Windows
+-   package is faster due to better scheduling of recipe stages and
+    elimination of unnecessary autotools regeneration
+-   The following plugins are no longer built on Windows:
+    -   a52dec (another decoder is still available in libav)
+    -   dvdread
+    -   resindvd
 
 Windows MSI installer
 
--   this section will be filled in in due course
+-   no major changes
 
-Linux
+Linux specific Cerbero improvements
 
--   this section will be filled in in due course
+-   Fedora, Debian OS release support is now more future-proof
+-   Amazon Linux 2 support has been added
 
-Android
+Android specific Cerbero improvements
 
--   this section will be filled in in due course
+-   no major changes
 
 Platform-specific changes and improvements
 
 Android
 
--   this section will be filled in in due course
+-   No major changes
 
 macOS and iOS
 
--   this section will be filled in in due course
+-   applemedia: add ProRes support to vtenc and vtdec
 
 Windows
 
--   this section will be filled in in due course
+-   On Windows the high-resolution clock is enabled now in the
+    gst-launch-1.0 and gst-play-1.0 command line tools, which provides
+    better clock and timer performance on Windows, at the cost of higher
+    power consumption. By default, without the high-resolution clock
+    enabled, the timer precision on Windows is system-dependent and may
+    be as bad as 15ms which is not good enough for many multimedia
+    applications. Developers may want to do the same in their Windows
+    applications if they think it’s a good idea for their application
+    use case, and depending on the Windows version they target. This is
+    not done automatically by GStreamer because on older Windows
+    versions (pre-Windows 10) this affects a global Windows setting and
+    also there’s a power consumption vs. performance trade-off that may
+    differ from application to application.
+
+-   dxgiscreencapsrc now supports resolution changes
+
+-   The wasapi2 audio plugin was rewritten and now has a higher rank
+    than the old wasapi plugin since it has a number of additional
+    features such as automatic stream routing, and no
+    known-but-hard-to-fix issues. The plugin is always built if the
+    Windows 10 SDK is available now.
+
+-   The wasapi device providers now detect and notify dynamic device
+    additions/removals
+
+-   d3d11screencapturesrc: new desktop capture element, including
+    GstDeviceProvider implementation to enumerate/select target monitors
+    for capture.
+
+-   Direct3D11/DXVA decoder now supports AV1 and MPEG2 codecs
+    (d3d11av1dec, d3d11mpeg2dec)
+
+-   VP9 decoding got more reliable and stable thanks to a newly written
+    codec parser
+
+-   Support for decoding interlaced H.264/AVC streams
+
+-   Hardware-accelerated video deinterlacing (d3d11deinterlace) and
+    video mixing (d3d11compositor)
+
+-   Video mixing with the Direct3D11 API (d3d11compositor)
+
+-   MediaFoundation API based hardware encoders gained the ability to
+    receive Direct3D11 textures as an input
+
+-   Seungha’s blog post “GStreamer ❤ Windows: A primer on the cool stuff
+    you’ll find in the 1.20 release” describes many of the
+    Windows-related improvements in more detail
 
 Linux
 
--   this section will be filled in in due course
+-   bluez: LDAC Bluetooth audio codec support in a2dpsink and avdtpsink,
+    as well as an LDAC RTP payloader (rtpldacpay) and an LDAC audio
+    encoder (ldacenc)
+
+-   kmssink: gained support for NV24, NV61, RGB16/BGR16 formats;
+    auto-detect NVIDIA Tegra driver
 
 Documentation improvements
 
--   this section will be filled in in due course
+-   hardware-accelerated GPU plugins will now no longer always list all
+    the element variants for all available GPUs, since those are
+    system-dependent and it’s confusing for users to see those in the
+    documentation just because the GStreamer developer who generated the
+    docs had multiple GPUs to play with at the time. Instead just show
+    the default elements.
 
-Possibly Breaking Changes
+Possibly Breaking and Other Noteworthy Behavioural Changes
+
+-   gst_parse_launch(), gst_parse_bin_from_description() and friends
+    will now error out when setting properties that don’t exist on
+    top-level bins. They were silently ignored before.
+
+-   The GstWebRTC library does not expose any objects anymore with
+    public fields. Instead properties have been added to replace that
+    functionality. If you are accessing such fields in your application,
+    switch to the corresponding properties.
 
--   FIXME: this section will be filled in in due course
--   MPEG-TS SCTE-35 API changes (FIXME: flesh out)
--   gst_parse_launch() and friends now error out on non-existing
-    properties on top-level bins where they would silently fail and
-    ignore those before.
 -   playbin and uridecodebin now emit the source-setup signal before the
     element is added to the bin and linked so that the source element is
     already configured before any scheduling query comes in, which is
-    useful for elements such as appsrc or giostreamsrc. (Merge Request)
+    useful for elements such as appsrc or giostreamsrc.
+
+-   The source element inside urisourcebin (used inside uridecodebin3
+    which is used inside playbin3) is no longer called "source". This
+    shouldn’t affect anyone hopefully, because there’s a "setup-source"
+    signal to configure the source element and no one should rely on
+    names of internal elements anyway.
+
+-   The vp8enc element now expects bps (bits per second) for the
+    "temporal-scalability-target-bitrate" property, which is consistent
+    with the "target-bitrate" property. Since additional configuration
+    is required with modern libvpx to make temporal scaling work anyway,
+    chances are that very few people will have been using this property
+
+-   vp8enc and vp9enc now default to “good quality” for the "deadline"
+    property rather then “best quality”. Having the deadline set to best
+    quality causes the encoder to be absurdly slow, most real-life users
+    will want the good quality tradeoff instead.
+
+-   The experimental GstTranscoder library API in gst-plugins-bad was
+    changed from a GObject signal-based notification mechanism to a
+    GstBus/message-based mechanism akin to GstPlayer/GstPlay.
+
+-   MPEG-TS SCTE-35 API: semantic change for SCTE-35 splice commands:
+    timestamps passed by the application should be in running time now,
+    since users of the API can’t really be expected to predict the local
+    PTS of the muxer.
+
+-   The GstContext used by souphttpsrc to share the session between
+    multiple element instances has changed. Previously it provided
+    direct access to the internal SoupSession object, now it only
+    provides access to an opaque, internal type. This change is
+    necessary because SoupSession is not thread-safe at all and can’t be
+    shared safely between arbitrary external code and souphttpsrc.
+
+-   Python bindings: GObject-introspection related Annotation fixes have
+    led to a case of a GstVideo.VideoInfo-related function signature
+    changing in the Python bindings (possibly one or two other cases
+    too). This is for a function that should never have been exposed in
+    the first place though, so the bindings are being updated to throw
+    an exception in that case, and the correct replacement API has been
+    added in form of an override.
 
 Known Issues
 
--   this section will be filled in in due course
-
--   There are a couple of known WebRTC-related regressions/blockers:
-
-    -   webrtc: DTLS setup with Chrome is broken
-    -   webrtcbin: First keyframe is usually lost
+-   nothing in particular at this point (but also see possibly breaking
+    changes section above)
 
 Contributors
 
--   this section will be filled in in due course
+Aaron Boxer, Adam Leppky, Adam Williamson, Alba Mendez, Alejandro
+González, Aleksandr Slobodeniuk, Alexander Vandenbulcke, Alex Ashley,
+Alicia Boya García, Andika Triwidada, Andoni Morales Alastruey, Andrew
+Wesie, Andrey Moiseev, Antonio Ospite, Antonio Rojas, Arthur Crippa
+Búrigo, Arun Raghavan, Ashley Brighthope, Axel Kellermann, Baek, Bastien
+Nocera, Bastien Reboulet, Benjamin Gaignard, Bing Song, Binh Truong,
+Biswapriyo Nath, Brad Hards, Brad Smith, Brady J. Garvin, Branko
+Subasic, Camilo Celis Guzman, Chris Bass, ChrisDuncanAnyvision, Chris
+White, Corentin Damman, Daniel Almeida, Daniel Knobe, Daniel Stone,
+david, David Fernandez, David Keijser, David Phung, Devarsh Thakkar,
+Dinesh Manajipet, Dmitry Samoylov, Dmitry Shusharin, Dominique Martinet,
+Doug Nazar, Ederson de Souza, Edward Hervey, Emmanuel Gil Peyrot,
+Enrique Ocaña González, Ezequiel Garcia, Fabian Orccon, Fabrice
+Fontaine, Fernando Jimenez Moreno, Florian Karydes, Francisco Javier
+Velázquez-García, François Laignel, Frederich Munch, Fredrik Pålsson,
+George Kiagiadakis, Georg Lippitsch, Göran Jönsson, Guido Günther,
+Guillaume Desmottes, Guiqin Zou, Haakon Sporsheim, Haelwenn (lanodan)
+Monnier, Haihao Xiang, Haihua Hu, Havard Graff, He Junyan, Helmut
+Januschka, Henry Wilkes, Hosang Lee, Hou Qi, Ignacio Casal Quinteiro,
+Igor Kovalenko, Ilya Kreymer, Imanol Fernandez, Jacek Tomaszewski, Jade
+Macho, Jakub Adam, Jakub Janků, Jan Alexander Steffens (heftig), Jan
+Schmidt, Jason Carrete, Jason Pereira, Jay Douglass, Jeongki Kim, Jérôme
+Laheurte, Jimmi Holst Christensen, Johan Sternerup, John Hassell, John
+Lindgren, John-Mark Bell, Jonathan Matthew, Jordan Petridis, Jose
+Quaresma, Julian Bouzas, Julien, Kai Uwe Broulik, Kasper Steensig
+Jensen, Kellermann Axel, Kevin Song, Khem Raj, Knut Inge Hvidsten, Knut
+Saastad, Kristofer Björkström, Lars Lundqvist, Lawrence Troup, Lim Siew
+Hoon, Lucas Stach, Ludvig Rappe, Luis Paulo Fernandes de Barros, Luke
+Yelavich, Mads Buvik Sandvei, Marc Leeman, Marco Felsch, Marek Vasut,
+Marian Cichy, Marijn Suijten, Marius Vlad, Markus Ebner, Mart Raudsepp,
+Matej Knopp, Mathieu Duponchelle, Matthew Waters, Matthieu De Beule,
+Mengkejiergeli Ba, Michael de Gans, Michael Olbrich, Michael Tretter,
+Michal Dzik, Miguel Paris, Mikhail Fludkov, mkba, Nazar Mokrynskyi,
+Nicholas Jackson, Nicola Murino, Nicolas Dufresne, Niklas Hambüchen,
+Nikolay Sivov, Nirbheek Chauhan, Olivier Blin, Olivier Crete, Olivier
+Crête, Paul Goulpié, Per Förlin, Peter Boba, P H, Philippe Normand,
+Philipp Zabel, Pieter Willem Jordaan, Piotrek Brzeziński, Rafał
+Dzięgiel, Rafostar, raghavendra, Raghavendra, Raju Babannavar, Raleigh
+Littles III, Randy Li, Randy Li (ayaka), Ratchanan Srirattanamet, Raul
+Tambre, reed.lawrence, Ricky Tang, Robert Rosengren, Robert Swain, Robin
+Burchell, Roman Sivriver, R S Nikhil Krishna, Ruben Gonzalez, Ruslan
+Khamidullin, Sanchayan Maity, Scott Moreau, Sebastian Dröge, Sergei
+Kovalev, Seungha Yang, Sid Sethupathi, sohwan.park, Sonny Piers, Staz M,
+Stefan Brüns, Stéphane Cerveau, Stephan Hesse, Stian Selnes, Stirling
+Westrup, Théo MAILLART, Thibault Saunier, Tim, Timo Wischer, Tim-Philipp
+Müller, Tim Schneider, Tobias Ronge, Tom Schoonjans, Tulio Beloqui,
+tyler-aicradle, U. Artie Eoff, Ung, Val Doroshchuk, VaL Doroshchuk,
+Víctor Manuel Jáquez Leal, Vivek R, Vivia Nikolaidou, Vivienne
+Watermeier, Vladimir Menshakov, Will Miller, Wim Taymans, Xabier
+Rodriguez Calvar, Xavier Claessens, Xℹ Ruoyao, Yacine Bandou, Yinhang
+Liu, youngh.lee, youngsoo.lee, yychao, Zebediah Figura, Zhang yuankun,
+Zhang Yuankun, Zhao, Zhao Zhili, , Aleksandar Topic, Antonio Ospite,
+Bastien Nocera, Benjamin Gaignard, Brad Hards, Carlos Falgueras García,
+Célestin Marot, Corentin Damman, Corentin Noël, Daniel Almeida, Daniel
+Knobe, Danny Smith, Dave Piché, Dmitry Osipenko, Fabrice Fontaine,
+fjmax, Florian Zwoch, Guillaume Desmottes, Haihua Hu, Heinrich Kruger,
+He Junyan, Jakub Adam, James Cowgill, Jan Alexander Steffens (heftig),
+Jean Felder, Jeongki Kim, Jiri Uncovsky, Joe Todd, Jordan Petridis,
+Krystian Wojtas, Marc-André Lureau, Marcin Kolny, Marc Leeman, Mark
+Nauwelaerts, Martin Reboredo, Mathieu Duponchelle, Matthew Waters,
+Mengkejiergeli Ba, Michael Gruner, Nicolas Dufresne, Nirbheek Chauhan,
+Olivier Crête, Philippe Normand, Rafał Dzięgiel, Ralf Sippl, Robert
+Mader, Sanchayan Maity, Sangchul Lee, Sebastian Dröge, Seungha Yang,
+Stéphane Cerveau, Teh Yule Kim, Thibault Saunier, Thomas Klausner, Timo
+Wischer, Tim-Philipp Müller, Tobias Reineke, Tomasz Andrzejak, Trung Do,
+Tyler Compton, Ung, Víctor Manuel Jáquez Leal, Vivia Nikolaidou, Wim
+Taymans, wngecn, Wonchul Lee, wuchang li, Xavier Claessens, Xi Ruoyao,
+Yoshiharu Hirose, Zhao,
 
 … and many others who have contributed bug reports, translations, sent
 suggestions or helped testing.
@@ -307,7 +1863,7 @@ the git 1.20 branch, which will be a stable branch.
 
 1.20.0
 
-1.20.0 is scheduled to be released around October/November 2021.
+1.20.0 is scheduled to be released around early February 2022.
 
 Schedule for 1.22
 
@@ -315,7 +1871,9 @@ Our next major feature release will be 1.22, and 1.21 will be the
 unstable development version leading up to the stable 1.22 release. The
 development of 1.21/1.22 will happen in the git main branch.
 
-The plan for the 1.22 development cycle is yet to be confirmed.
+The plan for the 1.22 development cycle is yet to be confirmed. Assuming
+no major project-wide reorganisations in the 1.22 cycle we might try and
+aim for a release around August 2022.
 
 1.22 will be backwards-compatible to the stable 1.20, 1.18, 1.16, 1.14,
 1.12, 1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series.
@@ -323,6 +1881,7 @@ The plan for the 1.22 development cycle is yet to be confirmed.
 ------------------------------------------------------------------------
 
 These release notes have been prepared by Tim-Philipp Müller with
-contributions from …
+contributions from Matthew Waters, Nicolas Dufresne, Nirbheek Chauhan,
+Sebastian Dröge and Seungha Yang.
 
 License: CC BY-SA 4.0
diff --git a/subprojects/gst-rtsp-server/RELEASE b/subprojects/gst-rtsp-server/RELEASE
index 229d2c5554..b7f80bf179 100644
--- a/subprojects/gst-rtsp-server/RELEASE
+++ b/subprojects/gst-rtsp-server/RELEASE
@@ -1,4 +1,4 @@
-This is GStreamer gst-rtsp-server 1.19.3.
+This is GStreamer gst-rtsp-server 1.19.90.
 
 GStreamer 1.19 is the development branch leading up to the next major
 stable version which will be 1.20.
diff --git a/subprojects/gst-rtsp-server/docs/gst_plugins_cache.json b/subprojects/gst-rtsp-server/docs/gst_plugins_cache.json
index 060da9e54d..509ca4e683 100644
--- a/subprojects/gst-rtsp-server/docs/gst_plugins_cache.json
+++ b/subprojects/gst-rtsp-server/docs/gst_plugins_cache.json
@@ -321,7 +321,7 @@
                         "construct": false,
                         "construct-only": false,
                         "controllable": false,
-                        "default": "GStreamer/1.19.3.1",
+                        "default": "GStreamer/1.19.90",
                         "mutable": "null",
                         "readable": true,
                         "type": "gchararray",
diff --git a/subprojects/gst-rtsp-server/gst-rtsp-server.doap b/subprojects/gst-rtsp-server/gst-rtsp-server.doap
index fac8f83316..7a097e0563 100644
--- a/subprojects/gst-rtsp-server/gst-rtsp-server.doap
+++ b/subprojects/gst-rtsp-server/gst-rtsp-server.doap
@@ -32,6 +32,16 @@ RTSP server library based on GStreamer
 
  <release>
   <Version>
+   <revision>1.19.90</revision>
+   <branch>main</branch>
+   <name></name>
+   <created>2022-01-28</created>
+   <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.19.90.tar.xz" />
+  </Version>
+ </release>
+
+ <release>
+  <Version>
    <revision>1.19.3</revision>
    <branch>main</branch>
    <name></name>
diff --git a/subprojects/gst-rtsp-server/meson.build b/subprojects/gst-rtsp-server/meson.build
index 61bbae83b9..834f20d1f2 100644
--- a/subprojects/gst-rtsp-server/meson.build
+++ b/subprojects/gst-rtsp-server/meson.build
@@ -1,5 +1,5 @@
 project('gst-rtsp-server', 'c',
-  version : '1.19.3.1',
+  version : '1.19.90',
   meson_version : '>= 0.59',
   default_options : ['warning_level=1', 'buildtype=debugoptimized'])
 
diff --git a/subprojects/gstreamer-sharp/meson.build b/subprojects/gstreamer-sharp/meson.build
index 55e2693327..b48ea71548 100644
--- a/subprojects/gstreamer-sharp/meson.build
+++ b/subprojects/gstreamer-sharp/meson.build
@@ -1,4 +1,4 @@
-project('gstreamer-sharp', ['cs', 'c'], version: '1.19.3.1',
+project('gstreamer-sharp', ['cs', 'c'], version: '1.19.90',
     meson_version : '>= 0.59', license: 'LGPL')
 
 if host_machine.system() == 'osx'
diff --git a/subprojects/gstreamer-sharp/sources/generated/Gst.PbUtils/Constants.cs b/subprojects/gstreamer-sharp/sources/generated/Gst.PbUtils/Constants.cs
index 0e2455bb4a..0fe1bf7176 100644
--- a/subprojects/gstreamer-sharp/sources/generated/Gst.PbUtils/Constants.cs
+++ b/subprojects/gstreamer-sharp/sources/generated/Gst.PbUtils/Constants.cs
@@ -17,9 +17,9 @@ namespace Gst.PbUtils {
 		public const string ENCODING_CATEGORY_ONLINE_SERVICE = @"online-service";
 		public const string ENCODING_CATEGORY_STORAGE_EDITING = @"storage-editing";
 		public const int PLUGINS_BASE_VERSION_MAJOR = 1;
-		public const int PLUGINS_BASE_VERSION_MICRO = 3;
+		public const int PLUGINS_BASE_VERSION_MICRO = 90;
 		public const int PLUGINS_BASE_VERSION_MINOR = 19;
-		public const int PLUGINS_BASE_VERSION_NANO = 1;
+		public const int PLUGINS_BASE_VERSION_NANO = 0;
 #endregion
 	}
 }
diff --git a/subprojects/gstreamer-sharp/sources/generated/Gst/Constants.cs b/subprojects/gstreamer-sharp/sources/generated/Gst/Constants.cs
index 2dc1302a25..b63105a303 100644
--- a/subprojects/gstreamer-sharp/sources/generated/Gst/Constants.cs
+++ b/subprojects/gstreamer-sharp/sources/generated/Gst/Constants.cs
@@ -170,9 +170,9 @@ namespace Gst {
 		public const int VALUE_LESS_THAN = -1;
 		public const int VALUE_UNORDERED = 2;
 		public const int VERSION_MAJOR = 1;
-		public const int VERSION_MICRO = 3;
+		public const int VERSION_MICRO = 90;
 		public const int VERSION_MINOR = 19;
-		public const int VERSION_NANO = 1;
+		public const int VERSION_NANO = 0;
 #endregion
 	}
 }
diff --git a/subprojects/gstreamer-sharp/sources/generated/gstreamer-sharp-api.xml b/subprojects/gstreamer-sharp/sources/generated/gstreamer-sharp-api.xml
index 30df232299..3e879bba1d 100644
--- a/subprojects/gstreamer-sharp/sources/generated/gstreamer-sharp-api.xml
+++ b/subprojects/gstreamer-sharp/sources/generated/gstreamer-sharp-api.xml
@@ -12127,10 +12127,10 @@
       <constant value="1" ctype="gint" gtype="gint" name="VALUE_GREATER_THAN" />
       <constant value="-1" ctype="gint" gtype="gint" name="VALUE_LESS_THAN" />
       <constant value="2" ctype="gint" gtype="gint" name="VALUE_UNORDERED" />
-      <constant value="1" ctype="gint" gtype="gint"   name="VERSION_MAJOR" />
-      <constant value="3" ctype="gint" gtype="gint"   name="VERSION_MICRO" />
-      <constant value="19" ctype="gint" gtype="gint"   name="VERSION_MINOR" />
-      <constant value="1" ctype="gint" gtype="gint"   name="VERSION_NANO" />
+      <constant value="1" ctype="gint" gtype="gint"    name="VERSION_MAJOR" />
+      <constant value="90" ctype="gint" gtype="gint"    name="VERSION_MICRO" />
+      <constant value="19" ctype="gint" gtype="gint"    name="VERSION_MINOR" />
+      <constant value="0" ctype="gint" gtype="gint"    name="VERSION_NANO" />
     </object>
     <class name="Parse" cname="GstParse" disable_void_ctor="1">
       <method name="ParseBinFromDescription" cname="gst_parse_bin_from_description" shared="true">
@@ -21582,10 +21582,10 @@
       <constant value="file-extension" ctype="gchar*" gtype="gchar*" name="ENCODING_CATEGORY_FILE_EXTENSION" />
       <constant value="online-service" ctype="gchar*" gtype="gchar*" name="ENCODING_CATEGORY_ONLINE_SERVICE" />
       <constant value="storage-editing" ctype="gchar*" gtype="gchar*" name="ENCODING_CATEGORY_STORAGE_EDITING" />
-      <constant value="1" ctype="gint" gtype="gint"   name="PLUGINS_BASE_VERSION_MAJOR" />
-      <constant value="3" ctype="gint" gtype="gint"   name="PLUGINS_BASE_VERSION_MICRO" />
-      <constant value="19" ctype="gint" gtype="gint"   name="PLUGINS_BASE_VERSION_MINOR" />
-      <constant value="1" ctype="gint" gtype="gint"   name="PLUGINS_BASE_VERSION_NANO" />
+      <constant value="1" ctype="gint" gtype="gint"    name="PLUGINS_BASE_VERSION_MAJOR" />
+      <constant value="90" ctype="gint" gtype="gint"    name="PLUGINS_BASE_VERSION_MICRO" />
+      <constant value="19" ctype="gint" gtype="gint"    name="PLUGINS_BASE_VERSION_MINOR" />
+      <constant value="0" ctype="gint" gtype="gint"    name="PLUGINS_BASE_VERSION_NANO" />
     </object>
   </namespace>
   <namespace name="Gst.Rtp" library="gstrtp-1.0-0.dll">
diff --git a/subprojects/gstreamer-vaapi/ChangeLog b/subprojects/gstreamer-vaapi/ChangeLog
index a3e382fc2f..5a994594fc 100644
--- a/subprojects/gstreamer-vaapi/ChangeLog
+++ b/subprojects/gstreamer-vaapi/ChangeLog
@@ -1,3 +1,18 @@
+=== release 1.19.90 ===
+
+2022-01-28 14:28:35 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* NEWS:
+	* RELEASE:
+	* gstreamer-vaapi.doap:
+	* meson.build:
+	  Release 1.19.90
+
+2022-01-28 14:28:28 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* ChangeLog:
+	  Update ChangeLogs for 1.19.90
+
 2022-01-05 02:07:59 +0530  Nirbheek Chauhan <nirbheek@centricular.com>
 
 	* docs/meson.build:
diff --git a/subprojects/gstreamer-vaapi/NEWS b/subprojects/gstreamer-vaapi/NEWS
index e9f5227c15..1a512c1848 100644
--- a/subprojects/gstreamer-vaapi/NEWS
+++ b/subprojects/gstreamer-vaapi/NEWS
@@ -1,23 +1,19 @@
 GStreamer 1.20 Release Notes
 
 GStreamer 1.20 has not been released yet. It is scheduled for release in
-November 2021.
+late January / early February 2022.
 
 1.19.x is the unstable development version that is being developed in
-the git main branch and which will eventually result in 1.20, and 1.19.3
-is the current development release in that series
-
-It is expected that feature freeze will be in early October 2021,
-followed by one or two 1.19.9x pre-releases and the new 1.20 stable
-release around October/November 2021.
+the git main branch and which will eventually result in 1.20, and
+1.19.90 is the first release candidate in that series (1.20rc1).
 
 1.20 will be backwards-compatible to the stable 1.18, 1.16, 1.14, 1.12,
-1.10, 1.8, 1.6,, 1.4, 1.2 and 1.0 release series.
+1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series.
 
 See https://gstreamer.freedesktop.org/releases/1.20/ for the latest
 version of this document.
 
-Last updated: Monday 1 November 2021, 01:00 UTC (log)
+Last updated: Wednesday 26 January 2022, 01:00 UTC (log)
 
 Introduction
 
@@ -30,25 +26,971 @@ fixes and other improvements.
 
 Highlights
 
--   this section will be completed in due course
+-   Development in GitLab was switched to a single git repository
+    containing all the modules
+-   GstPlay: new high-level playback library, replaces GstPlayer
+-   WebM Alpha decoding support
+-   Encoding profiles can now be tweaked with additional
+    application-specified element properties
+-   Compositor: multi-threaded video conversion and mixing
+-   RTP header extensions: unified support in RTP depayloader and
+    payloader base classes
+-   SMPTE 2022-1 2-D Forward Error Correction support
+-   Smart encoding (passthrough) support for VP8, VP9, H.265 in
+    encodebin and transcodebin
+-   Runtime compatibility support for libsoup2 and libsoup3 (libsoup3
+    support experimental)
+-   Video decoder subframe support
+-   Video decoder automatic packet-loss, data corruption, and keyframe
+    request handling for RTP / WebRTC / RTSP
+-   MP4 and Matroska muxers now support profile/level/resolution changes
+    for H264/H265 input streams (i.e. codec data changing on the fly)
+-   MP4 muxing mode that initially creates a fragmented mp4 which is
+    converted to a regular mp4 on EOS
+-   Audio support for the WebKit Port for Embedded (WPE) web page source
+    element
+-   CUDA based video color space convert and rescale elements and
+    upload/download elements
+-   NVIDIA memory:NVMM support for OpenGL glupload and gldownload
+    elements
+-   Many WebRTC improvements
+-   The new VA-API plugin implemention fleshed out with more decoders
+    and new postproc elements
+-   AppSink API to retrieve events in addition to buffers and buffer
+    lists
+-   AppSrc gained more configuration options for the internal queue
+    (leakiness, limits in buffers and time, getters to read current
+    levels)
+-   Updated Rust bindings and many new Rust plugins
+-   Improved support for custom minimal GStreamer builds
+-   Support build against FFmpeg 5.0
+-   Linux Stateless CODEC support gained MPEG2 and VP9
+-   Windows Direct3D11/DXVA decoder gained AV1 and MPEG2 support
+-   Lots of new plugins, features, performance improvements and bug
+    fixes
 
 Major new features and changes
 
 Noteworthy new features and API
 
--   this section will be filled in in due course
+-   gst_element_get_request_pad() has been deprecated in favour of the
+    newly-added gst_element_request_pad_simple() which does the exact
+    same thing but has a less confusing name that hopefully makes clear
+    that the function request a new pad rather than just retrieves an
+    already-existing request pad.
+
+Development in GitLab was switched to a single git repository containing all the modules
+
+The GStreamer multimedia framework is a set of libraries and plugins
+split into a number of distinct modules which are released independently
+and which have so far been developed in separate git repositories in
+freedesktop.org GitLab.
+
+In addition to these separate git repositories there was a gst-build
+module that would use the Meson build systems’s subproject feature to
+download each individual module and then build everything in one go. It
+would also provide an uninstalled development environment that made it
+easy to work on GStreamer and use or test versions other than the
+system-installed GStreamer version.
+
+All of these modules have now (as of 28 September 2021) been merged into
+a single git repository (“Mono repository” or “monorepo”) which should
+simplify development workflows and continuous integration, especially
+where changes need to be made to multiple modules at once.
+
+This mono repository merge will primarily affect GStreamer developers
+and contributors and anyone who has workflows based on the GStreamer git
+repositories.
+
+The Rust bindings and Rust plugins modules have not been merged into the
+mono repository at this time because they follow a different release
+cycle.
+
+The mono repository lives in the existing GStreamer core git repository
+in GitLab in the new main branch and all future development will happen
+on this branch.
+
+Modules will continue to be released as separate tarballs.
+
+For more details, please see the GStreamer mono repository FAQ.
+
+GstPlay: new high-level playback library replacing GstPlayer
+
+-   GstPlay is a new high-level playback library that replaces the older
+    GstPlayer API. It is basically the same API as GstPlayer but
+    refactored to use bus messages for application notifications instead
+    of GObject signals. There is still a signal adapter object for those
+    who prefer signals. Since the existing GstPlayer API is already in
+    use in various applications, it didn’t seem like a good idea to
+    break it entirely. Instead a new API was added, and it is expected
+    that this new GstPlay API will be moved to gst-plugins-base in
+    future.
+
+-   The existing GstPlayer API is scheduled for deprecation and will be
+    removed at some point in the future (e.g. in GStreamer 1.24), so
+    application developers are urged to migrate to the new GstPlay API
+    at their earliest convenience.
+
+WebM alpha decoding
+
+-   Implement WebM alpha decoding (VP8/VP9 with alpha), which required
+    support and additions in various places. This is supported both with
+    software decoders and hardware-accelerated decoders.
+
+-   VP8/VP9 don’t support alpha components natively in the codec, so the
+    way this is implemented in WebM is by encoding the alpha plane with
+    transparency data as a separate VP8/VP9 stream. Inside the WebM
+    container (a variant of Matroska) this is coded as a single video
+    track with the “normal” VP8/VP9 video data making up the main video
+    data and each frame of video having an encoded alpha frame attached
+    to it as extra data ("BlockAdditional").
+
+-   matroskademux has been extended extract this per-frame alpha side
+    data and attach it in form of a GstVideoCodecAlphaMeta to the
+    regular video buffers. Note that this new meta is specific to this
+    VP8/VP9 alpha support and can’t be used to just add alpha support to
+    other codecs that don’t support it. Lastly, matroskademux also
+    advertises the fact that the streams contain alpha in the caps.
+
+-   The new codecalpha plugin contains various bits of infrastructure to
+    support autoplugging and debugging:
+
+    -   codecalphademux splits out the alpha stream from the metas on
+        the regular VP8/VP9 buffers
+    -   alphacombine takes two decoded raw video streams (one alpha, one
+        the regular video) and combines it into a video stream with
+        alpha
+    -   vp8alphadecodebin + vp9alphadecodebin are wrapper bins that use
+        the regular vp8dec and vp9dec software decoders to decode
+        regular and alpha streams and combine them again. To decodebin
+        these look like regular decoders which ju
+    -   The V4L2 CODEC plugin has stateless VP8/VP9 decoders that can
+        decode both alpha and non-alpha stream with a single decoder
+        instance
+
+-   A new AV12 video format was added which is basically NV12 with an
+    alpha plane, which is more convenient for many hardware-accelerated
+    decoders.
+
+-   Watch Nicolas Dufresne’s LCA 2022 talk “Bringing WebM Alpha support
+    to GStreamer” for all the details and a demo.
+
+RTP Header Extensions Base Class and Automatic Header Extension Handling in RTP Payloaders and Depayloaders
+
+-   RTP Header Extensions are specified in RFC 5285 and provide a way to
+    add small pieces of data to RTP packets in between the RTP header
+    and the RTP payload. This is often used for per-frame metadata,
+    extended timestamps or other application-specific extra data. There
+    are several commonly-used extensions specified in various RFCs, but
+    senders are free to put any kind of data in there, as long as sender
+    and receiver both know what that data is. Receivers that don’t know
+    about the header extensions will just skip the extra data without
+    ever looking at it. These header extensions can often be combined
+    with any kind of payload format, so may need to be supported by many
+    RTP payloader and depayloader elements.
+
+-   Inserting and extracting RTP header extension data has so far been a
+    bit inconvenient in GStreamer: There are functions to add and
+    retrieve RTP header extension data from RTP packets, but nothing
+    works automatically, even for common extensions. People would have
+    to do the insertion/extraction either in custom elements
+    before/after the RTP payloader/depayloader, or inside pad probes,
+    which isn’t very nice.
+
+-   This release adds various pieces of new infrastructure for generic
+    RTP header extension handling, as well as some implementations for
+    common extensions:
+
+    -   GstRTPHeaderExtension is a new helper base class for reading and
+        writing RTP header extensions. Nominally this subclasses
+        GstElement, but only so these extensions are stored in the
+        registry where they can be looked up by URI or name. They don’t
+        have pads and don’t get added to the pipeline graph as an
+        element.
+
+    -   "add-extension" and "clear-extension" action signals on RTP
+        payloaders and depayloaders for manual extension management
+
+    -   The "request-extension" signal will be emitted if an extension
+        is encountered that requires explicit mapping by the application
+
+    -   new "auto-header-extension" property on RTP payloaders and
+        depayloaders for automatic handling of known header extensions.
+        This is enabled by default. The extensions must be signalled via
+        caps / SDP.
+
+    -   RTP header extension implementations:
+
+        -   rtphdrextclientaudiolevel: Client-to-Mixer Audio Level
+            Indication (RFC 6464) (also see below)
+        -   rtphdrextcolorspace: Color Space extension, extends RTP
+            packets with color space and high dynamic range (HDR)
+            information
+        -   rtphdrexttwcc: Transport Wide Congestion Control support
+
+-   gst_rtp_buffer_remove_extension_data() is a new helper function to
+    remove an RTP header extension from an RTP buffer
+
+-   The existing gst_rtp_buffer_set_extension_data() now also supports
+    shrinking the extension data in size
+
+AppSink and AppSrc improvements
+
+-   appsink: new API to pull events out of appsink in addition to
+    buffers and buffer lists.
+
+    There was previously no way for users to receive incoming events
+    from appsink properly serialised with the data flow, even if they
+    are serialised events. The reason for that is that the only way to
+    intercept events was via a pad probe on the appsink sink pad, but
+    there is also internal queuing inside of appsink, so it’s difficult
+    to ascertain the right order of everything in all cases.
+
+    There is now a new "new-serialized-event" signal which will be
+    emitted when there’s a new event pending (just like the existing
+    "new-sample" signal). The "emit-signals" property must be set to
+    TRUE in order to activate this (but it’s also fine to just pull from
+    the application thread without using the signals).
+
+    gst_app_sink_pull_object() and gst_app_sink_try_pull_object() can be
+    used to pull out either an event or a new sample carrying a buffer
+    or buffer list, whatever is next in the queue.
+
+    EOS events will be filtered and will not be returned. EOS handling
+    can be done the ususal way, same as with _pull_sample().
+
+-   appsrc: allow configuration of internal queue limits in time and
+    buffers and add leaky mode.
+
+    There is internal queuing inside appsrc so the application thread
+    can push data into the element which will then be picked up by the
+    source element’s streaming thread and pushed into the pipeline from
+    that streaming thread. This queue is unlimited by default and until
+    now it was only possible to set a maximum size limit in bytes. When
+    that byte limit is reached, the pushing thread (application thread)
+    would be blocked until more space becomes available.
+
+    A limit in bytes is not particularly useful for many use cases, so
+    now it is possible to also configure limits in time and buffers
+    using the new "max-time" and "max-buffers" properties. Of course
+    there are also matching new read-only"current-level-buffers" and
+    "current-level-time properties" properties to query the current fill
+    level of the internal queue in time and buffers.
+
+    And as if that wasn’t enough the internal queue can also be
+    configured as leaky using the new "leaky-type" property. That way
+    when the queue is full the application thread won’t be blocked when
+    it tries to push in more data, but instead either the new buffer
+    will be dropped or the oldest data in the queue will be dropped.
+
+Better string serialization of nested GstCaps and GstStructures
+
+-   New string serialisation format for structs and caps that can handle
+    nested structs and caps properly by using brackets to delimit nested
+    items (e.g. some-struct, some-field=[nested-struct, nested=true]).
+    Unlike the default format the new variant can also support more than
+    one level of nesting. For backwards-compatibility reasons the old
+    format is still output by default when serialising caps and structs
+    using the existing API. The new functions gst_caps_serialize() and
+    gst_structure_serialize() can be used to output strings in the new
+    format.
+
+Convenience API for custom GstMetas
+
+-   New convenience API to register and create custom GstMetas:
+    gst_meta_register_custom() and gst_buffer_add_custom_meta(). Such
+    custom meta is backed by a GstStructure and does not require that
+    users of the API expose their GstMeta implementation as public API
+    for other components to make use of it. In addition, it provides a
+    simpler interface by ignoring the impl vs. api distinction that the
+    regular API exposes. This new API is meant to be the meta
+    counterpart to custom events and messages, and to be more convenient
+    than the lower-level API when the absolute best performance isn’t a
+    requirement. The reason it’s less performant than a “proper” meta is
+    that a proper meta is just a C struct in the end whereas this goes
+    through the GstStructure API which has a bit more overhead, which
+    for most scenarios is negligible however. This new API is useful for
+    experimentation or proprietary metas, but also has some limitations:
+    it can only be used if there’s a single producer of these metas;
+    it’s not allowed to register the same custom meta multiple times or
+    from multiple places.
+
+Additional Element Properties on Encoding Profiles
+
+-   GstEncodingProfile: The new "element-properties" and
+    gst_encoding_profile_set_element_properties() API allows
+    applications to set additional element properties on encoding
+    profiles to configure muxers and encoders. So far the encoding
+    profile template was the only place where this could be specified,
+    but often what applications want to do is take a ready-made encoding
+    profile shipped by GStreamer or the application and then tweak the
+    settings on top of that, which is now possible with this API. Since
+    applications can’t always know in advance what encoder element will
+    be used in the end, it’s even possible to specify properties on a
+    per-element basis.
+
+    Encoding Profiles are used in the encodebin, transcodebin and
+    camerabin elements and APIs to configure output formats (containers
+    and elementary streams).
+
+Audio Level Indication Meta for RFC 6464
+
+-   New GstAudioLevelMeta containing Audio Level Indication as per RFC
+    6464
+
+-   The level element has been updated to add GstAudioLevelMeta on
+    buffers if the "audio-level-meta" property is set to TRUE. This can
+    then in turn be picked up by RTP payloaders to signal the audio
+    level to receivers through RTP header extensions (see above).
+
+-   New Client-to-Mixer Audio Level Indication (RFC6464) RTP Header
+    Extension which should be automatically created and used by RTP
+    payloaders and depayloaders if their "auto-header-extension"
+    property is enabled and if the extension is part of the RTP caps.
+
+Automatic packet loss, data corruption and keyframe request handling for video decoders
+
+-   The GstVideoDecoder base class has gained various new APIs to
+    automatically handle packet loss and data corruption better by
+    default, especially in RTP, RTSP and WebRTC streaming scenarios, and
+    to give subclasses more control about how they want to handle
+    missing data:
+
+    -   Video decoder subclasses can mark output frames as corrupted via
+        the new GST_VIDEO_CODEC_FRAME_FLAG_CORRUPTED flag
+
+    -   A new "discard-corrupted-frames" property allows applications to
+        configure decoders so that corrupted frames are directly
+        discarded instead of being forwarded inside the pipeline. This
+        is a replacement for the "output-corrupt" property of the FFmpeg
+        decoders.
+
+    -   RTP depayloaders can now signal to decoders that data is missing
+        when sending GAP events for lost packets. GAP events can be sent
+        for various reason in a GStreamer pipeline. Often they are just
+        used to let downstream elements know that there isn’t a buffer
+        available at the moment, so downstream elements can move on
+        instead of waiting for one. They are also sent by RTP
+        depayloaders in the case that packets are missing, however, and
+        so far a decoder was not able to differentiate the two cases.
+        This has been remedied now: GAP events can be decorated with
+        gst_event_set_gap_flags() and GST_GAP_FLAG_MISSING_DATA to let
+        decoders now what happened, and decoders can then use that in
+        some cases to handle missing data better.
+
+    -   The GstVideoDecoder::handle_missing_data vfunc was added to
+        inform subclasses about packet loss or missing data and let them
+        handle it in their own way if they like.
+
+    -   gst_video_decoder_set_needs_sync_point() lets subclasses signal
+        that they need the stream to start with a sync point. If
+        enabled, the base class will discard all non-sync point frames
+        in the beginning and after a flush and does not pass them to the
+        subclass. Furthermore, if the first frame is not a sync point,
+        the base class will try and request a sync frame from upstream
+        by sending a force-key-unit event (see next items).
+
+    -   New "automatic-request-sync-points" and
+        "automatic-request-sync-point-flags" properties to automatically
+        request sync points when needed, e.g. on packet loss or if the
+        first frame is not a keyframe. Applications may want to enable
+        this on decoders operating in e.g. RTP/WebRTC/RTSP receiver
+        pipelines.
+
+    -   The new "min-force-key-unit-interval" property can be used to
+        ensure there’s a minimal interval between keyframe requests to
+        upstream (and/or the sender) and we’re not flooding the sender
+        with key unit requests.
+
+    -   gst_video_decoder_request_sync_point() allows subclasses to
+        request a new sync point (e.g. if they choose to do their own
+        missing data handling). This will still honour the
+        "min-force-key-unit-interval" property if set.
+
+Improved support for custom minimal GStreamer builds
+
+-   Element registration and registration of other plugin features
+    inside plugin init functions has been improved in order to
+    facilitate minimal custom GStreamer builds.
+
+-   A number of new macros have been added to declare and create
+    per-element and per-pluginfeature register functions in all plugins,
+    and then call those from the per-plugin plugin_init functions:
+
+    -   GST_ELEMENT_REGISTER_DEFINE,
+        GST_DEVICE_PROVIDER_REGISTER_DEFINE,
+        GST_DYNAMIC_TYPE_REGISTER_DEFINE, GST_TYPE_FIND_REGISTER_DEFINE
+        for the actual registration call with GStreamer
+    -   GST_ELEMENT_REGISTER, GST_DEVICE_PROVIDER_REGISTER,
+        GST_DYNAMIC_TYPE_REGISTER, GST_PLUGIN_STATIC_REGISTER,
+        GST_TYPE_FIND_REGISTER to call the registration function defined
+        by the REGISTER_DEFINE macro
+    -   GST_ELEMENT_REGISTER_DECLARE,
+        GST_DEVICE_PROVIDER_REGISTER_DECLARE,
+        GST_DYNAMIC_TYPE_REGISTER_DECLARE,
+        GST_TYPE_FIND_REGISTER_DECLARE to declare the registration
+        function defined by the REGISTER_DEFINE macro
+    -   and various variants for advanced use cases.
+
+-   This means that applications can call the per-element and
+    per-pluginfeature registration functions for only the elements they
+    need instead of registering plugins as a whole with all kinds of
+    elements that may not be required (e.g. encoder and decoder instead
+    of just decoder). In case of static linking all unused functions and
+    their dependencies would be removed in this case by the linker,
+    which helps minimise binary size for custom builds.
+
+-   gst_init() will automatically call a gst_init_static_plugins()
+    function if one exists.
+
+-   See the GStreamer static build documentation and Stéphane’s blog
+    post Generate a minimal GStreamer build, tailored to your needs for
+    more details.
 
 New elements
 
--   this section will be filled in in due course
+-   New aesdec and aesenc elements for AES encryption and decryption in
+    a custom format.
+
+-   New encodebin2 element with dynamic/sometimes source pads in order
+    to support the option of doing the muxing outside of encodebin,
+    e.g. in combination with a splitmuxsink.
+
+-   New fakeaudiosink and videocodectestsink elements for testing and
+    debugging (see below for more details)
+
+-   rtpisacpay, rtpisacdepay: new RTP payloader and depayloader for iSAC
+    audio codec
+
+-   rtpst2022-1-fecdec, rtpst2022-1-fecenc: new elements providing SMPTE
+    2022-1 2-D Forward Error Correction. More details in Mathieu’s blog
+    post.
+
+-   isac: new plugin wrapping the Internet Speech Audio Codec reference
+    encoder and decoder from the WebRTC project.
+
+-   asio: plugin for Steinberg ASIO (Audio Streaming Input/Output) API
+
+-   gssrc, gssink: add source and sink for Google Cloud Storage
+
+-   onnx: new plugin to apply ONNX neural network models to video
+
+-   openaptx: aptX and aptX-HD codecs using libopenaptx (v0.2.0)
+
+-   qroverlay, debugqroverlay: new elements that allows overlaying data
+    on top of video in form of a QR code
+
+-   cvtracker: new OpenCV-based tracker element
+
+-   av1parse, vp9parse: new parsers for AV1 and VP9 video
+
+-   va: work on the new VA-API plugin implementation for
+    hardware-accelerated video decoding and encoding has continued at
+    pace, with various new decoders and filters having joined the
+    initial vah264dec:
+
+    -   vah265dec: VA-API H.265 decoder
+    -   vavp8dec: VA-API VP8 decoder
+    -   vavp9dec: VA-API VP9 decoder
+    -   vaav1dec: VA-API AV1 decoder
+    -   vampeg2dec: VA-API MPEG-2 decoder
+    -   vadeinterlace: : VA-API deinterlace filter
+    -   vapostproc: : VA-API postproc filter (color conversion,
+        resizing, cropping, color balance, video rotation, skin tone
+        enhancement, denoise, sharpen)
+
+    See Víctor’s blog post “GstVA in GStreamer 1.20” for more details
+    and what’s coming up next.
+
+-   vaapiav1dec: new AV1 decoder element (in gstreamer-vaapi)
+
+-   msdkav1dec: hardware-accelerated AV1 decoder using the Intel Media
+    SDK / oneVPL
+
+-   nvcodec plugin for NVIDIA NVCODEC API for hardware-accelerated video
+    encoding and decoding:
+
+    -   cudaconvert, cudascale: new CUDA based video color space convert
+        and rescale elements
+    -   cudaupload, cudadownload: new helper elements for memory
+        transfer between CUDA and system memory spaces
+    -   nvvp8sldec, nvvp9sldec: new GstCodecs-based VP8/VP9 decoders
+
+-   Various new hardware-accelerated elements for Windows:
+
+    -   d3d11screencapturesrc: new desktop capture element, including a
+        GstDeviceProvider implementation to enumerate/select target
+        monitors for capture.
+    -   d3d11av1dec and d3d11mpeg2dec: AV1 and MPEG-2 decoders
+    -   d3d11deinterlace: deinterlacing filter
+    -   d3d11compositor: video composing element
+    -   see Windows section below for more details
+
+-   new Rust plugins:
+
+    -   audiornnoise: Removes noise from an audio stream
+    -   awstranscribeparse: Parses AWS audio transcripts into timed text
+        buffers
+    -   ccdetect: Detects if valid closed captions are present in a
+        closed captions stream
+    -   cea608tojson: Converts CEA-608 Closed Captions to a JSON
+        representation
+    -   cmafmux: CMAF fragmented MP4 muxer
+    -   dashmp4mux: DASH fragmented MP4 muxer
+    -   isofmp4mux: ISO fragmented MP4 muxer
+    -   ebur128level: EBU R128 Loudness Level Measurement
+    -   ffv1dec: FFV1 video decoder
+    -   gtk4paintablesink: GTK4 video sink, which provides a
+        GdkPaintable that can be rendered in various widgets
+    -   hlssink3: HTTP Live Streaming sink
+    -   hrtfrender: Head-Related Transfer Function (HRTF) renderer
+    -   hsvdetector: HSV colorspace detector
+    -   hsvfilter: HSV colorspace filter
+    -   jsongstenc: Wraps buffers containing any valid top-level JSON
+        structures into higher level JSON objects, and outputs those as
+        ndjson
+    -   jsongstparse: Parses ndjson as output by jsongstenc
+    -   jsontovtt: converts JSON to WebVTT subtitles
+    -   regex: Applies regular expression operations on text
+    -   roundedcorners: Adds rounded corners to video
+    -   spotifyaudiosrc: Spotify source
+    -   textahead: Display upcoming text buffers ahead (e.g. for
+        Karaoke)
+    -   transcriberbin: passthrough bin that transcribes raw audio to
+        closed captions using awstranscriber and puts the captions as
+        metas onto the video
+    -   tttojson: Converts timed text to a JSON representation
+    -   uriplaylistbin: Playlist source bin
+    -   webpdec-rs: WebP image decoder with animation support
+
+-   New plugin codecalpha with elements to assist with WebM Alpha
+    decoding
+
+    -   codecalphademux: Split stream with GstVideoCodecAlphaMeta into
+        two streams
+    -   alphacombine: Combine two raw video stream (I420 or NV12) as one
+        stream with alpha channel (A420 or AV12)
+    -   vp8alphadecodebin: A bin to handle software decoding of VP8 with
+        alpha
+    -   vp9alphadecodebin: A bin to handle software decoding of VP9 with
+        alpha
+
+-   New hardware accelerated elements for Linux:
+
+    -   v4l2slmpeg2dec: Support for Linux Stateless MPEG2 decoders
+    -   v4l2slvp9dec: Support for Linux Stateless VP9 decoders
+    -   v4l2slvp8alphadecodebin: Support HW accelerated VP8 with alpha
+        layer decoding
+    -   v4l2slvp9alphadecodebin: Support HW accelerated VP9 with alpha
+        layer decoding
 
 New element features and additions
 
--   this section will be filled in in due course
+-   assrender: handle more font mime types; better interaction with
+    matroskademux for embedded fonts
+
+-   audiobuffersplit: Add support for specifying output buffer size in
+    bytes (not just duration)
+
+-   audiolatency: new "samplesperbuffer" property so users can configure
+    the number of samples per buffer. The default value is 240 samples
+    which is equivalent to 5ms latency with a sample rate of 48000,
+    which might be larger than actual buffer size of audio capture
+    device.
+
+-   audiomixer, audiointerleave, GstAudioAggregator: now keep a count of
+    samples that are dropped or processed as statistic and can be made
+    to post QoS messages on the bus whenever samples are dropped by
+    setting the "qos-messages" property on input pads.
+
+-   audiomixer, compositor: improved handling of new inputs added at
+    runtime. New API was added to the GstAggregator base class to allow
+    subclasses to opt into an aggregation mode where inactive pads are
+    ignored when processing input buffers
+    (gst_aggregator_set_ignore_inactive_pads(),
+    gst_aggregator_pad_is_inactive()). An “inactive pad” in this context
+    is a pad which, in live mode, hasn’t yet received a first buffer,
+    but has been waited on at least once. What would happen usually in
+    this case is that the aggregator would wait for data on this pad
+    every time, up to the maximum configured latency. This would
+    inadvertently push mixer elements in live mode to the configured
+    latency envelope and delay processing when new inputs are added at
+    runtime until these inputs have actually produced data. This is
+    usually undesirable. With this new API, new inputs can be added
+    (requested) and configured and they won’t delay the data processing.
+    Applications can opt into this new behaviour by setting the
+    "ignore-inactive-pads" property on compositor, audiomixer or other
+    GstAudioAggregator-based elements.
+
+-   cccombiner: implement “scheduling” of captions. So far cccombiner’s
+    behaviour was essentially that of a funnel: it strictly looked at
+    input timestamps to associate together video and caption buffers.
+    Now it will try to smoothly schedule caption buffers in order to
+    have exactly one per output video buffer. This might involve
+    rewriting input captions, for example when the input is CDP then
+    sequence counters are rewritten, time codes are dropped and
+    potentially re-injected if the input video frame had a time code
+    meta. This can also lead to the input drifting from synchronization,
+    when there isn’t enough padding in the input stream to catch up. In
+    that case the element will start dropping old caption buffers once
+    the number of buffers in its internal queue reaches a certain limit
+    (configurable via the "max-scheduled" property). The new original
+    funnel-like behaviour can be restored by setting the "scheduling"
+    property to FALSE.
+
+-   ccconverter: new "cdp-mode" property to specify which sections to
+    include in CDP packets (timecode, CC data, service info). Various
+    software, including ffmpeg’s Decklink support, fails parsing CDP
+    packets that contain anything but CC data in the CDP packets.
+
+-   clocksync: new "sync-to-first" property for automatic timestamp
+    offset setup: if set clocksync will set up the "ts-offset" value
+    based on the first buffer and the pipeline’s running time when the
+    first buffer arrived. The newly configured "ts-offset" in this case
+    would be the value that allows outputting the first buffer without
+    waiting on the clock. This is useful for example to feed a non-live
+    input into an already-running pipeline.
+
+-   compositor:
+
+    -   multi-threaded input conversion and compositing. Set the
+        "max-threads" property to activate this.
+    -   new "sizing-policy" property to support display aspect ratio
+        (DAR)-aware scaling. By default the image is scaled to fill the
+        configured destination rectangle without padding and without
+        keeping the aspect ratio. With sizing-policy=keep-aspect-ratio
+        the input image is scaled to fit the destination rectangle
+        specified by GstCompositorPad:{xpos, ypos, width, height}
+        properties preserving the aspect ratio. As a result, the image
+        will be centered in the destination rectangle with padding if
+        necessary.
+    -   new "zero-size-is-unscaled" property on input pads. By default
+        pad width=0 or pad height=0 mean that the stream should not be
+        scaled in that dimension. But if the "zero-size-is-unscaled"
+        property is set to FALSE a width or height of 0 is instead
+        interpreted to mean that the input image on that pad should not
+        be composited, which is useful when creating animations where an
+        input image is made smaller and smaller until it disappears.
+    -   improved handling of new inputs at runtime via
+        "ignore-inactive-pads"property (see above for details)
+    -   allow output format with alpha even if none of the inputs have
+        alpha (also glvideomixer and other GstVideoAggregator
+        subclasses)
+
+-   dashsink: add h265 codec support and signals for allowing custom
+    playlist/fragment output
+
+-   decodebin3:
+
+    -   improved decoder selection, especially for hardware decoders
+    -   make input activation “atomic” when adding inputs dynamically
+    -   better interleave handling: take into account decoder latency
+        for interleave size
+
+-   decklink:
+
+    -   Updated DeckLink SDK to 11.2 to support DeckLink 8K Pro
+    -   decklinkvideosrc:
+        -   More accurate and stable capture timestamps: use the
+            hardware reference clock time when the frame was finished
+            being captured instead of a clock time much further down the
+            road.
+        -   Automatically detect widescreen vs. normal NTSC/PAL
+
+-   encodebin:
+
+    -   add “smart encoding” support for H.265, VP8 and VP9 (i.e. only
+        re-encode where needed and otherwise pass through encoded video
+        as-is).
+    -   H264/H265 smart encoding improvements: respect user-specified
+        stream-format, but if not specified default to avc3/hvc1 with
+        in-band SPS/PPS/VPS signalling for more flexibility.
+    -   new encodebin2 element with dynamic/sometimes source pads in
+        order to support the option of doing the muxing outside of
+        encodebin, e.g. in combination with splitmuxsink.
+    -   add APIs to set element properties on encoding profiles (see
+        below)
+
+-   errorignore: new "ignore-eos" property to also ignore FLOW_EOS from
+    downstream elements
+
+-   giosrc: add support for growing source files: applications can
+    specify that the underlying file being read is growing by setting
+    the "is-growing" property. If set, the source won’t EOS when it
+    reaches the end of the file, but will instead start monitoring it
+    and will start reading data again whenever a change is detected. The
+    new "waiting-data" and "done-waiting-data" signals keep the
+    application informed about the current state.
+
+-   gtksink, gtkglsink:
+
+    -   scroll event support: forwarded as navigation events into the
+        pipeline
+    -   "video-aspect-ratio-override" property to force a specific
+        aspect ratio
+    -   "rotate-method" property and support automatic rotation based on
+        image tags
+
+-   identity: new "stats" property allows applications to retrieve the
+    number of bytes and buffers that have passed through so far.
+
+-   interlace: add support for more formats, esp 10-bit, 12-bit and
+    16-bit ones
+
+-   jack: new "low-latency" property for automatic latency-optimized
+    setting and "port-names" property to select ports explicitly
+
+-   jpegdec: support output conversion to RGB using libjpeg-turbo (for
+    certain input files)
+
+-   line21dec:
+
+    -   "mode" property to control whether and how detected closed
+        captions should be inserted in the list of existing close
+        caption metas on the input frame (if any): add, drop, or
+        replace.
+    -   "ntsc-only" property to only look for captions if video has NTSC
+        resolution
+
+-   line21enc: new "remove-caption-meta" to remove metas from output
+    buffers after encoding the captions into the video data; support for
+    CDP closed captions
+
+-   matroskademux, matroskamux: Add support for ffv1, a lossless
+    intra-frame video coding format.
+
+-   matroskamux: accept in-band SPS/PPS/VPS for H264 and H265
+    (i.e. stream-format avc3 and hev1) which allows on-the-fly
+    profile/level/resolution changes.
+
+-   matroskamux: new "cluster-timestamp-offset" property, useful for use
+    cases where the container timestamps should map to some absolute
+    wall clock time, for example.
+
+-   rtpsrc: add "caps" property to allow explicit setting of the caps
+    where needed
+
+-   mpegts: support SCTE-35 passthrough via new "send-scte35-events"
+    property on MPEG-TS demuxer tsdemux. When enabled, SCTE 35 sections
+    (eg ad placement opportunities) are forwarded as events donwstream
+    where they can be picked up again by mpegtsmux. This required a
+    semantic change in the SCTE-35 section API: timestamps are now in
+    running time instead of muxer pts.
+
+-   tsdemux: Handle PCR-less MPEG-TS streams; more robust timestamp
+    handling in certain corner cases and for poorly muxed streams.
+
+-   mpegtsmux:
+
+    -   More conformance improvements to make MPEG-TS analyzers happy:
+        -   PCR timing accuracy: Improvements to the way mpegtsmux
+            outputs PCR observations in CBR mode, so that a PCR
+            observation is always inserted when needed, so that we never
+            miss the configured pcr-interval, as that triggers various
+            MPEG-TS analyser errors.
+        -   Improved PCR/SI scheduling
+    -   Don’t write PCR until PAT/PMT are output to make sure streams
+        start cleanly with a PAT/PMT.
+    -   Allow overriding the automatic PMT PID selection via
+        application-supplied PMT_%d fields in the prog-map
+        structure/property.
+
+-   mp4mux:
+
+    -   new "first-moov-then-finalise" mode for fragmented output where
+        the output will start with a self-contained moov atom for the
+        first fragment, and then produce regular fragments. Then at the
+        end when the file is finalised, the initial moov is invalidated
+        and a new moov is written covering the entire file. This way the
+        file is a “fragmented mp4” file while it is still being written
+        out, and remains playable at all times, but at the end it is
+        turned into a regular mp4 file (with former fragment headers
+        remaining as unused junk data in the file).
+    -   support H.264 avc3 and H.265 hvc1 stream formats as input where
+        the codec data is signalled in-band inside the bitstream instead
+        of caps/file headers.
+    -   support profile/level/resolution changes for H264/H265 input
+        streams (i.e. codec data changing on the fly). Each codec_data
+        is put into its own SampleTableEntry inside the stsd, unless the
+        input is in avc3 stream format in which case it’s written
+        in-band an not in the headers.
+
+-   multifilesink: new ""min-keyframe-distance"" property to make
+    minimum distance between keyframes in next-file=key-frame mode
+    configurable instead of hard-coding it to 10 seconds.
+
+-   mxfdemux has seen a big refactoring to support non-frame wrappings
+    and more accurate timestamp/seek handling for some formats
+
+-   msdk plugin for hardware-accelerated video encoding and decoding
+    using the Intel Media SDK:
+
+    -   oneVPL support (Intel oneAPI Video Processing Library)
+    -   AV1 decoding support
+    -   H.264 decoder now supports constrained-high and progressive-high
+        profiles
+    -   H.264 encoder:
+        -   more configuration options (properties):
+            "intra-refresh-type", "min-qp" , "max-qp", "p-pyramid",
+            "dblk-idc"
+    -   H.265 encoder:
+        -   can output main-still-picture profile
+        -   now inserts HDR SEIs (mastering display colour volume and
+            content light level)
+        -   more configuration options (properties):
+            "intra-refresh-type", "min-qp" , "max-qp", "p-pyramid",
+            "b-pyramid", "dblk-idc", "transform-skip"
+        -   support for RGB 10bit format
+    -   External bitrate control in encoders
+    -   Video post proc element msdkvpp gained support for 12-bit pixel
+        formats P012_LE, Y212_LE and Y412_LE
+
+-   nvh264sldec: interlaced stream support
+
+-   openh264enc: support main, high, constrained-high and
+    progressive-high profiles
+
+-   openjpeg: support for multithreaded decoding and encoding
+
+-   rtspsrc: now supports IPv6 also for tunneled mode (RTSP-over-HTTP);
+    new "ignore-x-server-reply" property to ignore the
+    x-server-ip-address server header reply in case of HTTP tunneling,
+    as it is often broken.
+
+-   souphttpsrc: Runtime compatibility support for libsoup2 and
+    libsoup3. libsoup3 is the latest major version of libsoup, but
+    libsoup2 and libsoup3 can’t co-exist in the same process because
+    there is no namespacing or versioning for GObject types. As a
+    result, it would be awkward if the GStreamer souphttpsrc plugin
+    linked to a specific version of libsoup, because it would only work
+    with applications that use the same version of libsoup. To make this
+    work, the soup plugin now tries to determine the libsoup version
+    used by the application (and its other dependencies) at runtime on
+    systems where GStreamer is linked dynamically. libsoup3 support is
+    still considered somewhat experimental at this point.
+
+-   srtsrc, srtsink: add signals for the application to accept/reject
+    incoming connections
+
+-   timeoverlay: new elapsed-running-time time mode which shows the
+    running time since the first running time (and each flush-stop).
+
+-   udpsrc: new timestamping mode to retrieve packet receive timestamps
+    from the kernel via socket control messages (SO_TIMESTAMPNS) on
+    supported platforms
+
+-   uritranscodebin: new setup-source and element-setup signals for
+    applications to configure elements used
+
+-   v4l2codecs plugin gained support for 4x4 and 32x32 tile formats
+    enabling some platforms or direct renders. Important memory usage
+    improvement.
+
+-   v4l2slh264dec now implements the final Linux uAPI as shipped on
+    Linux 5.11 and later.
+
+-   valve: add "drop-mode" property and provide two new modes of
+    operation: in drop-mode=forward-sticky-events sticky events
+    (stream-start, segment, tags, caps, etc.) are forwarded downstream
+    even when dropping is enabled; drop-mode=transform-to-gap will in
+    addition also convert buffers into gap events when dropping is
+    enabled, which lets downstream elements know that time is advancing
+    and might allow for preroll in many scenarios. By default all events
+    and all buffers are dropped when dropping is enabled, which can
+    cause problems with caps negotiation not progressing or branches not
+    prerolling when dropping is enabled.
+
+-   videocrop: support for many more pixel formats, e.g. planar YUV
+    formats with > 8bits and GBR* video formats; can now also accept
+    video not backed by system memory as long as downstream supports the
+    GstCropMeta
+
+-   videotestsrc: new smpte-rp-219 pattern for SMPTE75 RP-219 conformant
+    color bars
+
+-   vp8enc: finish support for temporal scalability: two new properties
+    ("temporal-scalability-layer-flags",
+    "temporal-scalability-layer-sync-flags") and a unit change on the
+    "temporal-scalability-target-bitrate" property (now expects bps);
+    also make temporal scalability details available to RTP payloaders
+    as buffer metadata.
+
+-   vp9enc: new properties to tweak encoder performance:
+
+    -   "aq-mode" to configure adaptive quantization modes
+    -   "frame-parallel-decoding" to configure whether to create a
+        bitstream that reduces decoding dependencies between frames
+        which allows staged parallel processing of more than one video
+        frames in the decoder. (Defaults to TRUE)
+    -   "row-mt", "tile-columns" and "tile-rows" so multithreading can
+        be enabled on a per-tile basis, instead of on a per tile-column
+        basis. In combination with the new "tile-rows" property, this
+        allows the encoder to make much better use of the available CPU
+        power.
+
+-   vp9dec, vp9enc: add support for 10-bit 4:2:0 and 4:2:2 YUV, as well
+    as 8-bit 4:4:4
+
+-   vp8enc, vp9enc now default to “good quality” for the deadline
+    property rather then “best quality”. Having the deadline set to best
+    quality causes the encoder to be absurdly slow, most real-life users
+    will prefer good-enough quality with better performance instead.
+
+-   wpesrc:
+
+    -   implement audio support: a new sometimes source pad will be
+        created for each audio stream created by the web engine.
+    -   move wpesrc to wpevideosrc and add a wrapper bin wpesrc to also
+        support audio
+    -   also handles web:// URIs now (same as cefsrc)
+    -   post messages with the estimated load progress on the bus
 
-Plugin and library moves
+-   x265enc: add negative DTS support, which means timestamps are now
+    offset by 1h same as with x264enc
+
+RTP Payloaders and Depayloaders
+
+-   rtpisacpay, rtpisacdepay: new RTP payloader and depayloader for iSAC
+    audio codec
+
+-   rtph264depay:
+
+    -   new "request-keyframe" property to make the depayloader
+        automatically request a new keyframe from the sender on packet
+        loss, consistent with the new property on rtpvp8depay.
+    -   new "wait-for-keyframe" property to make depayloader wait for a
+        new keyframe at the beginning and after packet loss (only
+        effective if the depayloader outputs AUs), consistent with the
+        existing property on rtpvp8depay.
+
+-   rtpopuspay, rtpopusdepay: support libwebrtc-compatible multichannel
+    audio in addition to the previously supported multichannel audio
+    modes
+
+-   rtpopuspay: add DTX (Discontinuous Transmission) support
+
+-   rtpvp8depay: new "request-keyframe" property to make the depayloader
+    automatically request a new keyframe from the sender on packet loss.
+
+-   rtpvp8pay: temporal scaling support
 
--   this section will be filled in in due course
+-   rtpvp9depay: Improved SVC handling (aggregate all layers)
+
+RTP Infrastructure
+
+-   rtpst2022-1-fecdec, rtpst2022-1-fecenc: new elements providing SMPTE
+    2022-1 2-D Forward Error Correction. More details in Mathieu’s blog
+    post.
+
+-   rtpreddec: BUNDLE support
+
+-   rtpredenc, rtpulpfecenc: add support for Transport-wide Congestion
+    Control (TWCC)
+
+-   rtpsession: new "twcc-feedback-interval" property to allow RTCP TWCC
+    reports to be scheduled on a timer instead of per marker-bit.
+
+Plugin and library moves
 
 -   There were no plugin moves or library moves in this cycle.
 
@@ -56,8 +998,6 @@ Plugin removals
 
 The following elements or plugins have been removed:
 
--   this section will be filled in in due course
-
 -   The ofa audio fingerprinting plugin has been removed. The MusicIP
     database has been defunct for years so this plugin is likely neither
     useful nor used by anyone.
@@ -71,135 +1011,575 @@ The following elements or plugins have been removed:
 
 Miscellaneous API additions
 
--   this section will be filled in in due course
+Core
+
+-   gst_buffer_new_memdup() is a convenience function for the
+    widely-used gst_buffer_new_wrapped(g_memdup(data,size),size)
+    pattern.
+
+-   gst_caps_features_new_single() creates a new single GstCapsFeatures,
+    avoiding the need to use the vararg function with NULL terminator
+    for simple cases.
+
+-   gst_element_type_set_skip_documentation() can be used by plugins to
+    signal that certain elements should not be included in the GStreamer
+    plugin documentation. This is useful for plugins where elements are
+    registered dynamically based on hardware capabilities and/or where
+    the available plugins and properties vary from system to system.
+    This is used in the d3d11 plugin for example to ensure that only the
+    list of default elements is advertised in the documentation.
+
+-   gst_type_find_suggest_empty_simple() is a new convenience function
+    for typefinders for cases where there’s only a media type and no
+    other fields.
+
+-   New API to create elements and set properties at construction time,
+    which is not only convenient, but also allows GStreamer elements to
+    have construct-only properties: gst_element_factory_make_full(),
+    gst_element_factory_make_valist(),
+    gst_element_factory_make_with_properties(),
+    gst_element_factory_create_full(),
+    gst_element_factory_create_valist(),
+    gst_element_factory_create_with_properties().
+
+-   GstSharedTaskPool: new “shared” task pool subclass with slightly
+    different default behaviour than the existing GstTaskPool which
+    would create unlimited number of threads for new tasks. The shared
+    taskpool creates up to N threads (default: 1) and then distributes
+    pending tasks to those threads round-robin style, and blocks if no
+    thread is available. It is possible to join tasks. This can be used
+    by plugins to implement simple multi-threaded processing and is used
+    for the new multi-threaded video conversion and compositing done in
+    GstVideoAggregator, videoconverter and compositor.
+
+Plugins Base Utils library
+
+-   GstDiscoverer:
+
+    -   gst_discoverer_container_info_get_tags() was added to retrieve
+        global/container tags (vs. per-stream tags). Per-Stream tags can
+        be retrieved via the existing
+        gst_discoverer_stream_info_get_tags().
+        gst_discoverer_info_get_tags(), which for many files returns a
+        confusing mix of stream and container tags, has been deprecated
+        in favour of the container/stream-specific functions.
+    -   gst_discoverer_stream_info_get_stream_number() returns a unique
+        integer identifier for a given stream within the given
+        GstDiscoverer context. (If this matches the stream number inside
+        the container bitstream that’s by coincidence and not by
+        design.)
+
+-   gst_pb_utils_get_caps_description_flags() can be used to query
+    whether certain caps represent a container, audio, video, image,
+    subtitles, tags, or something else. This only works for formats
+    known to GStreamer.
+
+-   gst_pb_utils_get_file_extension_from_caps() returns a possible file
+    extension for given caps.
+
+-   gst_codec_utils_h264_get_profile_flags_level(): Parses profile,
+    flags, and level from H264 AvcC codec_data. The format of H264 AVCC
+    extradata/sequence_header is documented in the ITU-T H.264
+    specification section 7.3.2.1.1 as well as in ISO/IEC 14496-15
+    section 5.3.3.1.2.
+
+-   gst_codec_utils_caps_get_mime_codec() to convert caps to a RFC 6381
+    compatible MIME codec string codec. Useful for providing the codecs
+    field inside the Content-Type HTTP header for containerized formats,
+    such as mp4 or matroska.
+
+GStreamer OpenGL integration library and plugins
+
+-   glcolorconvert: added suppport for converting the video formats
+    A420, AV12, BGR, BGRA, RGBP and BGRP.
+
+-   Added support to GstGLBuffer for persistent buffer mappings where a
+    Pixel Buffer Object (PBO) can be mapped by both the CPU and the GPU.
+    This removes a memcpy() when uploading textures or vertices
+    particularly when software decoders (e.g. libav) are direct
+    rendering into our memory. Improves transfer performance
+    significantly. Requires OpenGL 4.4, GL_ARB_buffer_storage or
+    GL_EXT_buffer_storage
+
+-   Added various helper functions for handling 4x4 matrices of affine
+    transformations as used by GstVideoAffineTransformationMeta.
+
+-   Add support to GstGLContext for allowing the application to control
+    the config (EGLConfig, GLXConfig, etc) used when creating the OpenGL
+    context. This allows the ability to choose between RGB16 or RGB10A2
+    or RGBA8 back/front buffer configurations that were previously
+    hardcoded. GstGLContext also supports retrieving the configuration
+    it was created with or from an externally provide OpenGL context
+    handle. This infrastructure is also used to create a compatible
+    config from an application/externally provided OpenGL context in
+    order to improve compatibility with other OpenGL frameworks and GUI
+    toolkits. A new environment variable GST_GL_CONFIG was also added to
+    be able to request a specific configuration from the command line.
+    Note: different platforms will have different functionality
+    available.
+
+-   Add support for choosing between EGL and WGL at runtime when running
+    on Windows. Previously this was a build-time switch. Allows use in
+    e.g. Gtk applications on Windows that target EGL/ANGLE without
+    recompiling GStreamer. gst_gl_display_new_with_type() can be used by
+    applications to choose a specific display type to use.
+
+-   Build fixes to explicitly check for Broadcom-specific libraries on
+    older versions of the Raspberry Pi platform. The Broadcom OpenGL ES
+    and EGL libraries have different filenames. Using the vc4 Mesa
+    driver on the Raspberry Pi is not affected.
+
+-   Added support to glupload and gldownload for transferring RGBA
+    buffers using the memory:NVMM available on the Nvidia Tegra family
+    of embedded devices.
+
+-   Added support for choosing libOpenGL and libGLX as used in a GLVND
+    environment on unix-based platforms. This allows using desktop
+    OpenGL and EGL without pulling in any GLX symbols as would be
+    required with libGL.
+
+Video library
+
+-   New raw video formats:
+
+    -   AV12 (NV12 with alpha plane)
+    -   RGBP and BGRP (planar RGB formats)
+    -   ARGB64 variants with specified endianness instead of host
+        endianness:
+        -   ARGB64_LE, ARGB64_BE
+        -   RGBA64_BE, RGBA64_LE
+        -   BGRA64_BE, BGRA64_LE
+        -   ABGR64_BE, ABGR64_LE
+
+-   gst_video_orientation_from_tag() is new convenience API to parse the
+    image orientation from a GstTagList.
+
+-   GstVideoDecoder subframe support (see below)
+
+-   GstVideoCodecState now also carries some HDR metadata
+
+-   Ancillary video data: implement transform functions for AFD/Bar
+    metas, so they will be forwarded in more cases
+
+MPEG-TS library
+
+This library only handles section parsing and such, see above for
+changes to the actual mpegtsmux and mpegtsdemux elements.
+
+-   many additions and improvements to SCTE-35 section parsing
+-   new API for fetching extended descriptors:
+    gst_mpegts_find_descriptor_with_extension()
+-   add support for SIT sections (Selection Information Tables)
+-   expose event-from-section constructor gst_event_new_mpegts_section()
+-   parse Audio Preselection Descriptor needed for Dolby AC-4
+
+GstWebRTC library + webrtcbin
+
+-   Change the way in which sink pads and transceivers are matched
+    together to support easier usage. If a pad is created without a
+    specific index (i.e. using sink_%u as the pad template), then an
+    available compatible transceiver will be searched for. If a specific
+    index is requested (i.e. sink_1) then if a transceiver for that
+    m-line already exists, that transceiver must match the new sink pad
+    request. If there is no transceiver available in either scenario, a
+    new transceiver is created. If a mixture of both sink_1 and sink_%u
+    requests result in an impossible situation, an error will be
+    produced at pad request time or from create offer/answer.
+
+-   webrtcbin now uses regular ICE nomination instead of libnice’s
+    default of aggressive ICE nomination. Regular ICE nomination is the
+    default recommended by various relevant standards and improves
+    connectivity in specific network scenarios.
+
+-   Add support for limiting the port range used for RTP with the
+    addition of the min-rtp-port and max-rtp-port properties on the ICE
+    object.
+
+-   Expose the SCTP transport as a property on webrtcbin to more closely
+    match the WebRTC specification.
+
+-   Added support for taking into account the data channel transport
+    state when determining the value of the "connection-state" property.
+    Previous versions of the WebRTC spec did not include the data
+    channel state when computing this value.
+
+-   Add configuration for choosing the size of the underlying sockets
+    used for transporting media data
+
+-   Always advertise support for the transport-cc RTCP feedback protocol
+    as rtpbin supports it. For full support, the configured caps (input
+    or through codec-preferences) need to include the relevant RTP
+    header extension.
+
+-   Numerous fixes to caps and media handling to fail-fast when an
+    incompatible situation is detected.
+
+-   Improved support for attaching the required media after a remote
+    offer has been set.
+
+-   Add support for dynamically changing the amount of FEC used for a
+    particular stream.
+
+-   webrtcbin now stops further SDP processing at the first error it
+    encounters.
+
+-   Completed support for either local or the remote closing a data
+    channel.
+
+-   Various fixes when performing BUNDLEing of the media streams in
+    relation to RTX and FEC usage.
+
+-   Add support for writing out QoS DSCP marking on outgoing packets to
+    improve reliability in some network scenarios.
+
+-   Improvements to the statistics returned by the get-stats signal
+    including the addition of the raw statistics from the internal
+    RTPSource, the TWCC stats when available.
+
+-   The webrtc library does not expose any objects anymore with public
+    fields. Instead properties have been added to replace that
+    functionality. If you are accessing such fields in your application,
+    switch to the corresponding properties.
+
+GstCodecs and Video Parsers
+
+-   Support for render delays to improve throughput across all CODECs
+    (used with NVDEC and V4L2).
+-   lots of improvements to parsers and the codec parsing decoder base
+    classes (H264, H265, VP8, VP9, AV1, MPEG-2) used for various
+    hardware-accelerate decoder APIs.
+
+Bindings support
+
+-   gst_allocation_params_new() allocates a GstAllocationParams struct
+    on the heap. This should only be used by bindings (and freed via
+    gst_allocation_params_free() then). In C code you would allocate
+    this on the stack and only init it in place.
+
+-   gst_debug_log_literal() can be used to log a string to the debug log
+    without going through any printf format expansion and associated
+    overhead. This is mostly useful for bindings such as the Rust
+    bindings which may have done their own formatting already .
+
+-   Provide non-inlined versions of refcounting APIs for various
+    GStreamer mini objects, so that they can be consumed by bindings
+    (e.g. gstreamer-sharp): gst_buffer_ref, gst_buffer_unref,
+    gst_clear_buffer, gst_buffer_copy, gst_buffer_replace,
+    gst_buffer_list_ref, gst_buffer_list_unref, gst_clear_buffer_list,
+    gst_buffer_list_copy, gst_buffer_list_replace, gst_buffer_list_take,
+    gst_caps_ref, gst_caps_unref, gst_clear_caps, gst_caps_replace,
+    gst_caps_take, gst_context_ref, gst_context_unref, gst_context_copy,
+    gst_context_replace, gst_event_replace, gst_event_steal,
+    gst_event_take, gst_event_ref, gst_event_unref, gst_clear_event,
+    gst_event_copy, gst_memory_ref, gst_memory_unref, gst_message_ref,
+    gst_message_unref, gst_clear_message, gst_message_copy,
+    gst_message_replace, gst_message_take, gst_promise_ref,
+    gst_promise_unref, gst_query_ref, gst_query_unref, gst_clear_query,
+    gst_query_copy, gst_query_replace, gst_query_take, gst_sample_ref,
+    gst_sample_unref, gst_sample_copy, gst_tag_list_ref,
+    gst_tag_list_unref, gst_clear_tag_list, gst_tag_list_replace,
+    gst_tag_list_take, gst_uri_copy, gst_uri_ref, gst_uri_unref,
+    gst_clear_uri.
+
+-   expose a GType for GstMiniObject
+
+-   gst_device_provider_probe() now returns non-floating device object
+
+API Deprecations
+
+-   gst_element_get_request_pad() has been deprecated in favour of the
+    newly-added gst_element_request_pad_simple() which does the exact
+    same thing but has a less confusing name that hopefully makes clear
+    that the function request a new pad rather than just retrieves an
+    already-existing request pad.
+
+-   gst_discoverer_info_get_tags(), which for many files returns a
+    confusing mix of stream and container tags, has been deprecated in
+    favour of the container-specific and stream-specific functions,
+    gst_discoverer_container_info_get_tags() and
+    gst_discoverer_stream_info_get_tags().
+
+-   gst_video_sink_center_rect() was deprecated in favour of the more
+    generic newly-added gst_video_center_rect().
+
+-   The GST_MEMORY_FLAG_NO_SHARE flag has been deprecated, as it tends
+    to cause problems and prevents sub-buffering. If pooling or lifetime
+    tracking is required, memories should be allocated through a custom
+    GstAllocator instead of relying on the lifetime of the buffers the
+    memories were originally attached to, which is fragile anyway.
+
+-   The GstPlayer high-level playback library is being replaced with the
+    new GstPlay library (see above). GstPlayer should be considered
+    deprecated at this point and will be marked as such in the next
+    development cycle. Applications should be ported to GstPlay.
+
+-   Gstreamer Editing Services: ges_video_transition_set_border(),
+    ges_video_transition_get_border()
+    ges_video_transition_set_inverted()
+    ges_video_transition_is_inverted() have been deprecated, use
+    ges_timeline_element_set_children_properties() instead.
 
 Miscellaneous performance, latency and memory optimisations
 
--   this section will be filled in in due course
+More video conversion fast paths
 
-Miscellaneous other changes and enhancements
+-   v210 ↔ I420, YV12, Y42B, UYVY and YUY2
+-   A420 → RGB
+
+Less jitter when waiting on the system clock
+
+-   Better system clock wait accuracy, less jitter: where available,
+    clock_nanosleep is used for higher accuracy for waits below 500
+    usecs, and waits below 2ms will first use the regular waiting system
+    and then clock_nanosleep for the remainder. The various wait
+    implementation have a latency ranging from 50 to 500+ microseconds.
+    While this is not a major issue when dealing with a low number of
+    waits per second (for ex: video), it does introduce a non-negligible
+    jitter for synchronisation of higher packet rate systems.
+
+Video decoder subframe support
 
--   this section will be filled in in due course
+-   The GstVideoDecoder base class gained API to process input at the
+    sub-frame level. That way video decoders can start decoding slices
+    before they have received the full input frame in its entirety (to
+    the extent this is supported by the codec, of course). This helps
+    with CPU utilisation and reduces latency.
 
-Tracing framework and debugging improvements
+-   This functionality is now being used in the OpenJPEG JPEG 2000
+    decoder, the FFmpeg H.264 decoder (in case of NAL-aligned input) and
+    the OpenMAX H.264/H.265 decoders (in case of NAL-aligned input).
 
--   this section will be filled in in due course
+Miscellaneous other changes and enhancements
+
+-   GstDeviceMonitor no longer fails to start just because one of the
+    device providers failed to start. That could happen for example on
+    systems where the pulseaudio device provider is installed, but
+    pulseaudio isn’t actually running but ALSA is used for audio
+    instead. In the same vein the device monitor now keeps track of
+    which providers have been started (via the new
+    gst_device_provider_is_started()) and only stops actually running
+    device providers when stopping the device monitor.
+
+-   On embedded systems it can be useful to create a registry that can
+    be shared and read by multiple processes running as different users.
+    It is now possible to set the new GST_REGISTRY_MODE environment
+    variable to specify the file mode for the registry file, which by
+    default is set to be only user readable/writable.
+
+-   GstNetClientClock will signal lost sync in case the remote time
+    resets (e.g. because device power cycles), by emitting the “synced”
+    signal with synced=FALSE parameter, so applications can take action.
+
+-   gst_value_deserialize_with_pspec() allows deserialization with a
+    hint for what the target GType should be. This allows for example
+    passing arrays of flags through the command line or
+    gst_util_set_object_arg(), eg: foo="<bar,bar+baz>".
+
+-   It’s now allowed to create an empty GstVideoOverlayComposition
+    without any rectangles by passing a NULL rectangle to
+    gst_video_overlay_composition_new(). This is useful for bindings and
+    simplifies application code in some places.
+
+Tracing framework, debugging and testing improvements
+
+-   New factories tracer to list loaded elements (and other plugin
+    features). This can be useful to collect a list of elements needed
+    for an application, which then in turn can be used to create a
+    tailored minimal GStreamer build that contains just the elements
+    needed and nothing else.
+-   New plugin-feature-loaded tracing hook for use by tracers like the
+    new factories tracer
+
+-   GstHarness: Add gst_harness_set_live() so that harnesses can be set
+    to non-live and return is-live=false in latency queries if needed.
+    Default behaviour is to always return is-live=true in latency
+    queries.
+
+-   navseek: new "hold-eos" property. When enabled, the element will
+    hold back an EOS event until the next keystroke (via navigation
+    events). This can be used to keep a video sink showing the last
+    frame of a video pipeline until a key is pressed instead of tearing
+    it down immediately on EOS.
+
+-   New fakeaudiosink element: mimics an audio sink and can be used for
+    testing and CI pipelines on systems where no audio system is
+    installed or running. It differs from fakesink in that it only
+    support audio caps and syncs to the clock by default like a normal
+    audio sink. It also implements the GstStreamVolume interface like
+    most audio sinks do.
+
+-   New videocodectestsink element for video codec conformance testing:
+    Calculates MD5 checksums for video frames and skips any padding
+    whilst doing so. Can optionally also write back the video data with
+    padding removed into a file for easy byte-by-byte comparison with
+    reference data.
 
 Tools
 
--   this section will be filled in in due course
+gst-inspect-1.0
+
+-   Can sort the list of plugins by passing --sort=name as command line
+    option
+
+gst-launch-1.0
+
+-   will now error out on top-level properties that don’t exist and
+    which were silently ignored before
+-   On Windows the high-resolution clock is enabled now, which provides
+    better clock and timer performance on Windows (see Windows section
+    below for more details).
+
+gst-play-1.0
+
+-   New --start-position command line argument to start playback from
+    the specified position
+-   Audio can be muted/unmuted in interactive mode by pressing the m
+    key.
+-   On Windows the high-resolution clock is enabled now (see Windows
+    section below for more details)
+
+gst-device-monitor-1.0
+
+-   New --include-hidden command line argument to also show “hidden”
+    device providers
+
+ges-launch-1.0
+
+-   New interactive mode that allows seeking and such. Can be disabled
+    by passing the --no-interactive argument on the command line.
+-   Option to forward tags
+-   Allow using an existing clip to determine the rendering format (both
+    topology and profile) via new --profile-from command line argument.
 
 GStreamer RTSP server
 
--   this section will be filled in in due course
+-   GstRTSPMediaFactory gained API to disable RTCP
+    (gst_rtsp_media_factory_set_enable_rtcp(), "enable-rtcp" property).
+    Previously RTCP was always allowed for all RTSP medias. With this
+    change it is possible to disable RTCP completely, no matter if the
+    client wants to do RTCP or not.
+
+-   Make a mount point of / work correctly. While not allowed by the
+    RTSP 2 spec, the RTSP 1 spec is silent on this and it is used in the
+    wild. It is now possible to use / as a mount path in
+    gst-rtsp-server, e.g. rtsp://example.com/ would work with this now.
+    Note that query/fragment parts of the URI are not necessarily
+    correctly handled, and behaviour will differ between various
+    client/server implementations; so use it if you must but don’t bug
+    us if it doesn’t work with third party clients as you’d hoped.
+
+-   multithreading fixes (races, refcounting issues, deadlocks)
+
+-   ONVIF audio backchannel fixes
+
+-   ONVIF trick mode optimisations
+
+-   rtspclientsink: new "update-sdp" signal that allows updating the SDP
+    before sending it to the server via ANNOUNCE. This can be used to
+    add additional metadata to the SDP, for example. The order and
+    number of medias must not be changed, however.
 
 GStreamer VAAPI
 
--   this section will be filled in in due course
+-   new AV1 decoder element (vaapiav1dec)
+
+-   H264 decoder: handle stereoscopic 3D video with frame packing
+    arrangement SEI messages
+
+-   H265 encoder: added Screen Content Coding extensions support
+
+-   H265 decoder: gained MAIN_444_12 profile support (decoded to
+    Y412_LE), and 4:2:2 12-bits support (decoded to Y212_LE)
+
+-   vaapipostproc: gained BT2020 color standard support
+
+-   vaapidecode: now generates caps templates dynamically at runtime in
+    order to advertise actually supported caps instead of all
+    theoretically supported caps.
+
+-   GST_VAAPI_DRM_DEVICE environment variable to force a specified DRM
+    device when a DRM display is used. It is ignored when other types of
+    displays are used. By default /dev/dri/renderD128 is used for DRM
+    display.
 
 GStreamer OMX
 
--   this section will be filled in in due course
+-   subframe support in H.264/H.265 decoders
 
 GStreamer Editing Services and NLE
 
--   this section will be filled in in due course
+-   framepositioner: new "operator" property to access blending modes in
+    the compositor
+-   timeline: Implement snapping to markers
+-   smart-mixer: Add support for d3d11compositor and glvideomixer
+-   titleclip: add "draw-shadow" child property
+-   ges:// URI support to define a timeline from a description.
+-   command-line-formatter
+    -   Add track management to timeline description
+    -   Add keyframe support
+-   ges-launch-1.0:
+    -   Add an interactive mode where we can seek etc…
+    -   Add option to forward tags
+    -   Allow using an existing clip to determine the rendering format
+        (both topology and profile) via new --profile-from command line
+        argument.
+-   Fix static build
 
 GStreamer validate
 
--   this section will be filled in in due course
+-   report: Add a way to force backtraces on reports even if not a
+    critical issue (GST_VALIDATE_ISSUE_FLAGS_FORCE_BACKTRACE)
+-   Add a flag to gst_validate_replace_variables_in_string() allow
+    defining how to resolve variables in structs
+-   Add gst_validate_bin_monitor_get_scenario() to get the bin monitor
+    scenario, which is useful for applications that use Validate
+    directly.
+-   Add an expected-values parameter to wait, message-type=XX allowing
+    more precise filtering of the message we are waiting for.
+-   Add config file support: each test can now use a config file for the
+    given media file used to test.
+-   Add support to check properties of object properties
+-   scenario: Add an "action-done" signal to signal when an action is
+    done
+-   scenario: Add a "run-command" action type
+-   scenario: Allow forcing running action on idle from scenario file
+-   scenario: Allow iterating over arrays in foreach
+-   scenario: Rename ‘interlaced’ action to ‘non-blocking’
+-   scenario: Add a non-blocking flag to the wait signal
 
 GStreamer Python Bindings
 
--   this section will be filled in in due course
+-   Fixes for Python 3.10
+-   Various build fixes
+-   at least one known breaking change caused by g-i annotation changes
+    (see below)
 
 GStreamer C# Bindings
 
--   this section will be filled in in due course
+-   Fix GstDebugGraphDetails enum
+-   Updated to latests GtkSharp
+-   Updated to include GStreamer 1.20 API
 
 GStreamer Rust Bindings and Rust Plugins
 
-The GStreamer Rust bindings are released separately with a different
-release cadence that’s tied to gtk-rs, but the latest release has
-already been updated for the upcoming new GStreamer 1.20 API.
-
-gst-plugins-rs, the module containing GStreamer plugins written in Rust,
-has also seen lots of activity with many new elements and plugins.
-
-What follows is a list of elements and plugins available in
-gst-plugins-rs, so people don’t miss out on all those potentially useful
-elements that have no C equivalent.
-
--   FIXME: add new elements
-
-Rust audio plugins
-
--   audiornnoise: New element for audio denoising which implements the
-    noise removal algorithm of the Xiph RNNoise library, in Rust
--   rsaudioecho: Port of the audioecho element from gst-plugins-good
-    rsaudioloudnorm: Live audio loudness normalization element based on
-    the FFmpeg af_loudnorm filter
--   claxondec: FLAC lossless audio codec decoder element based on the
-    pure-Rust claxon implementation
--   csoundfilter: Audio filter that can use any filter defined via the
-    Csound audio programming language
--   lewtondec: Vorbis audio decoder element based on the pure-Rust
-    lewton implementation
-
-Rust video plugins
-
--   cdgdec/cdgparse: Decoder and parser for the CD+G video codec based
-    on a pure-Rust CD+G implementation, used for example by karaoke CDs
--   cea608overlay: CEA-608 Closed Captions overlay element
--   cea608tott: CEA-608 Closed Captions to timed-text (e.g. VTT or SRT
-    subtitles) converter
--   tttocea608: CEA-608 Closed Captions from timed-text converter
--   mccenc/mccparse: MacCaption Closed Caption format encoder and parser
--   sccenc/sccparse: Scenarist Closed Caption format encoder and parser
--   dav1dec: AV1 video decoder based on the dav1d decoder implementation
-    by the VLC project
--   rav1enc: AV1 video encoder based on the fast and pure-Rust rav1e
-    encoder implementation
--   rsflvdemux: Alternative to the flvdemux FLV demuxer element from
-    gst-plugins-good, not feature-equivalent yet
--   rsgifenc/rspngenc: GIF/PNG encoder elements based on the pure-Rust
-    implementations by the image-rs project
-
-Rust text plugins
-
--   textwrap: Element for line-wrapping timed text (e.g. subtitles) for
-    better screen-fitting, including hyphenation support for some
-    languages
-
-Rust network plugins
-
--   reqwesthttpsrc: HTTP(S) source element based on the Rust
-    reqwest/hyper HTTP implementations and almost feature-equivalent
-    with the main GStreamer HTTP source souphttpsrc
--   s3src/s3sink: Source/sink element for the Amazon S3 cloud storage
--   awstranscriber: Live audio to timed text transcription element using
-    the Amazon AWS Transcribe API
-
-Generic Rust plugins
-
--   sodiumencrypter/sodiumdecrypter: Encryption/decryption element based
-    on libsodium/NaCl
--   togglerecord: Recording element that allows to pause/resume
-    recordings easily and considers keyframe boundaries
--   fallbackswitch/fallbacksrc: Elements for handling potentially
-    failing (network) sources, restarting them on errors/timeout and
-    showing a fallback stream instead
--   threadshare: Set of elements that provide alternatives for various
-    existing GStreamer elements but allow to share the streaming threads
-    between each other to reduce the number of threads
--   rsfilesrc/rsfilesink: File source/sink elements as replacements for
-    the existing filesrc/filesink elements
+-   The GStreamer Rust bindings are released separately with a different
+    release cadence that’s tied to gtk-rs, but the latest release has
+    already been updated for the upcoming new GStreamer 1.20 API (v1_20
+    feature).
+
+-   gst-plugins-rs, the module containing GStreamer plugins written in
+    Rust, has also seen lots of activity with many new elements and
+    plugins. See the New Elements section above for a list of new Rust
+    elements.
 
 Build and Dependencies
 
 -   Meson 0.59 or newer is required to build GStreamer now.
 
--   FIXME: this section will be filled in in due course
+-   The GLib requirement has been bumped to GLib 2.56 or newer (from
+    March 2018).
+
+-   The wpe plugin now requires wpe >= 2.28 and wpebackend-fdo >= 1.8
 
 Explicit opt-in required for build of certain plugins with (A)GPL dependencies
 
@@ -211,11 +1591,9 @@ even if the required dependencies are available.
 See Building plugins with (A)GPL-licensed dependencies for more details
 and a non-exhaustive list of plugins affected.
 
-gst-build: replaced by Monorepo
+gst-build: replaced by mono repository
 
--   this section will be filled in in due course
-
--   FIXME: describe + link to Monorepo FAQ
+See mono repository section above and the GStreamer mono repository FAQ.
 
 Cerbero
 
@@ -223,76 +1601,254 @@ Cerbero is a meta build system used to build GStreamer plus dependencies
 on platforms where dependencies are not readily available, such as
 Windows, Android, iOS and macOS.
 
-General improvements
+General Cerbero improvements
 
--   this section will be filled in in due course
+-   Plugin removed: libvisual
+-   New plugins: rtpmanagerbad and rist
 
-macOS / iOS
+macOS / iOS specific Cerbero improvements
 
--   this section will be filled in in due course
+-   XCode 12 support
+-   macOS OS release support is now future-proof, similar to iOS
+-   macOS Apple Silicon (ARM64) cross-compile support has been added
+-   macOS Apple Silicon (ARM64) native support is currently experimental
 
-Windows
+Windows specific Cerbero improvements
 
--   this section will be filled in in due course
+-   Visual Studio 2022 support has been added
+-   bootstrap is faster since it requires building fewer build-tools
+    recipes on Windows
+-   package is faster due to better scheduling of recipe stages and
+    elimination of unnecessary autotools regeneration
+-   The following plugins are no longer built on Windows:
+    -   a52dec (another decoder is still available in libav)
+    -   dvdread
+    -   resindvd
 
 Windows MSI installer
 
--   this section will be filled in in due course
+-   no major changes
 
-Linux
+Linux specific Cerbero improvements
 
--   this section will be filled in in due course
+-   Fedora, Debian OS release support is now more future-proof
+-   Amazon Linux 2 support has been added
 
-Android
+Android specific Cerbero improvements
 
--   this section will be filled in in due course
+-   no major changes
 
 Platform-specific changes and improvements
 
 Android
 
--   this section will be filled in in due course
+-   No major changes
 
 macOS and iOS
 
--   this section will be filled in in due course
+-   applemedia: add ProRes support to vtenc and vtdec
 
 Windows
 
--   this section will be filled in in due course
+-   On Windows the high-resolution clock is enabled now in the
+    gst-launch-1.0 and gst-play-1.0 command line tools, which provides
+    better clock and timer performance on Windows, at the cost of higher
+    power consumption. By default, without the high-resolution clock
+    enabled, the timer precision on Windows is system-dependent and may
+    be as bad as 15ms which is not good enough for many multimedia
+    applications. Developers may want to do the same in their Windows
+    applications if they think it’s a good idea for their application
+    use case, and depending on the Windows version they target. This is
+    not done automatically by GStreamer because on older Windows
+    versions (pre-Windows 10) this affects a global Windows setting and
+    also there’s a power consumption vs. performance trade-off that may
+    differ from application to application.
+
+-   dxgiscreencapsrc now supports resolution changes
+
+-   The wasapi2 audio plugin was rewritten and now has a higher rank
+    than the old wasapi plugin since it has a number of additional
+    features such as automatic stream routing, and no
+    known-but-hard-to-fix issues. The plugin is always built if the
+    Windows 10 SDK is available now.
+
+-   The wasapi device providers now detect and notify dynamic device
+    additions/removals
+
+-   d3d11screencapturesrc: new desktop capture element, including
+    GstDeviceProvider implementation to enumerate/select target monitors
+    for capture.
+
+-   Direct3D11/DXVA decoder now supports AV1 and MPEG2 codecs
+    (d3d11av1dec, d3d11mpeg2dec)
+
+-   VP9 decoding got more reliable and stable thanks to a newly written
+    codec parser
+
+-   Support for decoding interlaced H.264/AVC streams
+
+-   Hardware-accelerated video deinterlacing (d3d11deinterlace) and
+    video mixing (d3d11compositor)
+
+-   Video mixing with the Direct3D11 API (d3d11compositor)
+
+-   MediaFoundation API based hardware encoders gained the ability to
+    receive Direct3D11 textures as an input
+
+-   Seungha’s blog post “GStreamer ❤ Windows: A primer on the cool stuff
+    you’ll find in the 1.20 release” describes many of the
+    Windows-related improvements in more detail
 
 Linux
 
--   this section will be filled in in due course
+-   bluez: LDAC Bluetooth audio codec support in a2dpsink and avdtpsink,
+    as well as an LDAC RTP payloader (rtpldacpay) and an LDAC audio
+    encoder (ldacenc)
+
+-   kmssink: gained support for NV24, NV61, RGB16/BGR16 formats;
+    auto-detect NVIDIA Tegra driver
 
 Documentation improvements
 
--   this section will be filled in in due course
+-   hardware-accelerated GPU plugins will now no longer always list all
+    the element variants for all available GPUs, since those are
+    system-dependent and it’s confusing for users to see those in the
+    documentation just because the GStreamer developer who generated the
+    docs had multiple GPUs to play with at the time. Instead just show
+    the default elements.
 
-Possibly Breaking Changes
+Possibly Breaking and Other Noteworthy Behavioural Changes
+
+-   gst_parse_launch(), gst_parse_bin_from_description() and friends
+    will now error out when setting properties that don’t exist on
+    top-level bins. They were silently ignored before.
+
+-   The GstWebRTC library does not expose any objects anymore with
+    public fields. Instead properties have been added to replace that
+    functionality. If you are accessing such fields in your application,
+    switch to the corresponding properties.
 
--   FIXME: this section will be filled in in due course
--   MPEG-TS SCTE-35 API changes (FIXME: flesh out)
--   gst_parse_launch() and friends now error out on non-existing
-    properties on top-level bins where they would silently fail and
-    ignore those before.
 -   playbin and uridecodebin now emit the source-setup signal before the
     element is added to the bin and linked so that the source element is
     already configured before any scheduling query comes in, which is
-    useful for elements such as appsrc or giostreamsrc. (Merge Request)
+    useful for elements such as appsrc or giostreamsrc.
+
+-   The source element inside urisourcebin (used inside uridecodebin3
+    which is used inside playbin3) is no longer called "source". This
+    shouldn’t affect anyone hopefully, because there’s a "setup-source"
+    signal to configure the source element and no one should rely on
+    names of internal elements anyway.
+
+-   The vp8enc element now expects bps (bits per second) for the
+    "temporal-scalability-target-bitrate" property, which is consistent
+    with the "target-bitrate" property. Since additional configuration
+    is required with modern libvpx to make temporal scaling work anyway,
+    chances are that very few people will have been using this property
+
+-   vp8enc and vp9enc now default to “good quality” for the "deadline"
+    property rather then “best quality”. Having the deadline set to best
+    quality causes the encoder to be absurdly slow, most real-life users
+    will want the good quality tradeoff instead.
+
+-   The experimental GstTranscoder library API in gst-plugins-bad was
+    changed from a GObject signal-based notification mechanism to a
+    GstBus/message-based mechanism akin to GstPlayer/GstPlay.
+
+-   MPEG-TS SCTE-35 API: semantic change for SCTE-35 splice commands:
+    timestamps passed by the application should be in running time now,
+    since users of the API can’t really be expected to predict the local
+    PTS of the muxer.
+
+-   The GstContext used by souphttpsrc to share the session between
+    multiple element instances has changed. Previously it provided
+    direct access to the internal SoupSession object, now it only
+    provides access to an opaque, internal type. This change is
+    necessary because SoupSession is not thread-safe at all and can’t be
+    shared safely between arbitrary external code and souphttpsrc.
+
+-   Python bindings: GObject-introspection related Annotation fixes have
+    led to a case of a GstVideo.VideoInfo-related function signature
+    changing in the Python bindings (possibly one or two other cases
+    too). This is for a function that should never have been exposed in
+    the first place though, so the bindings are being updated to throw
+    an exception in that case, and the correct replacement API has been
+    added in form of an override.
 
 Known Issues
 
--   this section will be filled in in due course
-
--   There are a couple of known WebRTC-related regressions/blockers:
-
-    -   webrtc: DTLS setup with Chrome is broken
-    -   webrtcbin: First keyframe is usually lost
+-   nothing in particular at this point (but also see possibly breaking
+    changes section above)
 
 Contributors
 
--   this section will be filled in in due course
+Aaron Boxer, Adam Leppky, Adam Williamson, Alba Mendez, Alejandro
+González, Aleksandr Slobodeniuk, Alexander Vandenbulcke, Alex Ashley,
+Alicia Boya García, Andika Triwidada, Andoni Morales Alastruey, Andrew
+Wesie, Andrey Moiseev, Antonio Ospite, Antonio Rojas, Arthur Crippa
+Búrigo, Arun Raghavan, Ashley Brighthope, Axel Kellermann, Baek, Bastien
+Nocera, Bastien Reboulet, Benjamin Gaignard, Bing Song, Binh Truong,
+Biswapriyo Nath, Brad Hards, Brad Smith, Brady J. Garvin, Branko
+Subasic, Camilo Celis Guzman, Chris Bass, ChrisDuncanAnyvision, Chris
+White, Corentin Damman, Daniel Almeida, Daniel Knobe, Daniel Stone,
+david, David Fernandez, David Keijser, David Phung, Devarsh Thakkar,
+Dinesh Manajipet, Dmitry Samoylov, Dmitry Shusharin, Dominique Martinet,
+Doug Nazar, Ederson de Souza, Edward Hervey, Emmanuel Gil Peyrot,
+Enrique Ocaña González, Ezequiel Garcia, Fabian Orccon, Fabrice
+Fontaine, Fernando Jimenez Moreno, Florian Karydes, Francisco Javier
+Velázquez-García, François Laignel, Frederich Munch, Fredrik Pålsson,
+George Kiagiadakis, Georg Lippitsch, Göran Jönsson, Guido Günther,
+Guillaume Desmottes, Guiqin Zou, Haakon Sporsheim, Haelwenn (lanodan)
+Monnier, Haihao Xiang, Haihua Hu, Havard Graff, He Junyan, Helmut
+Januschka, Henry Wilkes, Hosang Lee, Hou Qi, Ignacio Casal Quinteiro,
+Igor Kovalenko, Ilya Kreymer, Imanol Fernandez, Jacek Tomaszewski, Jade
+Macho, Jakub Adam, Jakub Janků, Jan Alexander Steffens (heftig), Jan
+Schmidt, Jason Carrete, Jason Pereira, Jay Douglass, Jeongki Kim, Jérôme
+Laheurte, Jimmi Holst Christensen, Johan Sternerup, John Hassell, John
+Lindgren, John-Mark Bell, Jonathan Matthew, Jordan Petridis, Jose
+Quaresma, Julian Bouzas, Julien, Kai Uwe Broulik, Kasper Steensig
+Jensen, Kellermann Axel, Kevin Song, Khem Raj, Knut Inge Hvidsten, Knut
+Saastad, Kristofer Björkström, Lars Lundqvist, Lawrence Troup, Lim Siew
+Hoon, Lucas Stach, Ludvig Rappe, Luis Paulo Fernandes de Barros, Luke
+Yelavich, Mads Buvik Sandvei, Marc Leeman, Marco Felsch, Marek Vasut,
+Marian Cichy, Marijn Suijten, Marius Vlad, Markus Ebner, Mart Raudsepp,
+Matej Knopp, Mathieu Duponchelle, Matthew Waters, Matthieu De Beule,
+Mengkejiergeli Ba, Michael de Gans, Michael Olbrich, Michael Tretter,
+Michal Dzik, Miguel Paris, Mikhail Fludkov, mkba, Nazar Mokrynskyi,
+Nicholas Jackson, Nicola Murino, Nicolas Dufresne, Niklas Hambüchen,
+Nikolay Sivov, Nirbheek Chauhan, Olivier Blin, Olivier Crete, Olivier
+Crête, Paul Goulpié, Per Förlin, Peter Boba, P H, Philippe Normand,
+Philipp Zabel, Pieter Willem Jordaan, Piotrek Brzeziński, Rafał
+Dzięgiel, Rafostar, raghavendra, Raghavendra, Raju Babannavar, Raleigh
+Littles III, Randy Li, Randy Li (ayaka), Ratchanan Srirattanamet, Raul
+Tambre, reed.lawrence, Ricky Tang, Robert Rosengren, Robert Swain, Robin
+Burchell, Roman Sivriver, R S Nikhil Krishna, Ruben Gonzalez, Ruslan
+Khamidullin, Sanchayan Maity, Scott Moreau, Sebastian Dröge, Sergei
+Kovalev, Seungha Yang, Sid Sethupathi, sohwan.park, Sonny Piers, Staz M,
+Stefan Brüns, Stéphane Cerveau, Stephan Hesse, Stian Selnes, Stirling
+Westrup, Théo MAILLART, Thibault Saunier, Tim, Timo Wischer, Tim-Philipp
+Müller, Tim Schneider, Tobias Ronge, Tom Schoonjans, Tulio Beloqui,
+tyler-aicradle, U. Artie Eoff, Ung, Val Doroshchuk, VaL Doroshchuk,
+Víctor Manuel Jáquez Leal, Vivek R, Vivia Nikolaidou, Vivienne
+Watermeier, Vladimir Menshakov, Will Miller, Wim Taymans, Xabier
+Rodriguez Calvar, Xavier Claessens, Xℹ Ruoyao, Yacine Bandou, Yinhang
+Liu, youngh.lee, youngsoo.lee, yychao, Zebediah Figura, Zhang yuankun,
+Zhang Yuankun, Zhao, Zhao Zhili, , Aleksandar Topic, Antonio Ospite,
+Bastien Nocera, Benjamin Gaignard, Brad Hards, Carlos Falgueras García,
+Célestin Marot, Corentin Damman, Corentin Noël, Daniel Almeida, Daniel
+Knobe, Danny Smith, Dave Piché, Dmitry Osipenko, Fabrice Fontaine,
+fjmax, Florian Zwoch, Guillaume Desmottes, Haihua Hu, Heinrich Kruger,
+He Junyan, Jakub Adam, James Cowgill, Jan Alexander Steffens (heftig),
+Jean Felder, Jeongki Kim, Jiri Uncovsky, Joe Todd, Jordan Petridis,
+Krystian Wojtas, Marc-André Lureau, Marcin Kolny, Marc Leeman, Mark
+Nauwelaerts, Martin Reboredo, Mathieu Duponchelle, Matthew Waters,
+Mengkejiergeli Ba, Michael Gruner, Nicolas Dufresne, Nirbheek Chauhan,
+Olivier Crête, Philippe Normand, Rafał Dzięgiel, Ralf Sippl, Robert
+Mader, Sanchayan Maity, Sangchul Lee, Sebastian Dröge, Seungha Yang,
+Stéphane Cerveau, Teh Yule Kim, Thibault Saunier, Thomas Klausner, Timo
+Wischer, Tim-Philipp Müller, Tobias Reineke, Tomasz Andrzejak, Trung Do,
+Tyler Compton, Ung, Víctor Manuel Jáquez Leal, Vivia Nikolaidou, Wim
+Taymans, wngecn, Wonchul Lee, wuchang li, Xavier Claessens, Xi Ruoyao,
+Yoshiharu Hirose, Zhao,
 
 … and many others who have contributed bug reports, translations, sent
 suggestions or helped testing.
@@ -307,7 +1863,7 @@ the git 1.20 branch, which will be a stable branch.
 
 1.20.0
 
-1.20.0 is scheduled to be released around October/November 2021.
+1.20.0 is scheduled to be released around early February 2022.
 
 Schedule for 1.22
 
@@ -315,7 +1871,9 @@ Our next major feature release will be 1.22, and 1.21 will be the
 unstable development version leading up to the stable 1.22 release. The
 development of 1.21/1.22 will happen in the git main branch.
 
-The plan for the 1.22 development cycle is yet to be confirmed.
+The plan for the 1.22 development cycle is yet to be confirmed. Assuming
+no major project-wide reorganisations in the 1.22 cycle we might try and
+aim for a release around August 2022.
 
 1.22 will be backwards-compatible to the stable 1.20, 1.18, 1.16, 1.14,
 1.12, 1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series.
@@ -323,6 +1881,7 @@ The plan for the 1.22 development cycle is yet to be confirmed.
 ------------------------------------------------------------------------
 
 These release notes have been prepared by Tim-Philipp Müller with
-contributions from …
+contributions from Matthew Waters, Nicolas Dufresne, Nirbheek Chauhan,
+Sebastian Dröge and Seungha Yang.
 
 License: CC BY-SA 4.0
diff --git a/subprojects/gstreamer-vaapi/RELEASE b/subprojects/gstreamer-vaapi/RELEASE
index 3a1c57a714..ad28e846d8 100644
--- a/subprojects/gstreamer-vaapi/RELEASE
+++ b/subprojects/gstreamer-vaapi/RELEASE
@@ -1,4 +1,4 @@
-This is GStreamer gstreamer-vaapi 1.19.3.
+This is GStreamer gstreamer-vaapi 1.19.90.
 
 GStreamer 1.19 is the development branch leading up to the next major
 stable version which will be 1.20.
diff --git a/subprojects/gstreamer-vaapi/gstreamer-vaapi.doap b/subprojects/gstreamer-vaapi/gstreamer-vaapi.doap
index 52e9b019e6..69de91328b 100644
--- a/subprojects/gstreamer-vaapi/gstreamer-vaapi.doap
+++ b/subprojects/gstreamer-vaapi/gstreamer-vaapi.doap
@@ -27,6 +27,16 @@
 
  <release>
   <Version>
+   <revision>1.19.90</revision>
+   <branch>main</branch>
+   <name></name>
+   <created>2022-01-28</created>
+   <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gstreamer-vaapi/gstreamer-vaapi-1.19.90.tar.xz" />
+  </Version>
+ </release>
+
+ <release>
+  <Version>
    <revision>1.19.3</revision>
    <branch>main</branch>
    <name></name>
diff --git a/subprojects/gstreamer-vaapi/meson.build b/subprojects/gstreamer-vaapi/meson.build
index 09273b25e1..167a2bb7f3 100644
--- a/subprojects/gstreamer-vaapi/meson.build
+++ b/subprojects/gstreamer-vaapi/meson.build
@@ -1,5 +1,5 @@
 project('gstreamer-vaapi', 'c',
-  version : '1.19.3.1',
+  version : '1.19.90',
   meson_version : '>= 0.59',
   default_options : [ 'warning_level=1',
                       'buildtype=debugoptimized' ])
diff --git a/subprojects/gstreamer/ChangeLog b/subprojects/gstreamer/ChangeLog
index 2d4c2dcfb1..f904bda626 100644
--- a/subprojects/gstreamer/ChangeLog
+++ b/subprojects/gstreamer/ChangeLog
@@ -1,3 +1,18 @@
+=== release 1.19.90 ===
+
+2022-01-28 14:28:35 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* NEWS:
+	* RELEASE:
+	* gstreamer.doap:
+	* meson.build:
+	  Release 1.19.90
+
+2022-01-28 14:28:28 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* ChangeLog:
+	  Update ChangeLogs for 1.19.90
+
 2021-12-23 00:24:03 +0100  Mathieu Duponchelle <mathieu@centricular.com>
 
 	* libs/gst/base/gstaggregator.c:
diff --git a/subprojects/gstreamer/NEWS b/subprojects/gstreamer/NEWS
index e9f5227c15..1a512c1848 100644
--- a/subprojects/gstreamer/NEWS
+++ b/subprojects/gstreamer/NEWS
@@ -1,23 +1,19 @@
 GStreamer 1.20 Release Notes
 
 GStreamer 1.20 has not been released yet. It is scheduled for release in
-November 2021.
+late January / early February 2022.
 
 1.19.x is the unstable development version that is being developed in
-the git main branch and which will eventually result in 1.20, and 1.19.3
-is the current development release in that series
-
-It is expected that feature freeze will be in early October 2021,
-followed by one or two 1.19.9x pre-releases and the new 1.20 stable
-release around October/November 2021.
+the git main branch and which will eventually result in 1.20, and
+1.19.90 is the first release candidate in that series (1.20rc1).
 
 1.20 will be backwards-compatible to the stable 1.18, 1.16, 1.14, 1.12,
-1.10, 1.8, 1.6,, 1.4, 1.2 and 1.0 release series.
+1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series.
 
 See https://gstreamer.freedesktop.org/releases/1.20/ for the latest
 version of this document.
 
-Last updated: Monday 1 November 2021, 01:00 UTC (log)
+Last updated: Wednesday 26 January 2022, 01:00 UTC (log)
 
 Introduction
 
@@ -30,25 +26,971 @@ fixes and other improvements.
 
 Highlights
 
--   this section will be completed in due course
+-   Development in GitLab was switched to a single git repository
+    containing all the modules
+-   GstPlay: new high-level playback library, replaces GstPlayer
+-   WebM Alpha decoding support
+-   Encoding profiles can now be tweaked with additional
+    application-specified element properties
+-   Compositor: multi-threaded video conversion and mixing
+-   RTP header extensions: unified support in RTP depayloader and
+    payloader base classes
+-   SMPTE 2022-1 2-D Forward Error Correction support
+-   Smart encoding (passthrough) support for VP8, VP9, H.265 in
+    encodebin and transcodebin
+-   Runtime compatibility support for libsoup2 and libsoup3 (libsoup3
+    support experimental)
+-   Video decoder subframe support
+-   Video decoder automatic packet-loss, data corruption, and keyframe
+    request handling for RTP / WebRTC / RTSP
+-   MP4 and Matroska muxers now support profile/level/resolution changes
+    for H264/H265 input streams (i.e. codec data changing on the fly)
+-   MP4 muxing mode that initially creates a fragmented mp4 which is
+    converted to a regular mp4 on EOS
+-   Audio support for the WebKit Port for Embedded (WPE) web page source
+    element
+-   CUDA based video color space convert and rescale elements and
+    upload/download elements
+-   NVIDIA memory:NVMM support for OpenGL glupload and gldownload
+    elements
+-   Many WebRTC improvements
+-   The new VA-API plugin implemention fleshed out with more decoders
+    and new postproc elements
+-   AppSink API to retrieve events in addition to buffers and buffer
+    lists
+-   AppSrc gained more configuration options for the internal queue
+    (leakiness, limits in buffers and time, getters to read current
+    levels)
+-   Updated Rust bindings and many new Rust plugins
+-   Improved support for custom minimal GStreamer builds
+-   Support build against FFmpeg 5.0
+-   Linux Stateless CODEC support gained MPEG2 and VP9
+-   Windows Direct3D11/DXVA decoder gained AV1 and MPEG2 support
+-   Lots of new plugins, features, performance improvements and bug
+    fixes
 
 Major new features and changes
 
 Noteworthy new features and API
 
--   this section will be filled in in due course
+-   gst_element_get_request_pad() has been deprecated in favour of the
+    newly-added gst_element_request_pad_simple() which does the exact
+    same thing but has a less confusing name that hopefully makes clear
+    that the function request a new pad rather than just retrieves an
+    already-existing request pad.
+
+Development in GitLab was switched to a single git repository containing all the modules
+
+The GStreamer multimedia framework is a set of libraries and plugins
+split into a number of distinct modules which are released independently
+and which have so far been developed in separate git repositories in
+freedesktop.org GitLab.
+
+In addition to these separate git repositories there was a gst-build
+module that would use the Meson build systems’s subproject feature to
+download each individual module and then build everything in one go. It
+would also provide an uninstalled development environment that made it
+easy to work on GStreamer and use or test versions other than the
+system-installed GStreamer version.
+
+All of these modules have now (as of 28 September 2021) been merged into
+a single git repository (“Mono repository” or “monorepo”) which should
+simplify development workflows and continuous integration, especially
+where changes need to be made to multiple modules at once.
+
+This mono repository merge will primarily affect GStreamer developers
+and contributors and anyone who has workflows based on the GStreamer git
+repositories.
+
+The Rust bindings and Rust plugins modules have not been merged into the
+mono repository at this time because they follow a different release
+cycle.
+
+The mono repository lives in the existing GStreamer core git repository
+in GitLab in the new main branch and all future development will happen
+on this branch.
+
+Modules will continue to be released as separate tarballs.
+
+For more details, please see the GStreamer mono repository FAQ.
+
+GstPlay: new high-level playback library replacing GstPlayer
+
+-   GstPlay is a new high-level playback library that replaces the older
+    GstPlayer API. It is basically the same API as GstPlayer but
+    refactored to use bus messages for application notifications instead
+    of GObject signals. There is still a signal adapter object for those
+    who prefer signals. Since the existing GstPlayer API is already in
+    use in various applications, it didn’t seem like a good idea to
+    break it entirely. Instead a new API was added, and it is expected
+    that this new GstPlay API will be moved to gst-plugins-base in
+    future.
+
+-   The existing GstPlayer API is scheduled for deprecation and will be
+    removed at some point in the future (e.g. in GStreamer 1.24), so
+    application developers are urged to migrate to the new GstPlay API
+    at their earliest convenience.
+
+WebM alpha decoding
+
+-   Implement WebM alpha decoding (VP8/VP9 with alpha), which required
+    support and additions in various places. This is supported both with
+    software decoders and hardware-accelerated decoders.
+
+-   VP8/VP9 don’t support alpha components natively in the codec, so the
+    way this is implemented in WebM is by encoding the alpha plane with
+    transparency data as a separate VP8/VP9 stream. Inside the WebM
+    container (a variant of Matroska) this is coded as a single video
+    track with the “normal” VP8/VP9 video data making up the main video
+    data and each frame of video having an encoded alpha frame attached
+    to it as extra data ("BlockAdditional").
+
+-   matroskademux has been extended extract this per-frame alpha side
+    data and attach it in form of a GstVideoCodecAlphaMeta to the
+    regular video buffers. Note that this new meta is specific to this
+    VP8/VP9 alpha support and can’t be used to just add alpha support to
+    other codecs that don’t support it. Lastly, matroskademux also
+    advertises the fact that the streams contain alpha in the caps.
+
+-   The new codecalpha plugin contains various bits of infrastructure to
+    support autoplugging and debugging:
+
+    -   codecalphademux splits out the alpha stream from the metas on
+        the regular VP8/VP9 buffers
+    -   alphacombine takes two decoded raw video streams (one alpha, one
+        the regular video) and combines it into a video stream with
+        alpha
+    -   vp8alphadecodebin + vp9alphadecodebin are wrapper bins that use
+        the regular vp8dec and vp9dec software decoders to decode
+        regular and alpha streams and combine them again. To decodebin
+        these look like regular decoders which ju
+    -   The V4L2 CODEC plugin has stateless VP8/VP9 decoders that can
+        decode both alpha and non-alpha stream with a single decoder
+        instance
+
+-   A new AV12 video format was added which is basically NV12 with an
+    alpha plane, which is more convenient for many hardware-accelerated
+    decoders.
+
+-   Watch Nicolas Dufresne’s LCA 2022 talk “Bringing WebM Alpha support
+    to GStreamer” for all the details and a demo.
+
+RTP Header Extensions Base Class and Automatic Header Extension Handling in RTP Payloaders and Depayloaders
+
+-   RTP Header Extensions are specified in RFC 5285 and provide a way to
+    add small pieces of data to RTP packets in between the RTP header
+    and the RTP payload. This is often used for per-frame metadata,
+    extended timestamps or other application-specific extra data. There
+    are several commonly-used extensions specified in various RFCs, but
+    senders are free to put any kind of data in there, as long as sender
+    and receiver both know what that data is. Receivers that don’t know
+    about the header extensions will just skip the extra data without
+    ever looking at it. These header extensions can often be combined
+    with any kind of payload format, so may need to be supported by many
+    RTP payloader and depayloader elements.
+
+-   Inserting and extracting RTP header extension data has so far been a
+    bit inconvenient in GStreamer: There are functions to add and
+    retrieve RTP header extension data from RTP packets, but nothing
+    works automatically, even for common extensions. People would have
+    to do the insertion/extraction either in custom elements
+    before/after the RTP payloader/depayloader, or inside pad probes,
+    which isn’t very nice.
+
+-   This release adds various pieces of new infrastructure for generic
+    RTP header extension handling, as well as some implementations for
+    common extensions:
+
+    -   GstRTPHeaderExtension is a new helper base class for reading and
+        writing RTP header extensions. Nominally this subclasses
+        GstElement, but only so these extensions are stored in the
+        registry where they can be looked up by URI or name. They don’t
+        have pads and don’t get added to the pipeline graph as an
+        element.
+
+    -   "add-extension" and "clear-extension" action signals on RTP
+        payloaders and depayloaders for manual extension management
+
+    -   The "request-extension" signal will be emitted if an extension
+        is encountered that requires explicit mapping by the application
+
+    -   new "auto-header-extension" property on RTP payloaders and
+        depayloaders for automatic handling of known header extensions.
+        This is enabled by default. The extensions must be signalled via
+        caps / SDP.
+
+    -   RTP header extension implementations:
+
+        -   rtphdrextclientaudiolevel: Client-to-Mixer Audio Level
+            Indication (RFC 6464) (also see below)
+        -   rtphdrextcolorspace: Color Space extension, extends RTP
+            packets with color space and high dynamic range (HDR)
+            information
+        -   rtphdrexttwcc: Transport Wide Congestion Control support
+
+-   gst_rtp_buffer_remove_extension_data() is a new helper function to
+    remove an RTP header extension from an RTP buffer
+
+-   The existing gst_rtp_buffer_set_extension_data() now also supports
+    shrinking the extension data in size
+
+AppSink and AppSrc improvements
+
+-   appsink: new API to pull events out of appsink in addition to
+    buffers and buffer lists.
+
+    There was previously no way for users to receive incoming events
+    from appsink properly serialised with the data flow, even if they
+    are serialised events. The reason for that is that the only way to
+    intercept events was via a pad probe on the appsink sink pad, but
+    there is also internal queuing inside of appsink, so it’s difficult
+    to ascertain the right order of everything in all cases.
+
+    There is now a new "new-serialized-event" signal which will be
+    emitted when there’s a new event pending (just like the existing
+    "new-sample" signal). The "emit-signals" property must be set to
+    TRUE in order to activate this (but it’s also fine to just pull from
+    the application thread without using the signals).
+
+    gst_app_sink_pull_object() and gst_app_sink_try_pull_object() can be
+    used to pull out either an event or a new sample carrying a buffer
+    or buffer list, whatever is next in the queue.
+
+    EOS events will be filtered and will not be returned. EOS handling
+    can be done the ususal way, same as with _pull_sample().
+
+-   appsrc: allow configuration of internal queue limits in time and
+    buffers and add leaky mode.
+
+    There is internal queuing inside appsrc so the application thread
+    can push data into the element which will then be picked up by the
+    source element’s streaming thread and pushed into the pipeline from
+    that streaming thread. This queue is unlimited by default and until
+    now it was only possible to set a maximum size limit in bytes. When
+    that byte limit is reached, the pushing thread (application thread)
+    would be blocked until more space becomes available.
+
+    A limit in bytes is not particularly useful for many use cases, so
+    now it is possible to also configure limits in time and buffers
+    using the new "max-time" and "max-buffers" properties. Of course
+    there are also matching new read-only"current-level-buffers" and
+    "current-level-time properties" properties to query the current fill
+    level of the internal queue in time and buffers.
+
+    And as if that wasn’t enough the internal queue can also be
+    configured as leaky using the new "leaky-type" property. That way
+    when the queue is full the application thread won’t be blocked when
+    it tries to push in more data, but instead either the new buffer
+    will be dropped or the oldest data in the queue will be dropped.
+
+Better string serialization of nested GstCaps and GstStructures
+
+-   New string serialisation format for structs and caps that can handle
+    nested structs and caps properly by using brackets to delimit nested
+    items (e.g. some-struct, some-field=[nested-struct, nested=true]).
+    Unlike the default format the new variant can also support more than
+    one level of nesting. For backwards-compatibility reasons the old
+    format is still output by default when serialising caps and structs
+    using the existing API. The new functions gst_caps_serialize() and
+    gst_structure_serialize() can be used to output strings in the new
+    format.
+
+Convenience API for custom GstMetas
+
+-   New convenience API to register and create custom GstMetas:
+    gst_meta_register_custom() and gst_buffer_add_custom_meta(). Such
+    custom meta is backed by a GstStructure and does not require that
+    users of the API expose their GstMeta implementation as public API
+    for other components to make use of it. In addition, it provides a
+    simpler interface by ignoring the impl vs. api distinction that the
+    regular API exposes. This new API is meant to be the meta
+    counterpart to custom events and messages, and to be more convenient
+    than the lower-level API when the absolute best performance isn’t a
+    requirement. The reason it’s less performant than a “proper” meta is
+    that a proper meta is just a C struct in the end whereas this goes
+    through the GstStructure API which has a bit more overhead, which
+    for most scenarios is negligible however. This new API is useful for
+    experimentation or proprietary metas, but also has some limitations:
+    it can only be used if there’s a single producer of these metas;
+    it’s not allowed to register the same custom meta multiple times or
+    from multiple places.
+
+Additional Element Properties on Encoding Profiles
+
+-   GstEncodingProfile: The new "element-properties" and
+    gst_encoding_profile_set_element_properties() API allows
+    applications to set additional element properties on encoding
+    profiles to configure muxers and encoders. So far the encoding
+    profile template was the only place where this could be specified,
+    but often what applications want to do is take a ready-made encoding
+    profile shipped by GStreamer or the application and then tweak the
+    settings on top of that, which is now possible with this API. Since
+    applications can’t always know in advance what encoder element will
+    be used in the end, it’s even possible to specify properties on a
+    per-element basis.
+
+    Encoding Profiles are used in the encodebin, transcodebin and
+    camerabin elements and APIs to configure output formats (containers
+    and elementary streams).
+
+Audio Level Indication Meta for RFC 6464
+
+-   New GstAudioLevelMeta containing Audio Level Indication as per RFC
+    6464
+
+-   The level element has been updated to add GstAudioLevelMeta on
+    buffers if the "audio-level-meta" property is set to TRUE. This can
+    then in turn be picked up by RTP payloaders to signal the audio
+    level to receivers through RTP header extensions (see above).
+
+-   New Client-to-Mixer Audio Level Indication (RFC6464) RTP Header
+    Extension which should be automatically created and used by RTP
+    payloaders and depayloaders if their "auto-header-extension"
+    property is enabled and if the extension is part of the RTP caps.
+
+Automatic packet loss, data corruption and keyframe request handling for video decoders
+
+-   The GstVideoDecoder base class has gained various new APIs to
+    automatically handle packet loss and data corruption better by
+    default, especially in RTP, RTSP and WebRTC streaming scenarios, and
+    to give subclasses more control about how they want to handle
+    missing data:
+
+    -   Video decoder subclasses can mark output frames as corrupted via
+        the new GST_VIDEO_CODEC_FRAME_FLAG_CORRUPTED flag
+
+    -   A new "discard-corrupted-frames" property allows applications to
+        configure decoders so that corrupted frames are directly
+        discarded instead of being forwarded inside the pipeline. This
+        is a replacement for the "output-corrupt" property of the FFmpeg
+        decoders.
+
+    -   RTP depayloaders can now signal to decoders that data is missing
+        when sending GAP events for lost packets. GAP events can be sent
+        for various reason in a GStreamer pipeline. Often they are just
+        used to let downstream elements know that there isn’t a buffer
+        available at the moment, so downstream elements can move on
+        instead of waiting for one. They are also sent by RTP
+        depayloaders in the case that packets are missing, however, and
+        so far a decoder was not able to differentiate the two cases.
+        This has been remedied now: GAP events can be decorated with
+        gst_event_set_gap_flags() and GST_GAP_FLAG_MISSING_DATA to let
+        decoders now what happened, and decoders can then use that in
+        some cases to handle missing data better.
+
+    -   The GstVideoDecoder::handle_missing_data vfunc was added to
+        inform subclasses about packet loss or missing data and let them
+        handle it in their own way if they like.
+
+    -   gst_video_decoder_set_needs_sync_point() lets subclasses signal
+        that they need the stream to start with a sync point. If
+        enabled, the base class will discard all non-sync point frames
+        in the beginning and after a flush and does not pass them to the
+        subclass. Furthermore, if the first frame is not a sync point,
+        the base class will try and request a sync frame from upstream
+        by sending a force-key-unit event (see next items).
+
+    -   New "automatic-request-sync-points" and
+        "automatic-request-sync-point-flags" properties to automatically
+        request sync points when needed, e.g. on packet loss or if the
+        first frame is not a keyframe. Applications may want to enable
+        this on decoders operating in e.g. RTP/WebRTC/RTSP receiver
+        pipelines.
+
+    -   The new "min-force-key-unit-interval" property can be used to
+        ensure there’s a minimal interval between keyframe requests to
+        upstream (and/or the sender) and we’re not flooding the sender
+        with key unit requests.
+
+    -   gst_video_decoder_request_sync_point() allows subclasses to
+        request a new sync point (e.g. if they choose to do their own
+        missing data handling). This will still honour the
+        "min-force-key-unit-interval" property if set.
+
+Improved support for custom minimal GStreamer builds
+
+-   Element registration and registration of other plugin features
+    inside plugin init functions has been improved in order to
+    facilitate minimal custom GStreamer builds.
+
+-   A number of new macros have been added to declare and create
+    per-element and per-pluginfeature register functions in all plugins,
+    and then call those from the per-plugin plugin_init functions:
+
+    -   GST_ELEMENT_REGISTER_DEFINE,
+        GST_DEVICE_PROVIDER_REGISTER_DEFINE,
+        GST_DYNAMIC_TYPE_REGISTER_DEFINE, GST_TYPE_FIND_REGISTER_DEFINE
+        for the actual registration call with GStreamer
+    -   GST_ELEMENT_REGISTER, GST_DEVICE_PROVIDER_REGISTER,
+        GST_DYNAMIC_TYPE_REGISTER, GST_PLUGIN_STATIC_REGISTER,
+        GST_TYPE_FIND_REGISTER to call the registration function defined
+        by the REGISTER_DEFINE macro
+    -   GST_ELEMENT_REGISTER_DECLARE,
+        GST_DEVICE_PROVIDER_REGISTER_DECLARE,
+        GST_DYNAMIC_TYPE_REGISTER_DECLARE,
+        GST_TYPE_FIND_REGISTER_DECLARE to declare the registration
+        function defined by the REGISTER_DEFINE macro
+    -   and various variants for advanced use cases.
+
+-   This means that applications can call the per-element and
+    per-pluginfeature registration functions for only the elements they
+    need instead of registering plugins as a whole with all kinds of
+    elements that may not be required (e.g. encoder and decoder instead
+    of just decoder). In case of static linking all unused functions and
+    their dependencies would be removed in this case by the linker,
+    which helps minimise binary size for custom builds.
+
+-   gst_init() will automatically call a gst_init_static_plugins()
+    function if one exists.
+
+-   See the GStreamer static build documentation and Stéphane’s blog
+    post Generate a minimal GStreamer build, tailored to your needs for
+    more details.
 
 New elements
 
--   this section will be filled in in due course
+-   New aesdec and aesenc elements for AES encryption and decryption in
+    a custom format.
+
+-   New encodebin2 element with dynamic/sometimes source pads in order
+    to support the option of doing the muxing outside of encodebin,
+    e.g. in combination with a splitmuxsink.
+
+-   New fakeaudiosink and videocodectestsink elements for testing and
+    debugging (see below for more details)
+
+-   rtpisacpay, rtpisacdepay: new RTP payloader and depayloader for iSAC
+    audio codec
+
+-   rtpst2022-1-fecdec, rtpst2022-1-fecenc: new elements providing SMPTE
+    2022-1 2-D Forward Error Correction. More details in Mathieu’s blog
+    post.
+
+-   isac: new plugin wrapping the Internet Speech Audio Codec reference
+    encoder and decoder from the WebRTC project.
+
+-   asio: plugin for Steinberg ASIO (Audio Streaming Input/Output) API
+
+-   gssrc, gssink: add source and sink for Google Cloud Storage
+
+-   onnx: new plugin to apply ONNX neural network models to video
+
+-   openaptx: aptX and aptX-HD codecs using libopenaptx (v0.2.0)
+
+-   qroverlay, debugqroverlay: new elements that allows overlaying data
+    on top of video in form of a QR code
+
+-   cvtracker: new OpenCV-based tracker element
+
+-   av1parse, vp9parse: new parsers for AV1 and VP9 video
+
+-   va: work on the new VA-API plugin implementation for
+    hardware-accelerated video decoding and encoding has continued at
+    pace, with various new decoders and filters having joined the
+    initial vah264dec:
+
+    -   vah265dec: VA-API H.265 decoder
+    -   vavp8dec: VA-API VP8 decoder
+    -   vavp9dec: VA-API VP9 decoder
+    -   vaav1dec: VA-API AV1 decoder
+    -   vampeg2dec: VA-API MPEG-2 decoder
+    -   vadeinterlace: : VA-API deinterlace filter
+    -   vapostproc: : VA-API postproc filter (color conversion,
+        resizing, cropping, color balance, video rotation, skin tone
+        enhancement, denoise, sharpen)
+
+    See Víctor’s blog post “GstVA in GStreamer 1.20” for more details
+    and what’s coming up next.
+
+-   vaapiav1dec: new AV1 decoder element (in gstreamer-vaapi)
+
+-   msdkav1dec: hardware-accelerated AV1 decoder using the Intel Media
+    SDK / oneVPL
+
+-   nvcodec plugin for NVIDIA NVCODEC API for hardware-accelerated video
+    encoding and decoding:
+
+    -   cudaconvert, cudascale: new CUDA based video color space convert
+        and rescale elements
+    -   cudaupload, cudadownload: new helper elements for memory
+        transfer between CUDA and system memory spaces
+    -   nvvp8sldec, nvvp9sldec: new GstCodecs-based VP8/VP9 decoders
+
+-   Various new hardware-accelerated elements for Windows:
+
+    -   d3d11screencapturesrc: new desktop capture element, including a
+        GstDeviceProvider implementation to enumerate/select target
+        monitors for capture.
+    -   d3d11av1dec and d3d11mpeg2dec: AV1 and MPEG-2 decoders
+    -   d3d11deinterlace: deinterlacing filter
+    -   d3d11compositor: video composing element
+    -   see Windows section below for more details
+
+-   new Rust plugins:
+
+    -   audiornnoise: Removes noise from an audio stream
+    -   awstranscribeparse: Parses AWS audio transcripts into timed text
+        buffers
+    -   ccdetect: Detects if valid closed captions are present in a
+        closed captions stream
+    -   cea608tojson: Converts CEA-608 Closed Captions to a JSON
+        representation
+    -   cmafmux: CMAF fragmented MP4 muxer
+    -   dashmp4mux: DASH fragmented MP4 muxer
+    -   isofmp4mux: ISO fragmented MP4 muxer
+    -   ebur128level: EBU R128 Loudness Level Measurement
+    -   ffv1dec: FFV1 video decoder
+    -   gtk4paintablesink: GTK4 video sink, which provides a
+        GdkPaintable that can be rendered in various widgets
+    -   hlssink3: HTTP Live Streaming sink
+    -   hrtfrender: Head-Related Transfer Function (HRTF) renderer
+    -   hsvdetector: HSV colorspace detector
+    -   hsvfilter: HSV colorspace filter
+    -   jsongstenc: Wraps buffers containing any valid top-level JSON
+        structures into higher level JSON objects, and outputs those as
+        ndjson
+    -   jsongstparse: Parses ndjson as output by jsongstenc
+    -   jsontovtt: converts JSON to WebVTT subtitles
+    -   regex: Applies regular expression operations on text
+    -   roundedcorners: Adds rounded corners to video
+    -   spotifyaudiosrc: Spotify source
+    -   textahead: Display upcoming text buffers ahead (e.g. for
+        Karaoke)
+    -   transcriberbin: passthrough bin that transcribes raw audio to
+        closed captions using awstranscriber and puts the captions as
+        metas onto the video
+    -   tttojson: Converts timed text to a JSON representation
+    -   uriplaylistbin: Playlist source bin
+    -   webpdec-rs: WebP image decoder with animation support
+
+-   New plugin codecalpha with elements to assist with WebM Alpha
+    decoding
+
+    -   codecalphademux: Split stream with GstVideoCodecAlphaMeta into
+        two streams
+    -   alphacombine: Combine two raw video stream (I420 or NV12) as one
+        stream with alpha channel (A420 or AV12)
+    -   vp8alphadecodebin: A bin to handle software decoding of VP8 with
+        alpha
+    -   vp9alphadecodebin: A bin to handle software decoding of VP9 with
+        alpha
+
+-   New hardware accelerated elements for Linux:
+
+    -   v4l2slmpeg2dec: Support for Linux Stateless MPEG2 decoders
+    -   v4l2slvp9dec: Support for Linux Stateless VP9 decoders
+    -   v4l2slvp8alphadecodebin: Support HW accelerated VP8 with alpha
+        layer decoding
+    -   v4l2slvp9alphadecodebin: Support HW accelerated VP9 with alpha
+        layer decoding
 
 New element features and additions
 
--   this section will be filled in in due course
+-   assrender: handle more font mime types; better interaction with
+    matroskademux for embedded fonts
+
+-   audiobuffersplit: Add support for specifying output buffer size in
+    bytes (not just duration)
+
+-   audiolatency: new "samplesperbuffer" property so users can configure
+    the number of samples per buffer. The default value is 240 samples
+    which is equivalent to 5ms latency with a sample rate of 48000,
+    which might be larger than actual buffer size of audio capture
+    device.
+
+-   audiomixer, audiointerleave, GstAudioAggregator: now keep a count of
+    samples that are dropped or processed as statistic and can be made
+    to post QoS messages on the bus whenever samples are dropped by
+    setting the "qos-messages" property on input pads.
+
+-   audiomixer, compositor: improved handling of new inputs added at
+    runtime. New API was added to the GstAggregator base class to allow
+    subclasses to opt into an aggregation mode where inactive pads are
+    ignored when processing input buffers
+    (gst_aggregator_set_ignore_inactive_pads(),
+    gst_aggregator_pad_is_inactive()). An “inactive pad” in this context
+    is a pad which, in live mode, hasn’t yet received a first buffer,
+    but has been waited on at least once. What would happen usually in
+    this case is that the aggregator would wait for data on this pad
+    every time, up to the maximum configured latency. This would
+    inadvertently push mixer elements in live mode to the configured
+    latency envelope and delay processing when new inputs are added at
+    runtime until these inputs have actually produced data. This is
+    usually undesirable. With this new API, new inputs can be added
+    (requested) and configured and they won’t delay the data processing.
+    Applications can opt into this new behaviour by setting the
+    "ignore-inactive-pads" property on compositor, audiomixer or other
+    GstAudioAggregator-based elements.
+
+-   cccombiner: implement “scheduling” of captions. So far cccombiner’s
+    behaviour was essentially that of a funnel: it strictly looked at
+    input timestamps to associate together video and caption buffers.
+    Now it will try to smoothly schedule caption buffers in order to
+    have exactly one per output video buffer. This might involve
+    rewriting input captions, for example when the input is CDP then
+    sequence counters are rewritten, time codes are dropped and
+    potentially re-injected if the input video frame had a time code
+    meta. This can also lead to the input drifting from synchronization,
+    when there isn’t enough padding in the input stream to catch up. In
+    that case the element will start dropping old caption buffers once
+    the number of buffers in its internal queue reaches a certain limit
+    (configurable via the "max-scheduled" property). The new original
+    funnel-like behaviour can be restored by setting the "scheduling"
+    property to FALSE.
+
+-   ccconverter: new "cdp-mode" property to specify which sections to
+    include in CDP packets (timecode, CC data, service info). Various
+    software, including ffmpeg’s Decklink support, fails parsing CDP
+    packets that contain anything but CC data in the CDP packets.
+
+-   clocksync: new "sync-to-first" property for automatic timestamp
+    offset setup: if set clocksync will set up the "ts-offset" value
+    based on the first buffer and the pipeline’s running time when the
+    first buffer arrived. The newly configured "ts-offset" in this case
+    would be the value that allows outputting the first buffer without
+    waiting on the clock. This is useful for example to feed a non-live
+    input into an already-running pipeline.
+
+-   compositor:
+
+    -   multi-threaded input conversion and compositing. Set the
+        "max-threads" property to activate this.
+    -   new "sizing-policy" property to support display aspect ratio
+        (DAR)-aware scaling. By default the image is scaled to fill the
+        configured destination rectangle without padding and without
+        keeping the aspect ratio. With sizing-policy=keep-aspect-ratio
+        the input image is scaled to fit the destination rectangle
+        specified by GstCompositorPad:{xpos, ypos, width, height}
+        properties preserving the aspect ratio. As a result, the image
+        will be centered in the destination rectangle with padding if
+        necessary.
+    -   new "zero-size-is-unscaled" property on input pads. By default
+        pad width=0 or pad height=0 mean that the stream should not be
+        scaled in that dimension. But if the "zero-size-is-unscaled"
+        property is set to FALSE a width or height of 0 is instead
+        interpreted to mean that the input image on that pad should not
+        be composited, which is useful when creating animations where an
+        input image is made smaller and smaller until it disappears.
+    -   improved handling of new inputs at runtime via
+        "ignore-inactive-pads"property (see above for details)
+    -   allow output format with alpha even if none of the inputs have
+        alpha (also glvideomixer and other GstVideoAggregator
+        subclasses)
+
+-   dashsink: add h265 codec support and signals for allowing custom
+    playlist/fragment output
+
+-   decodebin3:
+
+    -   improved decoder selection, especially for hardware decoders
+    -   make input activation “atomic” when adding inputs dynamically
+    -   better interleave handling: take into account decoder latency
+        for interleave size
+
+-   decklink:
+
+    -   Updated DeckLink SDK to 11.2 to support DeckLink 8K Pro
+    -   decklinkvideosrc:
+        -   More accurate and stable capture timestamps: use the
+            hardware reference clock time when the frame was finished
+            being captured instead of a clock time much further down the
+            road.
+        -   Automatically detect widescreen vs. normal NTSC/PAL
+
+-   encodebin:
+
+    -   add “smart encoding” support for H.265, VP8 and VP9 (i.e. only
+        re-encode where needed and otherwise pass through encoded video
+        as-is).
+    -   H264/H265 smart encoding improvements: respect user-specified
+        stream-format, but if not specified default to avc3/hvc1 with
+        in-band SPS/PPS/VPS signalling for more flexibility.
+    -   new encodebin2 element with dynamic/sometimes source pads in
+        order to support the option of doing the muxing outside of
+        encodebin, e.g. in combination with splitmuxsink.
+    -   add APIs to set element properties on encoding profiles (see
+        below)
+
+-   errorignore: new "ignore-eos" property to also ignore FLOW_EOS from
+    downstream elements
+
+-   giosrc: add support for growing source files: applications can
+    specify that the underlying file being read is growing by setting
+    the "is-growing" property. If set, the source won’t EOS when it
+    reaches the end of the file, but will instead start monitoring it
+    and will start reading data again whenever a change is detected. The
+    new "waiting-data" and "done-waiting-data" signals keep the
+    application informed about the current state.
+
+-   gtksink, gtkglsink:
+
+    -   scroll event support: forwarded as navigation events into the
+        pipeline
+    -   "video-aspect-ratio-override" property to force a specific
+        aspect ratio
+    -   "rotate-method" property and support automatic rotation based on
+        image tags
+
+-   identity: new "stats" property allows applications to retrieve the
+    number of bytes and buffers that have passed through so far.
+
+-   interlace: add support for more formats, esp 10-bit, 12-bit and
+    16-bit ones
+
+-   jack: new "low-latency" property for automatic latency-optimized
+    setting and "port-names" property to select ports explicitly
+
+-   jpegdec: support output conversion to RGB using libjpeg-turbo (for
+    certain input files)
+
+-   line21dec:
+
+    -   "mode" property to control whether and how detected closed
+        captions should be inserted in the list of existing close
+        caption metas on the input frame (if any): add, drop, or
+        replace.
+    -   "ntsc-only" property to only look for captions if video has NTSC
+        resolution
+
+-   line21enc: new "remove-caption-meta" to remove metas from output
+    buffers after encoding the captions into the video data; support for
+    CDP closed captions
+
+-   matroskademux, matroskamux: Add support for ffv1, a lossless
+    intra-frame video coding format.
+
+-   matroskamux: accept in-band SPS/PPS/VPS for H264 and H265
+    (i.e. stream-format avc3 and hev1) which allows on-the-fly
+    profile/level/resolution changes.
+
+-   matroskamux: new "cluster-timestamp-offset" property, useful for use
+    cases where the container timestamps should map to some absolute
+    wall clock time, for example.
+
+-   rtpsrc: add "caps" property to allow explicit setting of the caps
+    where needed
+
+-   mpegts: support SCTE-35 passthrough via new "send-scte35-events"
+    property on MPEG-TS demuxer tsdemux. When enabled, SCTE 35 sections
+    (eg ad placement opportunities) are forwarded as events donwstream
+    where they can be picked up again by mpegtsmux. This required a
+    semantic change in the SCTE-35 section API: timestamps are now in
+    running time instead of muxer pts.
+
+-   tsdemux: Handle PCR-less MPEG-TS streams; more robust timestamp
+    handling in certain corner cases and for poorly muxed streams.
+
+-   mpegtsmux:
+
+    -   More conformance improvements to make MPEG-TS analyzers happy:
+        -   PCR timing accuracy: Improvements to the way mpegtsmux
+            outputs PCR observations in CBR mode, so that a PCR
+            observation is always inserted when needed, so that we never
+            miss the configured pcr-interval, as that triggers various
+            MPEG-TS analyser errors.
+        -   Improved PCR/SI scheduling
+    -   Don’t write PCR until PAT/PMT are output to make sure streams
+        start cleanly with a PAT/PMT.
+    -   Allow overriding the automatic PMT PID selection via
+        application-supplied PMT_%d fields in the prog-map
+        structure/property.
+
+-   mp4mux:
+
+    -   new "first-moov-then-finalise" mode for fragmented output where
+        the output will start with a self-contained moov atom for the
+        first fragment, and then produce regular fragments. Then at the
+        end when the file is finalised, the initial moov is invalidated
+        and a new moov is written covering the entire file. This way the
+        file is a “fragmented mp4” file while it is still being written
+        out, and remains playable at all times, but at the end it is
+        turned into a regular mp4 file (with former fragment headers
+        remaining as unused junk data in the file).
+    -   support H.264 avc3 and H.265 hvc1 stream formats as input where
+        the codec data is signalled in-band inside the bitstream instead
+        of caps/file headers.
+    -   support profile/level/resolution changes for H264/H265 input
+        streams (i.e. codec data changing on the fly). Each codec_data
+        is put into its own SampleTableEntry inside the stsd, unless the
+        input is in avc3 stream format in which case it’s written
+        in-band an not in the headers.
+
+-   multifilesink: new ""min-keyframe-distance"" property to make
+    minimum distance between keyframes in next-file=key-frame mode
+    configurable instead of hard-coding it to 10 seconds.
+
+-   mxfdemux has seen a big refactoring to support non-frame wrappings
+    and more accurate timestamp/seek handling for some formats
+
+-   msdk plugin for hardware-accelerated video encoding and decoding
+    using the Intel Media SDK:
+
+    -   oneVPL support (Intel oneAPI Video Processing Library)
+    -   AV1 decoding support
+    -   H.264 decoder now supports constrained-high and progressive-high
+        profiles
+    -   H.264 encoder:
+        -   more configuration options (properties):
+            "intra-refresh-type", "min-qp" , "max-qp", "p-pyramid",
+            "dblk-idc"
+    -   H.265 encoder:
+        -   can output main-still-picture profile
+        -   now inserts HDR SEIs (mastering display colour volume and
+            content light level)
+        -   more configuration options (properties):
+            "intra-refresh-type", "min-qp" , "max-qp", "p-pyramid",
+            "b-pyramid", "dblk-idc", "transform-skip"
+        -   support for RGB 10bit format
+    -   External bitrate control in encoders
+    -   Video post proc element msdkvpp gained support for 12-bit pixel
+        formats P012_LE, Y212_LE and Y412_LE
+
+-   nvh264sldec: interlaced stream support
+
+-   openh264enc: support main, high, constrained-high and
+    progressive-high profiles
+
+-   openjpeg: support for multithreaded decoding and encoding
+
+-   rtspsrc: now supports IPv6 also for tunneled mode (RTSP-over-HTTP);
+    new "ignore-x-server-reply" property to ignore the
+    x-server-ip-address server header reply in case of HTTP tunneling,
+    as it is often broken.
+
+-   souphttpsrc: Runtime compatibility support for libsoup2 and
+    libsoup3. libsoup3 is the latest major version of libsoup, but
+    libsoup2 and libsoup3 can’t co-exist in the same process because
+    there is no namespacing or versioning for GObject types. As a
+    result, it would be awkward if the GStreamer souphttpsrc plugin
+    linked to a specific version of libsoup, because it would only work
+    with applications that use the same version of libsoup. To make this
+    work, the soup plugin now tries to determine the libsoup version
+    used by the application (and its other dependencies) at runtime on
+    systems where GStreamer is linked dynamically. libsoup3 support is
+    still considered somewhat experimental at this point.
+
+-   srtsrc, srtsink: add signals for the application to accept/reject
+    incoming connections
+
+-   timeoverlay: new elapsed-running-time time mode which shows the
+    running time since the first running time (and each flush-stop).
+
+-   udpsrc: new timestamping mode to retrieve packet receive timestamps
+    from the kernel via socket control messages (SO_TIMESTAMPNS) on
+    supported platforms
+
+-   uritranscodebin: new setup-source and element-setup signals for
+    applications to configure elements used
+
+-   v4l2codecs plugin gained support for 4x4 and 32x32 tile formats
+    enabling some platforms or direct renders. Important memory usage
+    improvement.
+
+-   v4l2slh264dec now implements the final Linux uAPI as shipped on
+    Linux 5.11 and later.
+
+-   valve: add "drop-mode" property and provide two new modes of
+    operation: in drop-mode=forward-sticky-events sticky events
+    (stream-start, segment, tags, caps, etc.) are forwarded downstream
+    even when dropping is enabled; drop-mode=transform-to-gap will in
+    addition also convert buffers into gap events when dropping is
+    enabled, which lets downstream elements know that time is advancing
+    and might allow for preroll in many scenarios. By default all events
+    and all buffers are dropped when dropping is enabled, which can
+    cause problems with caps negotiation not progressing or branches not
+    prerolling when dropping is enabled.
+
+-   videocrop: support for many more pixel formats, e.g. planar YUV
+    formats with > 8bits and GBR* video formats; can now also accept
+    video not backed by system memory as long as downstream supports the
+    GstCropMeta
+
+-   videotestsrc: new smpte-rp-219 pattern for SMPTE75 RP-219 conformant
+    color bars
+
+-   vp8enc: finish support for temporal scalability: two new properties
+    ("temporal-scalability-layer-flags",
+    "temporal-scalability-layer-sync-flags") and a unit change on the
+    "temporal-scalability-target-bitrate" property (now expects bps);
+    also make temporal scalability details available to RTP payloaders
+    as buffer metadata.
+
+-   vp9enc: new properties to tweak encoder performance:
+
+    -   "aq-mode" to configure adaptive quantization modes
+    -   "frame-parallel-decoding" to configure whether to create a
+        bitstream that reduces decoding dependencies between frames
+        which allows staged parallel processing of more than one video
+        frames in the decoder. (Defaults to TRUE)
+    -   "row-mt", "tile-columns" and "tile-rows" so multithreading can
+        be enabled on a per-tile basis, instead of on a per tile-column
+        basis. In combination with the new "tile-rows" property, this
+        allows the encoder to make much better use of the available CPU
+        power.
+
+-   vp9dec, vp9enc: add support for 10-bit 4:2:0 and 4:2:2 YUV, as well
+    as 8-bit 4:4:4
+
+-   vp8enc, vp9enc now default to “good quality” for the deadline
+    property rather then “best quality”. Having the deadline set to best
+    quality causes the encoder to be absurdly slow, most real-life users
+    will prefer good-enough quality with better performance instead.
+
+-   wpesrc:
+
+    -   implement audio support: a new sometimes source pad will be
+        created for each audio stream created by the web engine.
+    -   move wpesrc to wpevideosrc and add a wrapper bin wpesrc to also
+        support audio
+    -   also handles web:// URIs now (same as cefsrc)
+    -   post messages with the estimated load progress on the bus
 
-Plugin and library moves
+-   x265enc: add negative DTS support, which means timestamps are now
+    offset by 1h same as with x264enc
+
+RTP Payloaders and Depayloaders
+
+-   rtpisacpay, rtpisacdepay: new RTP payloader and depayloader for iSAC
+    audio codec
+
+-   rtph264depay:
+
+    -   new "request-keyframe" property to make the depayloader
+        automatically request a new keyframe from the sender on packet
+        loss, consistent with the new property on rtpvp8depay.
+    -   new "wait-for-keyframe" property to make depayloader wait for a
+        new keyframe at the beginning and after packet loss (only
+        effective if the depayloader outputs AUs), consistent with the
+        existing property on rtpvp8depay.
+
+-   rtpopuspay, rtpopusdepay: support libwebrtc-compatible multichannel
+    audio in addition to the previously supported multichannel audio
+    modes
+
+-   rtpopuspay: add DTX (Discontinuous Transmission) support
+
+-   rtpvp8depay: new "request-keyframe" property to make the depayloader
+    automatically request a new keyframe from the sender on packet loss.
+
+-   rtpvp8pay: temporal scaling support
 
--   this section will be filled in in due course
+-   rtpvp9depay: Improved SVC handling (aggregate all layers)
+
+RTP Infrastructure
+
+-   rtpst2022-1-fecdec, rtpst2022-1-fecenc: new elements providing SMPTE
+    2022-1 2-D Forward Error Correction. More details in Mathieu’s blog
+    post.
+
+-   rtpreddec: BUNDLE support
+
+-   rtpredenc, rtpulpfecenc: add support for Transport-wide Congestion
+    Control (TWCC)
+
+-   rtpsession: new "twcc-feedback-interval" property to allow RTCP TWCC
+    reports to be scheduled on a timer instead of per marker-bit.
+
+Plugin and library moves
 
 -   There were no plugin moves or library moves in this cycle.
 
@@ -56,8 +998,6 @@ Plugin removals
 
 The following elements or plugins have been removed:
 
--   this section will be filled in in due course
-
 -   The ofa audio fingerprinting plugin has been removed. The MusicIP
     database has been defunct for years so this plugin is likely neither
     useful nor used by anyone.
@@ -71,135 +1011,575 @@ The following elements or plugins have been removed:
 
 Miscellaneous API additions
 
--   this section will be filled in in due course
+Core
+
+-   gst_buffer_new_memdup() is a convenience function for the
+    widely-used gst_buffer_new_wrapped(g_memdup(data,size),size)
+    pattern.
+
+-   gst_caps_features_new_single() creates a new single GstCapsFeatures,
+    avoiding the need to use the vararg function with NULL terminator
+    for simple cases.
+
+-   gst_element_type_set_skip_documentation() can be used by plugins to
+    signal that certain elements should not be included in the GStreamer
+    plugin documentation. This is useful for plugins where elements are
+    registered dynamically based on hardware capabilities and/or where
+    the available plugins and properties vary from system to system.
+    This is used in the d3d11 plugin for example to ensure that only the
+    list of default elements is advertised in the documentation.
+
+-   gst_type_find_suggest_empty_simple() is a new convenience function
+    for typefinders for cases where there’s only a media type and no
+    other fields.
+
+-   New API to create elements and set properties at construction time,
+    which is not only convenient, but also allows GStreamer elements to
+    have construct-only properties: gst_element_factory_make_full(),
+    gst_element_factory_make_valist(),
+    gst_element_factory_make_with_properties(),
+    gst_element_factory_create_full(),
+    gst_element_factory_create_valist(),
+    gst_element_factory_create_with_properties().
+
+-   GstSharedTaskPool: new “shared” task pool subclass with slightly
+    different default behaviour than the existing GstTaskPool which
+    would create unlimited number of threads for new tasks. The shared
+    taskpool creates up to N threads (default: 1) and then distributes
+    pending tasks to those threads round-robin style, and blocks if no
+    thread is available. It is possible to join tasks. This can be used
+    by plugins to implement simple multi-threaded processing and is used
+    for the new multi-threaded video conversion and compositing done in
+    GstVideoAggregator, videoconverter and compositor.
+
+Plugins Base Utils library
+
+-   GstDiscoverer:
+
+    -   gst_discoverer_container_info_get_tags() was added to retrieve
+        global/container tags (vs. per-stream tags). Per-Stream tags can
+        be retrieved via the existing
+        gst_discoverer_stream_info_get_tags().
+        gst_discoverer_info_get_tags(), which for many files returns a
+        confusing mix of stream and container tags, has been deprecated
+        in favour of the container/stream-specific functions.
+    -   gst_discoverer_stream_info_get_stream_number() returns a unique
+        integer identifier for a given stream within the given
+        GstDiscoverer context. (If this matches the stream number inside
+        the container bitstream that’s by coincidence and not by
+        design.)
+
+-   gst_pb_utils_get_caps_description_flags() can be used to query
+    whether certain caps represent a container, audio, video, image,
+    subtitles, tags, or something else. This only works for formats
+    known to GStreamer.
+
+-   gst_pb_utils_get_file_extension_from_caps() returns a possible file
+    extension for given caps.
+
+-   gst_codec_utils_h264_get_profile_flags_level(): Parses profile,
+    flags, and level from H264 AvcC codec_data. The format of H264 AVCC
+    extradata/sequence_header is documented in the ITU-T H.264
+    specification section 7.3.2.1.1 as well as in ISO/IEC 14496-15
+    section 5.3.3.1.2.
+
+-   gst_codec_utils_caps_get_mime_codec() to convert caps to a RFC 6381
+    compatible MIME codec string codec. Useful for providing the codecs
+    field inside the Content-Type HTTP header for containerized formats,
+    such as mp4 or matroska.
+
+GStreamer OpenGL integration library and plugins
+
+-   glcolorconvert: added suppport for converting the video formats
+    A420, AV12, BGR, BGRA, RGBP and BGRP.
+
+-   Added support to GstGLBuffer for persistent buffer mappings where a
+    Pixel Buffer Object (PBO) can be mapped by both the CPU and the GPU.
+    This removes a memcpy() when uploading textures or vertices
+    particularly when software decoders (e.g. libav) are direct
+    rendering into our memory. Improves transfer performance
+    significantly. Requires OpenGL 4.4, GL_ARB_buffer_storage or
+    GL_EXT_buffer_storage
+
+-   Added various helper functions for handling 4x4 matrices of affine
+    transformations as used by GstVideoAffineTransformationMeta.
+
+-   Add support to GstGLContext for allowing the application to control
+    the config (EGLConfig, GLXConfig, etc) used when creating the OpenGL
+    context. This allows the ability to choose between RGB16 or RGB10A2
+    or RGBA8 back/front buffer configurations that were previously
+    hardcoded. GstGLContext also supports retrieving the configuration
+    it was created with or from an externally provide OpenGL context
+    handle. This infrastructure is also used to create a compatible
+    config from an application/externally provided OpenGL context in
+    order to improve compatibility with other OpenGL frameworks and GUI
+    toolkits. A new environment variable GST_GL_CONFIG was also added to
+    be able to request a specific configuration from the command line.
+    Note: different platforms will have different functionality
+    available.
+
+-   Add support for choosing between EGL and WGL at runtime when running
+    on Windows. Previously this was a build-time switch. Allows use in
+    e.g. Gtk applications on Windows that target EGL/ANGLE without
+    recompiling GStreamer. gst_gl_display_new_with_type() can be used by
+    applications to choose a specific display type to use.
+
+-   Build fixes to explicitly check for Broadcom-specific libraries on
+    older versions of the Raspberry Pi platform. The Broadcom OpenGL ES
+    and EGL libraries have different filenames. Using the vc4 Mesa
+    driver on the Raspberry Pi is not affected.
+
+-   Added support to glupload and gldownload for transferring RGBA
+    buffers using the memory:NVMM available on the Nvidia Tegra family
+    of embedded devices.
+
+-   Added support for choosing libOpenGL and libGLX as used in a GLVND
+    environment on unix-based platforms. This allows using desktop
+    OpenGL and EGL without pulling in any GLX symbols as would be
+    required with libGL.
+
+Video library
+
+-   New raw video formats:
+
+    -   AV12 (NV12 with alpha plane)
+    -   RGBP and BGRP (planar RGB formats)
+    -   ARGB64 variants with specified endianness instead of host
+        endianness:
+        -   ARGB64_LE, ARGB64_BE
+        -   RGBA64_BE, RGBA64_LE
+        -   BGRA64_BE, BGRA64_LE
+        -   ABGR64_BE, ABGR64_LE
+
+-   gst_video_orientation_from_tag() is new convenience API to parse the
+    image orientation from a GstTagList.
+
+-   GstVideoDecoder subframe support (see below)
+
+-   GstVideoCodecState now also carries some HDR metadata
+
+-   Ancillary video data: implement transform functions for AFD/Bar
+    metas, so they will be forwarded in more cases
+
+MPEG-TS library
+
+This library only handles section parsing and such, see above for
+changes to the actual mpegtsmux and mpegtsdemux elements.
+
+-   many additions and improvements to SCTE-35 section parsing
+-   new API for fetching extended descriptors:
+    gst_mpegts_find_descriptor_with_extension()
+-   add support for SIT sections (Selection Information Tables)
+-   expose event-from-section constructor gst_event_new_mpegts_section()
+-   parse Audio Preselection Descriptor needed for Dolby AC-4
+
+GstWebRTC library + webrtcbin
+
+-   Change the way in which sink pads and transceivers are matched
+    together to support easier usage. If a pad is created without a
+    specific index (i.e. using sink_%u as the pad template), then an
+    available compatible transceiver will be searched for. If a specific
+    index is requested (i.e. sink_1) then if a transceiver for that
+    m-line already exists, that transceiver must match the new sink pad
+    request. If there is no transceiver available in either scenario, a
+    new transceiver is created. If a mixture of both sink_1 and sink_%u
+    requests result in an impossible situation, an error will be
+    produced at pad request time or from create offer/answer.
+
+-   webrtcbin now uses regular ICE nomination instead of libnice’s
+    default of aggressive ICE nomination. Regular ICE nomination is the
+    default recommended by various relevant standards and improves
+    connectivity in specific network scenarios.
+
+-   Add support for limiting the port range used for RTP with the
+    addition of the min-rtp-port and max-rtp-port properties on the ICE
+    object.
+
+-   Expose the SCTP transport as a property on webrtcbin to more closely
+    match the WebRTC specification.
+
+-   Added support for taking into account the data channel transport
+    state when determining the value of the "connection-state" property.
+    Previous versions of the WebRTC spec did not include the data
+    channel state when computing this value.
+
+-   Add configuration for choosing the size of the underlying sockets
+    used for transporting media data
+
+-   Always advertise support for the transport-cc RTCP feedback protocol
+    as rtpbin supports it. For full support, the configured caps (input
+    or through codec-preferences) need to include the relevant RTP
+    header extension.
+
+-   Numerous fixes to caps and media handling to fail-fast when an
+    incompatible situation is detected.
+
+-   Improved support for attaching the required media after a remote
+    offer has been set.
+
+-   Add support for dynamically changing the amount of FEC used for a
+    particular stream.
+
+-   webrtcbin now stops further SDP processing at the first error it
+    encounters.
+
+-   Completed support for either local or the remote closing a data
+    channel.
+
+-   Various fixes when performing BUNDLEing of the media streams in
+    relation to RTX and FEC usage.
+
+-   Add support for writing out QoS DSCP marking on outgoing packets to
+    improve reliability in some network scenarios.
+
+-   Improvements to the statistics returned by the get-stats signal
+    including the addition of the raw statistics from the internal
+    RTPSource, the TWCC stats when available.
+
+-   The webrtc library does not expose any objects anymore with public
+    fields. Instead properties have been added to replace that
+    functionality. If you are accessing such fields in your application,
+    switch to the corresponding properties.
+
+GstCodecs and Video Parsers
+
+-   Support for render delays to improve throughput across all CODECs
+    (used with NVDEC and V4L2).
+-   lots of improvements to parsers and the codec parsing decoder base
+    classes (H264, H265, VP8, VP9, AV1, MPEG-2) used for various
+    hardware-accelerate decoder APIs.
+
+Bindings support
+
+-   gst_allocation_params_new() allocates a GstAllocationParams struct
+    on the heap. This should only be used by bindings (and freed via
+    gst_allocation_params_free() then). In C code you would allocate
+    this on the stack and only init it in place.
+
+-   gst_debug_log_literal() can be used to log a string to the debug log
+    without going through any printf format expansion and associated
+    overhead. This is mostly useful for bindings such as the Rust
+    bindings which may have done their own formatting already .
+
+-   Provide non-inlined versions of refcounting APIs for various
+    GStreamer mini objects, so that they can be consumed by bindings
+    (e.g. gstreamer-sharp): gst_buffer_ref, gst_buffer_unref,
+    gst_clear_buffer, gst_buffer_copy, gst_buffer_replace,
+    gst_buffer_list_ref, gst_buffer_list_unref, gst_clear_buffer_list,
+    gst_buffer_list_copy, gst_buffer_list_replace, gst_buffer_list_take,
+    gst_caps_ref, gst_caps_unref, gst_clear_caps, gst_caps_replace,
+    gst_caps_take, gst_context_ref, gst_context_unref, gst_context_copy,
+    gst_context_replace, gst_event_replace, gst_event_steal,
+    gst_event_take, gst_event_ref, gst_event_unref, gst_clear_event,
+    gst_event_copy, gst_memory_ref, gst_memory_unref, gst_message_ref,
+    gst_message_unref, gst_clear_message, gst_message_copy,
+    gst_message_replace, gst_message_take, gst_promise_ref,
+    gst_promise_unref, gst_query_ref, gst_query_unref, gst_clear_query,
+    gst_query_copy, gst_query_replace, gst_query_take, gst_sample_ref,
+    gst_sample_unref, gst_sample_copy, gst_tag_list_ref,
+    gst_tag_list_unref, gst_clear_tag_list, gst_tag_list_replace,
+    gst_tag_list_take, gst_uri_copy, gst_uri_ref, gst_uri_unref,
+    gst_clear_uri.
+
+-   expose a GType for GstMiniObject
+
+-   gst_device_provider_probe() now returns non-floating device object
+
+API Deprecations
+
+-   gst_element_get_request_pad() has been deprecated in favour of the
+    newly-added gst_element_request_pad_simple() which does the exact
+    same thing but has a less confusing name that hopefully makes clear
+    that the function request a new pad rather than just retrieves an
+    already-existing request pad.
+
+-   gst_discoverer_info_get_tags(), which for many files returns a
+    confusing mix of stream and container tags, has been deprecated in
+    favour of the container-specific and stream-specific functions,
+    gst_discoverer_container_info_get_tags() and
+    gst_discoverer_stream_info_get_tags().
+
+-   gst_video_sink_center_rect() was deprecated in favour of the more
+    generic newly-added gst_video_center_rect().
+
+-   The GST_MEMORY_FLAG_NO_SHARE flag has been deprecated, as it tends
+    to cause problems and prevents sub-buffering. If pooling or lifetime
+    tracking is required, memories should be allocated through a custom
+    GstAllocator instead of relying on the lifetime of the buffers the
+    memories were originally attached to, which is fragile anyway.
+
+-   The GstPlayer high-level playback library is being replaced with the
+    new GstPlay library (see above). GstPlayer should be considered
+    deprecated at this point and will be marked as such in the next
+    development cycle. Applications should be ported to GstPlay.
+
+-   Gstreamer Editing Services: ges_video_transition_set_border(),
+    ges_video_transition_get_border()
+    ges_video_transition_set_inverted()
+    ges_video_transition_is_inverted() have been deprecated, use
+    ges_timeline_element_set_children_properties() instead.
 
 Miscellaneous performance, latency and memory optimisations
 
--   this section will be filled in in due course
+More video conversion fast paths
 
-Miscellaneous other changes and enhancements
+-   v210 ↔ I420, YV12, Y42B, UYVY and YUY2
+-   A420 → RGB
+
+Less jitter when waiting on the system clock
+
+-   Better system clock wait accuracy, less jitter: where available,
+    clock_nanosleep is used for higher accuracy for waits below 500
+    usecs, and waits below 2ms will first use the regular waiting system
+    and then clock_nanosleep for the remainder. The various wait
+    implementation have a latency ranging from 50 to 500+ microseconds.
+    While this is not a major issue when dealing with a low number of
+    waits per second (for ex: video), it does introduce a non-negligible
+    jitter for synchronisation of higher packet rate systems.
+
+Video decoder subframe support
 
--   this section will be filled in in due course
+-   The GstVideoDecoder base class gained API to process input at the
+    sub-frame level. That way video decoders can start decoding slices
+    before they have received the full input frame in its entirety (to
+    the extent this is supported by the codec, of course). This helps
+    with CPU utilisation and reduces latency.
 
-Tracing framework and debugging improvements
+-   This functionality is now being used in the OpenJPEG JPEG 2000
+    decoder, the FFmpeg H.264 decoder (in case of NAL-aligned input) and
+    the OpenMAX H.264/H.265 decoders (in case of NAL-aligned input).
 
--   this section will be filled in in due course
+Miscellaneous other changes and enhancements
+
+-   GstDeviceMonitor no longer fails to start just because one of the
+    device providers failed to start. That could happen for example on
+    systems where the pulseaudio device provider is installed, but
+    pulseaudio isn’t actually running but ALSA is used for audio
+    instead. In the same vein the device monitor now keeps track of
+    which providers have been started (via the new
+    gst_device_provider_is_started()) and only stops actually running
+    device providers when stopping the device monitor.
+
+-   On embedded systems it can be useful to create a registry that can
+    be shared and read by multiple processes running as different users.
+    It is now possible to set the new GST_REGISTRY_MODE environment
+    variable to specify the file mode for the registry file, which by
+    default is set to be only user readable/writable.
+
+-   GstNetClientClock will signal lost sync in case the remote time
+    resets (e.g. because device power cycles), by emitting the “synced”
+    signal with synced=FALSE parameter, so applications can take action.
+
+-   gst_value_deserialize_with_pspec() allows deserialization with a
+    hint for what the target GType should be. This allows for example
+    passing arrays of flags through the command line or
+    gst_util_set_object_arg(), eg: foo="<bar,bar+baz>".
+
+-   It’s now allowed to create an empty GstVideoOverlayComposition
+    without any rectangles by passing a NULL rectangle to
+    gst_video_overlay_composition_new(). This is useful for bindings and
+    simplifies application code in some places.
+
+Tracing framework, debugging and testing improvements
+
+-   New factories tracer to list loaded elements (and other plugin
+    features). This can be useful to collect a list of elements needed
+    for an application, which then in turn can be used to create a
+    tailored minimal GStreamer build that contains just the elements
+    needed and nothing else.
+-   New plugin-feature-loaded tracing hook for use by tracers like the
+    new factories tracer
+
+-   GstHarness: Add gst_harness_set_live() so that harnesses can be set
+    to non-live and return is-live=false in latency queries if needed.
+    Default behaviour is to always return is-live=true in latency
+    queries.
+
+-   navseek: new "hold-eos" property. When enabled, the element will
+    hold back an EOS event until the next keystroke (via navigation
+    events). This can be used to keep a video sink showing the last
+    frame of a video pipeline until a key is pressed instead of tearing
+    it down immediately on EOS.
+
+-   New fakeaudiosink element: mimics an audio sink and can be used for
+    testing and CI pipelines on systems where no audio system is
+    installed or running. It differs from fakesink in that it only
+    support audio caps and syncs to the clock by default like a normal
+    audio sink. It also implements the GstStreamVolume interface like
+    most audio sinks do.
+
+-   New videocodectestsink element for video codec conformance testing:
+    Calculates MD5 checksums for video frames and skips any padding
+    whilst doing so. Can optionally also write back the video data with
+    padding removed into a file for easy byte-by-byte comparison with
+    reference data.
 
 Tools
 
--   this section will be filled in in due course
+gst-inspect-1.0
+
+-   Can sort the list of plugins by passing --sort=name as command line
+    option
+
+gst-launch-1.0
+
+-   will now error out on top-level properties that don’t exist and
+    which were silently ignored before
+-   On Windows the high-resolution clock is enabled now, which provides
+    better clock and timer performance on Windows (see Windows section
+    below for more details).
+
+gst-play-1.0
+
+-   New --start-position command line argument to start playback from
+    the specified position
+-   Audio can be muted/unmuted in interactive mode by pressing the m
+    key.
+-   On Windows the high-resolution clock is enabled now (see Windows
+    section below for more details)
+
+gst-device-monitor-1.0
+
+-   New --include-hidden command line argument to also show “hidden”
+    device providers
+
+ges-launch-1.0
+
+-   New interactive mode that allows seeking and such. Can be disabled
+    by passing the --no-interactive argument on the command line.
+-   Option to forward tags
+-   Allow using an existing clip to determine the rendering format (both
+    topology and profile) via new --profile-from command line argument.
 
 GStreamer RTSP server
 
--   this section will be filled in in due course
+-   GstRTSPMediaFactory gained API to disable RTCP
+    (gst_rtsp_media_factory_set_enable_rtcp(), "enable-rtcp" property).
+    Previously RTCP was always allowed for all RTSP medias. With this
+    change it is possible to disable RTCP completely, no matter if the
+    client wants to do RTCP or not.
+
+-   Make a mount point of / work correctly. While not allowed by the
+    RTSP 2 spec, the RTSP 1 spec is silent on this and it is used in the
+    wild. It is now possible to use / as a mount path in
+    gst-rtsp-server, e.g. rtsp://example.com/ would work with this now.
+    Note that query/fragment parts of the URI are not necessarily
+    correctly handled, and behaviour will differ between various
+    client/server implementations; so use it if you must but don’t bug
+    us if it doesn’t work with third party clients as you’d hoped.
+
+-   multithreading fixes (races, refcounting issues, deadlocks)
+
+-   ONVIF audio backchannel fixes
+
+-   ONVIF trick mode optimisations
+
+-   rtspclientsink: new "update-sdp" signal that allows updating the SDP
+    before sending it to the server via ANNOUNCE. This can be used to
+    add additional metadata to the SDP, for example. The order and
+    number of medias must not be changed, however.
 
 GStreamer VAAPI
 
--   this section will be filled in in due course
+-   new AV1 decoder element (vaapiav1dec)
+
+-   H264 decoder: handle stereoscopic 3D video with frame packing
+    arrangement SEI messages
+
+-   H265 encoder: added Screen Content Coding extensions support
+
+-   H265 decoder: gained MAIN_444_12 profile support (decoded to
+    Y412_LE), and 4:2:2 12-bits support (decoded to Y212_LE)
+
+-   vaapipostproc: gained BT2020 color standard support
+
+-   vaapidecode: now generates caps templates dynamically at runtime in
+    order to advertise actually supported caps instead of all
+    theoretically supported caps.
+
+-   GST_VAAPI_DRM_DEVICE environment variable to force a specified DRM
+    device when a DRM display is used. It is ignored when other types of
+    displays are used. By default /dev/dri/renderD128 is used for DRM
+    display.
 
 GStreamer OMX
 
--   this section will be filled in in due course
+-   subframe support in H.264/H.265 decoders
 
 GStreamer Editing Services and NLE
 
--   this section will be filled in in due course
+-   framepositioner: new "operator" property to access blending modes in
+    the compositor
+-   timeline: Implement snapping to markers
+-   smart-mixer: Add support for d3d11compositor and glvideomixer
+-   titleclip: add "draw-shadow" child property
+-   ges:// URI support to define a timeline from a description.
+-   command-line-formatter
+    -   Add track management to timeline description
+    -   Add keyframe support
+-   ges-launch-1.0:
+    -   Add an interactive mode where we can seek etc…
+    -   Add option to forward tags
+    -   Allow using an existing clip to determine the rendering format
+        (both topology and profile) via new --profile-from command line
+        argument.
+-   Fix static build
 
 GStreamer validate
 
--   this section will be filled in in due course
+-   report: Add a way to force backtraces on reports even if not a
+    critical issue (GST_VALIDATE_ISSUE_FLAGS_FORCE_BACKTRACE)
+-   Add a flag to gst_validate_replace_variables_in_string() allow
+    defining how to resolve variables in structs
+-   Add gst_validate_bin_monitor_get_scenario() to get the bin monitor
+    scenario, which is useful for applications that use Validate
+    directly.
+-   Add an expected-values parameter to wait, message-type=XX allowing
+    more precise filtering of the message we are waiting for.
+-   Add config file support: each test can now use a config file for the
+    given media file used to test.
+-   Add support to check properties of object properties
+-   scenario: Add an "action-done" signal to signal when an action is
+    done
+-   scenario: Add a "run-command" action type
+-   scenario: Allow forcing running action on idle from scenario file
+-   scenario: Allow iterating over arrays in foreach
+-   scenario: Rename ‘interlaced’ action to ‘non-blocking’
+-   scenario: Add a non-blocking flag to the wait signal
 
 GStreamer Python Bindings
 
--   this section will be filled in in due course
+-   Fixes for Python 3.10
+-   Various build fixes
+-   at least one known breaking change caused by g-i annotation changes
+    (see below)
 
 GStreamer C# Bindings
 
--   this section will be filled in in due course
+-   Fix GstDebugGraphDetails enum
+-   Updated to latests GtkSharp
+-   Updated to include GStreamer 1.20 API
 
 GStreamer Rust Bindings and Rust Plugins
 
-The GStreamer Rust bindings are released separately with a different
-release cadence that’s tied to gtk-rs, but the latest release has
-already been updated for the upcoming new GStreamer 1.20 API.
-
-gst-plugins-rs, the module containing GStreamer plugins written in Rust,
-has also seen lots of activity with many new elements and plugins.
-
-What follows is a list of elements and plugins available in
-gst-plugins-rs, so people don’t miss out on all those potentially useful
-elements that have no C equivalent.
-
--   FIXME: add new elements
-
-Rust audio plugins
-
--   audiornnoise: New element for audio denoising which implements the
-    noise removal algorithm of the Xiph RNNoise library, in Rust
--   rsaudioecho: Port of the audioecho element from gst-plugins-good
-    rsaudioloudnorm: Live audio loudness normalization element based on
-    the FFmpeg af_loudnorm filter
--   claxondec: FLAC lossless audio codec decoder element based on the
-    pure-Rust claxon implementation
--   csoundfilter: Audio filter that can use any filter defined via the
-    Csound audio programming language
--   lewtondec: Vorbis audio decoder element based on the pure-Rust
-    lewton implementation
-
-Rust video plugins
-
--   cdgdec/cdgparse: Decoder and parser for the CD+G video codec based
-    on a pure-Rust CD+G implementation, used for example by karaoke CDs
--   cea608overlay: CEA-608 Closed Captions overlay element
--   cea608tott: CEA-608 Closed Captions to timed-text (e.g. VTT or SRT
-    subtitles) converter
--   tttocea608: CEA-608 Closed Captions from timed-text converter
--   mccenc/mccparse: MacCaption Closed Caption format encoder and parser
--   sccenc/sccparse: Scenarist Closed Caption format encoder and parser
--   dav1dec: AV1 video decoder based on the dav1d decoder implementation
-    by the VLC project
--   rav1enc: AV1 video encoder based on the fast and pure-Rust rav1e
-    encoder implementation
--   rsflvdemux: Alternative to the flvdemux FLV demuxer element from
-    gst-plugins-good, not feature-equivalent yet
--   rsgifenc/rspngenc: GIF/PNG encoder elements based on the pure-Rust
-    implementations by the image-rs project
-
-Rust text plugins
-
--   textwrap: Element for line-wrapping timed text (e.g. subtitles) for
-    better screen-fitting, including hyphenation support for some
-    languages
-
-Rust network plugins
-
--   reqwesthttpsrc: HTTP(S) source element based on the Rust
-    reqwest/hyper HTTP implementations and almost feature-equivalent
-    with the main GStreamer HTTP source souphttpsrc
--   s3src/s3sink: Source/sink element for the Amazon S3 cloud storage
--   awstranscriber: Live audio to timed text transcription element using
-    the Amazon AWS Transcribe API
-
-Generic Rust plugins
-
--   sodiumencrypter/sodiumdecrypter: Encryption/decryption element based
-    on libsodium/NaCl
--   togglerecord: Recording element that allows to pause/resume
-    recordings easily and considers keyframe boundaries
--   fallbackswitch/fallbacksrc: Elements for handling potentially
-    failing (network) sources, restarting them on errors/timeout and
-    showing a fallback stream instead
--   threadshare: Set of elements that provide alternatives for various
-    existing GStreamer elements but allow to share the streaming threads
-    between each other to reduce the number of threads
--   rsfilesrc/rsfilesink: File source/sink elements as replacements for
-    the existing filesrc/filesink elements
+-   The GStreamer Rust bindings are released separately with a different
+    release cadence that’s tied to gtk-rs, but the latest release has
+    already been updated for the upcoming new GStreamer 1.20 API (v1_20
+    feature).
+
+-   gst-plugins-rs, the module containing GStreamer plugins written in
+    Rust, has also seen lots of activity with many new elements and
+    plugins. See the New Elements section above for a list of new Rust
+    elements.
 
 Build and Dependencies
 
 -   Meson 0.59 or newer is required to build GStreamer now.
 
--   FIXME: this section will be filled in in due course
+-   The GLib requirement has been bumped to GLib 2.56 or newer (from
+    March 2018).
+
+-   The wpe plugin now requires wpe >= 2.28 and wpebackend-fdo >= 1.8
 
 Explicit opt-in required for build of certain plugins with (A)GPL dependencies
 
@@ -211,11 +1591,9 @@ even if the required dependencies are available.
 See Building plugins with (A)GPL-licensed dependencies for more details
 and a non-exhaustive list of plugins affected.
 
-gst-build: replaced by Monorepo
+gst-build: replaced by mono repository
 
--   this section will be filled in in due course
-
--   FIXME: describe + link to Monorepo FAQ
+See mono repository section above and the GStreamer mono repository FAQ.
 
 Cerbero
 
@@ -223,76 +1601,254 @@ Cerbero is a meta build system used to build GStreamer plus dependencies
 on platforms where dependencies are not readily available, such as
 Windows, Android, iOS and macOS.
 
-General improvements
+General Cerbero improvements
 
--   this section will be filled in in due course
+-   Plugin removed: libvisual
+-   New plugins: rtpmanagerbad and rist
 
-macOS / iOS
+macOS / iOS specific Cerbero improvements
 
--   this section will be filled in in due course
+-   XCode 12 support
+-   macOS OS release support is now future-proof, similar to iOS
+-   macOS Apple Silicon (ARM64) cross-compile support has been added
+-   macOS Apple Silicon (ARM64) native support is currently experimental
 
-Windows
+Windows specific Cerbero improvements
 
--   this section will be filled in in due course
+-   Visual Studio 2022 support has been added
+-   bootstrap is faster since it requires building fewer build-tools
+    recipes on Windows
+-   package is faster due to better scheduling of recipe stages and
+    elimination of unnecessary autotools regeneration
+-   The following plugins are no longer built on Windows:
+    -   a52dec (another decoder is still available in libav)
+    -   dvdread
+    -   resindvd
 
 Windows MSI installer
 
--   this section will be filled in in due course
+-   no major changes
 
-Linux
+Linux specific Cerbero improvements
 
--   this section will be filled in in due course
+-   Fedora, Debian OS release support is now more future-proof
+-   Amazon Linux 2 support has been added
 
-Android
+Android specific Cerbero improvements
 
--   this section will be filled in in due course
+-   no major changes
 
 Platform-specific changes and improvements
 
 Android
 
--   this section will be filled in in due course
+-   No major changes
 
 macOS and iOS
 
--   this section will be filled in in due course
+-   applemedia: add ProRes support to vtenc and vtdec
 
 Windows
 
--   this section will be filled in in due course
+-   On Windows the high-resolution clock is enabled now in the
+    gst-launch-1.0 and gst-play-1.0 command line tools, which provides
+    better clock and timer performance on Windows, at the cost of higher
+    power consumption. By default, without the high-resolution clock
+    enabled, the timer precision on Windows is system-dependent and may
+    be as bad as 15ms which is not good enough for many multimedia
+    applications. Developers may want to do the same in their Windows
+    applications if they think it’s a good idea for their application
+    use case, and depending on the Windows version they target. This is
+    not done automatically by GStreamer because on older Windows
+    versions (pre-Windows 10) this affects a global Windows setting and
+    also there’s a power consumption vs. performance trade-off that may
+    differ from application to application.
+
+-   dxgiscreencapsrc now supports resolution changes
+
+-   The wasapi2 audio plugin was rewritten and now has a higher rank
+    than the old wasapi plugin since it has a number of additional
+    features such as automatic stream routing, and no
+    known-but-hard-to-fix issues. The plugin is always built if the
+    Windows 10 SDK is available now.
+
+-   The wasapi device providers now detect and notify dynamic device
+    additions/removals
+
+-   d3d11screencapturesrc: new desktop capture element, including
+    GstDeviceProvider implementation to enumerate/select target monitors
+    for capture.
+
+-   Direct3D11/DXVA decoder now supports AV1 and MPEG2 codecs
+    (d3d11av1dec, d3d11mpeg2dec)
+
+-   VP9 decoding got more reliable and stable thanks to a newly written
+    codec parser
+
+-   Support for decoding interlaced H.264/AVC streams
+
+-   Hardware-accelerated video deinterlacing (d3d11deinterlace) and
+    video mixing (d3d11compositor)
+
+-   Video mixing with the Direct3D11 API (d3d11compositor)
+
+-   MediaFoundation API based hardware encoders gained the ability to
+    receive Direct3D11 textures as an input
+
+-   Seungha’s blog post “GStreamer ❤ Windows: A primer on the cool stuff
+    you’ll find in the 1.20 release” describes many of the
+    Windows-related improvements in more detail
 
 Linux
 
--   this section will be filled in in due course
+-   bluez: LDAC Bluetooth audio codec support in a2dpsink and avdtpsink,
+    as well as an LDAC RTP payloader (rtpldacpay) and an LDAC audio
+    encoder (ldacenc)
+
+-   kmssink: gained support for NV24, NV61, RGB16/BGR16 formats;
+    auto-detect NVIDIA Tegra driver
 
 Documentation improvements
 
--   this section will be filled in in due course
+-   hardware-accelerated GPU plugins will now no longer always list all
+    the element variants for all available GPUs, since those are
+    system-dependent and it’s confusing for users to see those in the
+    documentation just because the GStreamer developer who generated the
+    docs had multiple GPUs to play with at the time. Instead just show
+    the default elements.
 
-Possibly Breaking Changes
+Possibly Breaking and Other Noteworthy Behavioural Changes
+
+-   gst_parse_launch(), gst_parse_bin_from_description() and friends
+    will now error out when setting properties that don’t exist on
+    top-level bins. They were silently ignored before.
+
+-   The GstWebRTC library does not expose any objects anymore with
+    public fields. Instead properties have been added to replace that
+    functionality. If you are accessing such fields in your application,
+    switch to the corresponding properties.
 
--   FIXME: this section will be filled in in due course
--   MPEG-TS SCTE-35 API changes (FIXME: flesh out)
--   gst_parse_launch() and friends now error out on non-existing
-    properties on top-level bins where they would silently fail and
-    ignore those before.
 -   playbin and uridecodebin now emit the source-setup signal before the
     element is added to the bin and linked so that the source element is
     already configured before any scheduling query comes in, which is
-    useful for elements such as appsrc or giostreamsrc. (Merge Request)
+    useful for elements such as appsrc or giostreamsrc.
+
+-   The source element inside urisourcebin (used inside uridecodebin3
+    which is used inside playbin3) is no longer called "source". This
+    shouldn’t affect anyone hopefully, because there’s a "setup-source"
+    signal to configure the source element and no one should rely on
+    names of internal elements anyway.
+
+-   The vp8enc element now expects bps (bits per second) for the
+    "temporal-scalability-target-bitrate" property, which is consistent
+    with the "target-bitrate" property. Since additional configuration
+    is required with modern libvpx to make temporal scaling work anyway,
+    chances are that very few people will have been using this property
+
+-   vp8enc and vp9enc now default to “good quality” for the "deadline"
+    property rather then “best quality”. Having the deadline set to best
+    quality causes the encoder to be absurdly slow, most real-life users
+    will want the good quality tradeoff instead.
+
+-   The experimental GstTranscoder library API in gst-plugins-bad was
+    changed from a GObject signal-based notification mechanism to a
+    GstBus/message-based mechanism akin to GstPlayer/GstPlay.
+
+-   MPEG-TS SCTE-35 API: semantic change for SCTE-35 splice commands:
+    timestamps passed by the application should be in running time now,
+    since users of the API can’t really be expected to predict the local
+    PTS of the muxer.
+
+-   The GstContext used by souphttpsrc to share the session between
+    multiple element instances has changed. Previously it provided
+    direct access to the internal SoupSession object, now it only
+    provides access to an opaque, internal type. This change is
+    necessary because SoupSession is not thread-safe at all and can’t be
+    shared safely between arbitrary external code and souphttpsrc.
+
+-   Python bindings: GObject-introspection related Annotation fixes have
+    led to a case of a GstVideo.VideoInfo-related function signature
+    changing in the Python bindings (possibly one or two other cases
+    too). This is for a function that should never have been exposed in
+    the first place though, so the bindings are being updated to throw
+    an exception in that case, and the correct replacement API has been
+    added in form of an override.
 
 Known Issues
 
--   this section will be filled in in due course
-
--   There are a couple of known WebRTC-related regressions/blockers:
-
-    -   webrtc: DTLS setup with Chrome is broken
-    -   webrtcbin: First keyframe is usually lost
+-   nothing in particular at this point (but also see possibly breaking
+    changes section above)
 
 Contributors
 
--   this section will be filled in in due course
+Aaron Boxer, Adam Leppky, Adam Williamson, Alba Mendez, Alejandro
+González, Aleksandr Slobodeniuk, Alexander Vandenbulcke, Alex Ashley,
+Alicia Boya García, Andika Triwidada, Andoni Morales Alastruey, Andrew
+Wesie, Andrey Moiseev, Antonio Ospite, Antonio Rojas, Arthur Crippa
+Búrigo, Arun Raghavan, Ashley Brighthope, Axel Kellermann, Baek, Bastien
+Nocera, Bastien Reboulet, Benjamin Gaignard, Bing Song, Binh Truong,
+Biswapriyo Nath, Brad Hards, Brad Smith, Brady J. Garvin, Branko
+Subasic, Camilo Celis Guzman, Chris Bass, ChrisDuncanAnyvision, Chris
+White, Corentin Damman, Daniel Almeida, Daniel Knobe, Daniel Stone,
+david, David Fernandez, David Keijser, David Phung, Devarsh Thakkar,
+Dinesh Manajipet, Dmitry Samoylov, Dmitry Shusharin, Dominique Martinet,
+Doug Nazar, Ederson de Souza, Edward Hervey, Emmanuel Gil Peyrot,
+Enrique Ocaña González, Ezequiel Garcia, Fabian Orccon, Fabrice
+Fontaine, Fernando Jimenez Moreno, Florian Karydes, Francisco Javier
+Velázquez-García, François Laignel, Frederich Munch, Fredrik Pålsson,
+George Kiagiadakis, Georg Lippitsch, Göran Jönsson, Guido Günther,
+Guillaume Desmottes, Guiqin Zou, Haakon Sporsheim, Haelwenn (lanodan)
+Monnier, Haihao Xiang, Haihua Hu, Havard Graff, He Junyan, Helmut
+Januschka, Henry Wilkes, Hosang Lee, Hou Qi, Ignacio Casal Quinteiro,
+Igor Kovalenko, Ilya Kreymer, Imanol Fernandez, Jacek Tomaszewski, Jade
+Macho, Jakub Adam, Jakub Janků, Jan Alexander Steffens (heftig), Jan
+Schmidt, Jason Carrete, Jason Pereira, Jay Douglass, Jeongki Kim, Jérôme
+Laheurte, Jimmi Holst Christensen, Johan Sternerup, John Hassell, John
+Lindgren, John-Mark Bell, Jonathan Matthew, Jordan Petridis, Jose
+Quaresma, Julian Bouzas, Julien, Kai Uwe Broulik, Kasper Steensig
+Jensen, Kellermann Axel, Kevin Song, Khem Raj, Knut Inge Hvidsten, Knut
+Saastad, Kristofer Björkström, Lars Lundqvist, Lawrence Troup, Lim Siew
+Hoon, Lucas Stach, Ludvig Rappe, Luis Paulo Fernandes de Barros, Luke
+Yelavich, Mads Buvik Sandvei, Marc Leeman, Marco Felsch, Marek Vasut,
+Marian Cichy, Marijn Suijten, Marius Vlad, Markus Ebner, Mart Raudsepp,
+Matej Knopp, Mathieu Duponchelle, Matthew Waters, Matthieu De Beule,
+Mengkejiergeli Ba, Michael de Gans, Michael Olbrich, Michael Tretter,
+Michal Dzik, Miguel Paris, Mikhail Fludkov, mkba, Nazar Mokrynskyi,
+Nicholas Jackson, Nicola Murino, Nicolas Dufresne, Niklas Hambüchen,
+Nikolay Sivov, Nirbheek Chauhan, Olivier Blin, Olivier Crete, Olivier
+Crête, Paul Goulpié, Per Förlin, Peter Boba, P H, Philippe Normand,
+Philipp Zabel, Pieter Willem Jordaan, Piotrek Brzeziński, Rafał
+Dzięgiel, Rafostar, raghavendra, Raghavendra, Raju Babannavar, Raleigh
+Littles III, Randy Li, Randy Li (ayaka), Ratchanan Srirattanamet, Raul
+Tambre, reed.lawrence, Ricky Tang, Robert Rosengren, Robert Swain, Robin
+Burchell, Roman Sivriver, R S Nikhil Krishna, Ruben Gonzalez, Ruslan
+Khamidullin, Sanchayan Maity, Scott Moreau, Sebastian Dröge, Sergei
+Kovalev, Seungha Yang, Sid Sethupathi, sohwan.park, Sonny Piers, Staz M,
+Stefan Brüns, Stéphane Cerveau, Stephan Hesse, Stian Selnes, Stirling
+Westrup, Théo MAILLART, Thibault Saunier, Tim, Timo Wischer, Tim-Philipp
+Müller, Tim Schneider, Tobias Ronge, Tom Schoonjans, Tulio Beloqui,
+tyler-aicradle, U. Artie Eoff, Ung, Val Doroshchuk, VaL Doroshchuk,
+Víctor Manuel Jáquez Leal, Vivek R, Vivia Nikolaidou, Vivienne
+Watermeier, Vladimir Menshakov, Will Miller, Wim Taymans, Xabier
+Rodriguez Calvar, Xavier Claessens, Xℹ Ruoyao, Yacine Bandou, Yinhang
+Liu, youngh.lee, youngsoo.lee, yychao, Zebediah Figura, Zhang yuankun,
+Zhang Yuankun, Zhao, Zhao Zhili, , Aleksandar Topic, Antonio Ospite,
+Bastien Nocera, Benjamin Gaignard, Brad Hards, Carlos Falgueras García,
+Célestin Marot, Corentin Damman, Corentin Noël, Daniel Almeida, Daniel
+Knobe, Danny Smith, Dave Piché, Dmitry Osipenko, Fabrice Fontaine,
+fjmax, Florian Zwoch, Guillaume Desmottes, Haihua Hu, Heinrich Kruger,
+He Junyan, Jakub Adam, James Cowgill, Jan Alexander Steffens (heftig),
+Jean Felder, Jeongki Kim, Jiri Uncovsky, Joe Todd, Jordan Petridis,
+Krystian Wojtas, Marc-André Lureau, Marcin Kolny, Marc Leeman, Mark
+Nauwelaerts, Martin Reboredo, Mathieu Duponchelle, Matthew Waters,
+Mengkejiergeli Ba, Michael Gruner, Nicolas Dufresne, Nirbheek Chauhan,
+Olivier Crête, Philippe Normand, Rafał Dzięgiel, Ralf Sippl, Robert
+Mader, Sanchayan Maity, Sangchul Lee, Sebastian Dröge, Seungha Yang,
+Stéphane Cerveau, Teh Yule Kim, Thibault Saunier, Thomas Klausner, Timo
+Wischer, Tim-Philipp Müller, Tobias Reineke, Tomasz Andrzejak, Trung Do,
+Tyler Compton, Ung, Víctor Manuel Jáquez Leal, Vivia Nikolaidou, Wim
+Taymans, wngecn, Wonchul Lee, wuchang li, Xavier Claessens, Xi Ruoyao,
+Yoshiharu Hirose, Zhao,
 
 … and many others who have contributed bug reports, translations, sent
 suggestions or helped testing.
@@ -307,7 +1863,7 @@ the git 1.20 branch, which will be a stable branch.
 
 1.20.0
 
-1.20.0 is scheduled to be released around October/November 2021.
+1.20.0 is scheduled to be released around early February 2022.
 
 Schedule for 1.22
 
@@ -315,7 +1871,9 @@ Our next major feature release will be 1.22, and 1.21 will be the
 unstable development version leading up to the stable 1.22 release. The
 development of 1.21/1.22 will happen in the git main branch.
 
-The plan for the 1.22 development cycle is yet to be confirmed.
+The plan for the 1.22 development cycle is yet to be confirmed. Assuming
+no major project-wide reorganisations in the 1.22 cycle we might try and
+aim for a release around August 2022.
 
 1.22 will be backwards-compatible to the stable 1.20, 1.18, 1.16, 1.14,
 1.12, 1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series.
@@ -323,6 +1881,7 @@ The plan for the 1.22 development cycle is yet to be confirmed.
 ------------------------------------------------------------------------
 
 These release notes have been prepared by Tim-Philipp Müller with
-contributions from …
+contributions from Matthew Waters, Nicolas Dufresne, Nirbheek Chauhan,
+Sebastian Dröge and Seungha Yang.
 
 License: CC BY-SA 4.0
diff --git a/subprojects/gstreamer/RELEASE b/subprojects/gstreamer/RELEASE
index 1f27cd4cd6..a0910fddd8 100644
--- a/subprojects/gstreamer/RELEASE
+++ b/subprojects/gstreamer/RELEASE
@@ -1,4 +1,4 @@
-This is GStreamer core 1.19.3.
+This is GStreamer core 1.19.90.
 
 GStreamer 1.19 is the development branch leading up to the next major
 stable version which will be 1.20.
diff --git a/subprojects/gstreamer/gstreamer.doap b/subprojects/gstreamer/gstreamer.doap
index f0ceeb6ba7..5807c216b7 100644
--- a/subprojects/gstreamer/gstreamer.doap
+++ b/subprojects/gstreamer/gstreamer.doap
@@ -40,6 +40,16 @@ hierarchy, and a set of media-agnostic core elements.
 
  <release>
   <Version>
+   <revision>1.19.90</revision>
+   <branch>main</branch>
+   <name></name>
+   <created>2022-01-28</created>
+   <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gstreamer/gstreamer-1.19.90.tar.xz" />
+  </Version>
+ </release>
+
+ <release>
+  <Version>
    <revision>1.19.3</revision>
    <branch>main</branch>
    <name></name>
diff --git a/subprojects/gstreamer/meson.build b/subprojects/gstreamer/meson.build
index 5bb5ccb3e6..b0c5c7d4df 100644
--- a/subprojects/gstreamer/meson.build
+++ b/subprojects/gstreamer/meson.build
@@ -1,5 +1,5 @@
 project('gstreamer', 'c',
-  version : '1.19.3.1',
+  version : '1.19.90',
   meson_version : '>= 0.59',
   default_options : [ 'warning_level=1',
                       'buildtype=debugoptimized' ])