-GStreamer 1.20 Release Notes
+GStreamer 1.22 Release Notes
-GStreamer 1.20.0 was released on 3 February 2022.
+GStreamer 1.22 has not been released yet. It is scheduled for release
+around in January 2023.
-See https://gstreamer.freedesktop.org/releases/1.20/ for the latest
+1.21.x is the unstable development version that is being developed in
+the git master branch and which will eventually result in 1.22, and
+1.21.90 is the first release candidate for 1.22 (rc1) in that series
+
+A feature freeze is now into effect for the 1.21 branch, but newly-added
+API might still change until the final 1.22.0 stable release, and minor
+features may also still be added until then.
+
+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.
+
+See https://gstreamer.freedesktop.org/releases/1.22/ for the latest
version of this document.
-Last updated: Wednesday 2 February 2022, 23:30 UTC (log)
+Last updated: Thursday 12 January 2023, 15:00 UTC (log)
Introduction
Highlights
-- 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 (pass through) 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 H.264/H.265 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 implementation 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 MPEG-2 and VP9
-- Windows Direct3D11/DXVA decoder gained AV1 and MPEG-2 support
-- Lots of new plugins, features, performance improvements and bug
- fixes
+- this section will be completed in due course
Major new features and changes
Noteworthy new features and API
-- 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 system’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.
- - 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 usual 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;
- registering the same custom meta multiple times or from multiple
- places is not allowed.
-
-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-plugin feature 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-plugin
- feature 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.
+- this section will be filled in in due course
New elements
-- 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 allow overlaying data
- on top of video in the 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 MPEG-2 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
+- this section will be filled in in due course
New element features and additions
-- 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 synchronisation,
- 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 H.265 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).
- - H.264/H.265 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 H.264 and H.265
- (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 pass-through via new "send-scte35-events"
- property on MPEG-TS demuxer tsdemux. When enabled, SCTE 35 sections
- (e.g. ad placement opportunities) are forwarded as events downstream
- 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 analysers 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 H.264/H.265 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 and 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. Distro
- packagers please take note of the souphttpsrc plugin dependency
- changes mentioned in the build and dependencies section below.
-
-- 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
-
-- 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
-
-- 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.
+- this section will be filled in in due course
Plugin and library moves
+- this section will be filled in in due course
+
- There were no plugin moves or library moves in this cycle.
Plugin removals
The following elements or plugins have been removed:
-- 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.
-
-- The mms plugin containing mmssrc has been removed. It seems unlikely
- anyone still needs this or that there are even any streams left out
- there. The MMS protocol was deprecated in 2003 (in favour of RTSP)
- and support for it was dropped with Microsoft Media Services 2008,
- and Windows Media Player apparently also does not support it any
- more.
+- this section will be filled in in due course
Miscellaneous API additions
-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
- task pool 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 H.264 AvcC codec_data. The format of H.264
- 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 container formats,
- such as mp4 or Matroska.
-
-GStreamer OpenGL integration library and plugins
-
-- glcolorconvert: added support 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 (H.264, H.265, VP8, VP9, AV1, MPEG-2) used for various
- hardware-accelerated 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() afterwards). 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.
+- this section will be filled in in due course
Miscellaneous performance, latency and memory optimisations
-More video conversion fast paths
-
-- 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
-
-- 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.
-
-- 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 deserialisation 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 possible 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 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
-
-gst-inspect-1.0
-
-- Can sort the list of plugins by passing --sort=name as command line
- option
-
-gst-launch-1.0
+- this section will be filled in in due course
-- 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).
+Tracing framework and debugging improvements
-gst-play-1.0
+- this section will be filled in in due course
-- 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
+Tools
-- 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.
+- this section will be filled in in due course
GStreamer RTSP server
-- 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, irrespective of
- whether 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.
+- this section will be filled in in due course
GStreamer VAAPI
-- new AV1 decoder element (vaapiav1dec)
-
-- H.264 decoder: handle stereoscopic 3D video with frame packing
- arrangement SEI messages
-
-- H.265 encoder: added Screen Content Coding extensions support
-
-- H.265 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.
+- this section will be filled in in due course
GStreamer OMX
-- subframe support in H.264/H.265 decoders
+- this section will be filled in in due course
GStreamer Editing Services and NLE
-- 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
+- this section will be filled in in due course
GStreamer validate
-- 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
+- this section will be filled in in due course
GStreamer Python Bindings
-- Fixes for Python 3.10
-- Various build fixes
-- at least one known breaking change caused by g-i annotation changes
- (see below)
+- this section will be filled in in due course
GStreamer C# Bindings
-- Fix GstDebugGraphDetails enum
-- Updated to latest GtkSharp
-- Updated to include GStreamer 1.20 API
+- this section will be filled in in due course
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 (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.
+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 1.22 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
Build and Dependencies
-- Meson 0.59 or newer is now required to build GStreamer.
-
-- The GLib requirement has been bumped to GLib 2.56 or newer (from
- March 2018).
+- this section will be filled in in due course
-- The wpe plugin now requires wpe >= 2.28 and wpebackend-fdo >= 1.8
+- GLib >= 2.64 is now required (technically it still says >= 2.62 but
+ the plan is to bump that to 2.64 soon once some CI issues are sorted
+ out)
-- The souphttpsrc plugin is no longer linked against libsoup but
- instead tries to pick up either libsoup2 or libsoup3 dynamically at
- runtime. Distro packagers please ensure to add a dependency on one
- of the libsoup runtimes to the gst-plugins-good package so that
- there is actually a libsoup for the plugin to find!
+- libnice >= 0.1.21 is now required and contains important fixes for
+ GStreamer’s WebRTC stack.
-Explicit opt-in required for build of certain plugins with (A)GPL dependencies
+gst-build
-Some plugins have GPL- or AGPL-licensed dependencies and those plugins
-will no longer be built by default unless you have explicitly opted in
-to allow (A)GPL-licensed dependencies by passing -Dgpl=enabled to Meson,
-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 mono repository
-
-See mono repository section above and the GStreamer mono repository FAQ.
+- this section will be filled in in due course
Cerbero
on platforms where dependencies are not readily available, such as
Windows, Android, iOS and macOS.
-General Cerbero improvements
+General improvements
-- Plugin removed: libvisual
-- New plugins: rtpmanagerbad and rist
+- this section will be filled in in due course
-macOS / iOS specific Cerbero improvements
+macOS / iOS
-- XCode 12 support
-- macOS OS release support is now future-proof, similar to iOS
-- macOS Apple Silicon (ARM64) cross-compile support has been added,
- including Universal binaries. There is a known bug regarding this on
- ARM64.
-- Running Cerbero itself on macOS Apple Silicon (ARM64) is currently
- experimental and is known to have bugs
+- this section will be filled in in due course
-Windows specific Cerbero improvements
+Windows
-- 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
+- this section will be filled in in due course
Windows MSI installer
-- no major changes
+- this section will be filled in in due course
-Linux specific Cerbero improvements
+Linux
-- Fedora, Debian OS release support is now more future-proof
-- Amazon Linux 2 support has been added
+- this section will be filled in in due course
-Android specific Cerbero improvements
+Android
-- no major changes
+- this section will be filled in in due course
Platform-specific changes and improvements
Android
-- No major changes
+- this section will be filled in in due course
macOS and iOS
-- applemedia: add ProRes support to vtenc and vtdec
-
-- The GStreamer.framework location is now relocatable and is not
- required to be /Library/Frameworks/
-
-- Cerbero now supports cross-compiling to macOS running on Apple
- Silicon (ARM64), and Universal binaries are now available that can
- be used on both X86_64 and ARM64 macOS.
+- this section will be filled in in due course
Windows
-- 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.
+- this section will be filled in in due course
-- Direct3D11/DXVA decoder now supports AV1 and MPEG-2 codecs
- (d3d11av1dec, d3d11mpeg2dec)
+- The Windows universal UWP build is currently non-functional and will
+ need updating after the recent GLib upgrade. It is unclear if anyone
+ is using these binaries, so if you are please make yourself known.
-- VP9 decoding got more reliable and stable thanks to a newly written
- codec parser
-
-- Support for decoding interlaced H.264/AVC streams
+Linux
-- Hardware-accelerated video deinterlacing (d3d11deinterlace) and
- video mixing (d3d11compositor)
+- this section will be filled in in due course
-- Video mixing with the Direct3D11 API (d3d11compositor)
+Documentation improvements
-- MediaFoundation API based hardware encoders gained the ability to
- receive Direct3D11 textures as an input
+- this section will be filled in in due course
-- 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
+Possibly Breaking Changes
-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)
+- the Opus audio payloader and depayloader no longer accept the lower
+ case encoding-format=multiopus but instead produce and accept only
+ the upper case variant encoding-format=MULTIOPUS, since those should
+ always be upper case in GStreamer (caps fields are always case
+ sensitive). This should hopefully only affect applications where RTP
+ caps are set manually and multi-channel audio (>= 3 channels) is
+ used.
-- kmssink: gained support for NV24, NV61, RGB16/BGR16 formats;
- auto-detect NVIDIA Tegra driver
+Known Issues
-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 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.
-
-- 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.
-
-- 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 regressions/blockers:
-Known Issues
+ - FIXME
-- nothing in particular at this point (but also see possibly breaking
- changes section above)
+- GStreamer may fail to build the hotdoc documentation with the Meson
+ 0.64.0 release owing to a Meson bug. This should only affect systems
+ where hotdoc is installed, and will be fixed in Meson 0.64.1 by
+ Meson PR 10982 in combination with GStreamer MR 3352. In the
+ meantime, users can pass -Ddoc=disabled or downgrade to an older
+ Meson version (< 0.64.0).
Contributors
-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,
+Ádám Balázs, Adam Doupe, Adrian Fiergolski, Adrian Perez de Castro, Alba
+Mendez, Aleix Conchillo Flaqué, Aleksandr Slobodeniuk, Alicia Boya
+García, Alireza Miryazdi, Andoni Morales Alastruey, Andrew Pritchard,
+Arun Raghavan, Bastian Krause, Bastien Nocera, Benjamin Gaignard, Brad
+Hards, Branko Subasic, Bruce Liang, Camilo Celis Guzman, Carlos
+Falgueras García, Carlos Rafael Giani, Célestin Marot, Christopher
+Obbard, Christoph Reiter, Chris Wiggins, Chun-wei Fan, Corentin Damman,
+Corentin Noël, Damian Hobson-Garcia, Daniel Almeida, Daniel Morin,
+Daniel Stone, Daniels Umanovskis, Danny Smith, David Svensson Fors,
+Devin Anderson, Diogo Goncalves, Dmitry Osipenko, Dongil Park, Doug
+Nazar, Edward Hervey, Eli Schwartz, Elliot Chen, Enrique Ocaña González,
+Eric Knapp, Erwann Gouesbet, Fabian Orccon, Fabrice Fontaine, Fan F He,
+fduncanh, Filip Hanes, Florian Zwoch, François Laignel, Fuga Kato,
+George Kiagiadakis, Guillaume Desmottes, Gu Yanjie, Haihao Xiang, Haihua
+Hu, Havard Graff, Heiko Becker, He Junyan, Hoonhee Lee, Hosang Lee, Hou
+Qi, Ignacio Casal Quinteiro, Ignazio Pillai, Igor V. Kovalenko, Jakub
+Adam, James Cowgill, James Hilliard, Jan Alexander Steffens (heftig),
+Jan Schmidt, Jianhui Dai, jinsl00000, Johan Sternerup, Jonas Bonn, Jonas
+Danielsson, Jordan Petridis, Joseph Donofry, Jose Quaresma, Julian
+Bouzas, Junsoo Park, Justin Chadwell, Khem Raj, Krystian Wojtas, László
+Károlyi, Linus Svensson, Loic Le Page, Loïc Le Page, Ludvig Rappe, Marc
+Leeman, Marek Vasut, Marijn Suijten, Mark Nauwelaerts, Martin Dørum,
+Martin Reboredo, Mart Raudsepp, Mathieu Duponchelle, Matt Crane, Matthew
+Waters, Matthias Clasen, Matthias Fuchs, Mengkejiergeli Ba, MG
+Lolenstine, Michael Gruner, Michal Kubiak, Mikhail Fludkov, Ming Qian,
+Myles Inglis, Nicolas Dufresne, Nirbheek Chauhan, Olivier Crête,
+Patricia Muscalu, Patrick Griffis, Paweł Stawicki, Peter Stensson,
+Philippe Normand, Philipp Zabel, Pierre Bourré, Piotr Brzeziński,
+Piotrek Brzeziński, Rabindra Harlalka, Rafael Caricio, Rafael Sobral,
+Raul Tambre, Robert Mader, Robert Rosengren, Rouven Czerwinski, Ruben
+Gonzalez, Sam Van Den Berge, Sanchayan Maity, Sangchul Lee, Sebastian
+Dröge, Sebastian Fricke, Sebastian Groß, Sebastian Mueller, Sebastian
+Wick, Sergei Kovalev, Seungha Yang, Sherrill Lin, Shingo Kitagawa,
+Stéphane Cerveau, Thibault Saunier, Tim Mooney, Tim-Philipp Müller,
+Tomasz Andrzejak, Tom Schuring, Tong Wu, toor, Tristan Matthews, Tulio
+Beloqui, U. Artie Eoff, Víctor Manuel Jáquez Leal, Vincent Cheah Beng
+Keat, Vivia Nikolaidou, Vivienne Watermeier, WANG Xuerui, Wojciech
+Kapsa, Wonchul Lee, Wu Tong, Xabier Rodriguez Calvar, Xavier Claessens,
+Yatin Maan, Yeongjin Jeong, Zebediah Figura, Zhao Zhili, Zhiyuan Liu,
… and many others who have contributed bug reports, translations, sent
suggestions or helped testing.
-Stable 1.20 branch
+Stable 1.22 branch
-After the 1.20.0 release there will be several 1.20.x bug-fix releases
+After the 1.22.0 release there will be several 1.22.x bug-fix releases
which will contain bug fixes which have been deemed suitable for a
stable branch, but no new features or intrusive changes will be added to
-a bug-fix release usually. The 1.20.x bug-fix releases will be made from
-the git 1.20 branch, which will be a stable branch.
+a bug-fix release usually. The 1.22.x bug-fix releases will be made from
+the git 1.22 branch, which will be a stable branch.
-1.20.0
+1.22.0
-1.20.0 was released on 3 February 2022.
+1.22.0 is scheduled to be released in January 2023.
-Schedule for 1.22
+Schedule for 1.24
-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.
+Our next major feature release will be 1.24, and 1.23 will be the
+unstable development version leading up to the stable 1.24 release. The
+development of 1.23/1.24 will happen in the git main branch of the
+GStreamer mono repository.
-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.
+The plan for the 1.24 development cycle is yet to be confirmed.
-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.
+1.24 will be backwards-compatible to the stable 1.22, 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.
------------------------------------------------------------------------
These release notes have been prepared by Tim-Philipp Müller with
-contributions from Matthew Waters, Nicolas Dufresne, Nirbheek Chauhan,
-Sebastian Dröge and Seungha Yang.
+contributions from …
License: CC BY-SA 4.0
projects / platform / upstream / gstreamer.git / blobdiff